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[Resolved] Call not ending (with CANCEL) after hanging up
28 posts
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[Resolved] Call not ending (with CANCEL) after hanging up
by iceonu » 10/29/2021
I switched my Linksys ATA with a new Grandstream GXP2135. All seems to work except for outgoing calls from the Grandstream do not end when you hang up or press the end call button. Incoming calls work fine and will end when you hangup on the Grandstream. Ive checked NAT settings, its set to stay-alive. Ive open port 5060 on my network. Cant seem to find what's causing the issue.
- iceonu
- Just Passing Thru
- Posts: 13
- Joined: 06/04/2013
- SIP Device Name: Grandstream GXP2135
- ISP Name: Telus
- Computer OS: OS11.6.1
- Router: Ubiquiti UDM-Pro
Re: Call not ending after hanging up
by Liptonbrisk » 10/29/2021
iceonu wrote:I switched my Linksys ATA with a new Grandstream GXP2135. All seems to work except for outgoing calls from the Grandstream do not end when you hang up or press the end call button.
How are you determining the call doesn't end? The other person has answered, you hang up, and then on your end, you see a 3 hour call in your call logs after logging in at https://www.freephoneline.ca/doGetCallLogs?
Call duration limit is 3 hours with Freephoneline. If the call doesn't end without BYE being received and 200 OK sent back, the call may last that long.
The following is based on the assumption that the call really doesn't end and just continues for hours after you hang up a call that has been answered by the other side.
BYE--->
<---200 OK
I would expect that BYE is not being received or SIP signalling is being interrupted. A SIP log and possibly a packet capture may be needed to determine what's going on.
However, the issue is likely SIP ALG or NAT related.
You haven't updated your user forum profile in some time.
1, a) What brand and model modem are you using?
b)What brand and model router are you using?
2. Refer to viewtopic.php?f=8&t=20199 and ensure SIP ALG or SIP Passthrough is disabled in both your modem/router combo (if you have one that's not just a modem) and in your router.
If you were issued a modem/router combo by your ISP, ensure that is in bridge mode if you are also using your own NAT router.
3. Change primary SIP server to voip4.freephoneline.ca:6060
The purpose of voip4.freephoneline.ca:6060 is to help circumvent SIP ALG.
4.a) For your IP Phone, ensure that you're using the latest firmware: http://www.grandstream.com/support/firmware
b) change SIP REGISTER Contact Header Uses to WAN address
c) Enable SIP OPTIONS/Notify Keep Alive Yes
d) OPTIONS/NOTIFY KeepAlive: Make sure that one is chosen (do not pick "No")
e) SIP OPTIONS/Notify Keep Alive Interval: 20 seconds
Apply all changes.
Reboot modem (wait for it to be fully up and running)-->Reboot router (wait for it to be fully up and running)-->reboot IP Phone
Test again.
If that doesn't work, then you may want to ask on the Grandstream forums for instructions on to obtain a SIP call log https://forums.grandstream.com and/or packet trace.
By the way, port forwarding or opening ports is a security risk, especially UDP 5060. You are opening yourself up to SIP scanners. If you have Use Random SIP Port set to YES, which you should according to https://support.freephoneline.ca/hc/en- ... redentials and are using voip4.freephoneline.ca:6060, then forwarding or opening UDP 5060 is useless anyway. Only port forward when no other solution exists, and never use DMZ.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up
by newaira » 10/29/2021
I've got the same issue with my Grandstream HT802. I've managed to figure out how to get the SIP logs, and one thing that stands out to me is that after every request sent to the server, the response is `SIP/2.0 401 Unauthorized`. Any idea if this is normal?
Last edited by Liptonbrisk on 10/29/2021, edited 1 time in total.
Reason: deleted log because public IP addess was shown
Reason: deleted log because public IP addess was shown
- newaira
- Just Passing Thru
- Posts: 4
- Joined: 10/29/2021
- SIP Device Name: Grandstream HT802
- Firmware Version: 1.0.31.1
- ISP Name: Acanac Cable
- Computer OS: Windows
- Router: SMART/RG SR808ac
Re: Call not ending after hanging up
by Liptonbrisk » 10/29/2021
I haven't looked at your logs fully because I saw your public WAN IP in the log. So I removed the log, and in the process, I stupidly lost your log that I had copied (sorry).
I did scroll and look for BYE, but I just saw CANCEL instead.
Did you hang up before the call was answered?
That's not necessarily the same scenario as the original poster's.
How are you determining that the call doesn't end?
A (wireless DECT) phone ringing and a SIP call session not ending are two separate things.
An initial 401 unauthorized can be normal. I see it my own call logs. The server sends a challenge, and then the user's device replies with a re-invite that includes the login and hashed password. My calls end normally.
401 in this case shouldn't be the problem, particularly because the call connects.
If I were you I'd go through the same steps I listed for the previous poster. The settings should be similar, if not the same.
When you call 416-342-9562 and hang up after the other end answers, the call continues?
http://thetestcall.blogspot.com/
I did scroll and look for BYE, but I just saw CANCEL instead.
Did you hang up before the call was answered?
That's not necessarily the same scenario as the original poster's.
How are you determining that the call doesn't end?
A (wireless DECT) phone ringing and a SIP call session not ending are two separate things.
An initial 401 unauthorized can be normal. I see it my own call logs. The server sends a challenge, and then the user's device replies with a re-invite that includes the login and hashed password. My calls end normally.
401 in this case shouldn't be the problem, particularly because the call connects.
If I were you I'd go through the same steps I listed for the previous poster. The settings should be similar, if not the same.
When you call 416-342-9562 and hang up after the other end answers, the call continues?
http://thetestcall.blogspot.com/
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up
by Liptonbrisk » 10/29/2021
It is also possible there's an interconnect carrier issue with BYEs not receiving 200 OK responses or being acknowledged.
Again, this is based on the assumption that the call really doesn't end and just continues for hours after you hang up on a call that has been answered by the other end.
That was the case back in 2020: viewtopic.php?f=8&t=19966.
If that is the case, then someone would have to figure out what carrier is involved and submit a ticket with the numbers they're calling.
For example, submit a ticket: https://support.fongo.com/hc/en-us/requests/new
Make sure you provide your account phone numbers.
Also, provide the phone numbers you're dialing that are causing the problem, and it may also help if you happen to know the carrier on the other end of the call.
It might be a good idea to let Fongo Support know what numbers are likely culprits so that they can try to narrow down which carrier (Rogers, for example) is involved in the problem.
I can't confirm there's a problem without calling the same numbers, and I don't want to do that unless the problem can be reproduced by calling a common business number (Shoppers, CIBC, etc.).
I have been calling businesses and Telus carrier numbers today. I haven't had issues with calls not ending after I hang up.
Again, this is based on the assumption that the call really doesn't end and just continues for hours after you hang up on a call that has been answered by the other end.
That was the case back in 2020: viewtopic.php?f=8&t=19966.
If that is the case, then someone would have to figure out what carrier is involved and submit a ticket with the numbers they're calling.
For example, submit a ticket: https://support.fongo.com/hc/en-us/requests/new
Make sure you provide your account phone numbers.
Also, provide the phone numbers you're dialing that are causing the problem, and it may also help if you happen to know the carrier on the other end of the call.
It might be a good idea to let Fongo Support know what numbers are likely culprits so that they can try to narrow down which carrier (Rogers, for example) is involved in the problem.
I can't confirm there's a problem without calling the same numbers, and I don't want to do that unless the problem can be reproduced by calling a common business number (Shoppers, CIBC, etc.).
I have been calling businesses and Telus carrier numbers today. I haven't had issues with calls not ending after I hang up.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
-
Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up
by newaira » 10/31/2021
So I switched to voip4.freephoneline.ca:6060 and when I hang up the voip, it ends the call successfully. Here is the SIP log for such a call (I've removed my phone numbers and IPs):
However, I'm still having the issue where if I call from FPL voip and the phone starts ringing on the other side, if I hang up, the phone keeps ringing on the other side. If the other side does pick up, there is no sound and it's just stays like that for what seems like 11 minutes. My main concern here is that my family members are calling Europe and if no one picks up after a few rings, they hang up. However the voicemail on the other side picks up the call after hanging up, and for 11 minutes we get charged every time for the line staying open. Here is the call log where we hang up before the other side picks up:
I've also included my Grandstream HT802 and Smart/RG configuration screenshots for reference:
- Code: Select all
HT802 --- 2021-10-30 15:01:36.592 SENDING TO voip4.freephoneline.ca:6060
INVITE sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK195183310;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
To: <sip:4165555555@voip4.freephoneline.ca:6060>
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 260 INVITE
Contact: "Lastname" <sip:12895555555@192.168.0.12:13409>
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Privacy: none
P-Preferred-Identity: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=E8-2C-6D-69-AC-25
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=C0-74-AD-42-C4-8F
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 275
v=0
o=12895555555 8000 8000 IN IP4 192.168.0.12
s=SIP Call
c=IN IP4 192.168.0.12
t=0 0
m=audio 38854 RTP/AVP 0 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
HT802 --- 2021-10-30 15:01:36.619 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK195183310;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-8B4.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 260 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm="sip-12.FSFEN-wsFongo",nonce="1635606096:71da530fb097783b2a77b85ecc3238b06a153d4c"
Content-Length: 0
HT802 --- 2021-10-30 15:01:36.630 SENDING TO voip4.freephoneline.ca:6060
ACK sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK195183310;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-8B4.i
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 260 ACK
Content-Length: 0
HT802 --- 2021-10-30 15:01:36.642 SENDING TO voip4.freephoneline.ca:6060
INVITE sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1592846356;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
To: <sip:4165555555@voip4.freephoneline.ca:6060>
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 261 INVITE
Contact: "Lastname" <sip:12895555555@192.168.0.12:13409>
Authorization: Digest username="12895555555", realm="sip-12.FSFEN-wsFongo", nonce="1635606096:71da530fb097783b2a77b85ecc3238b06a153d4c", uri="sip:4165555555@voip4.freephoneline.ca:6060", response="4fc2db5b6a8e6d302a2e95069e7eb00e", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Privacy: none
P-Preferred-Identity: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=E8-2C-6D-69-AC-25
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=C0-74-AD-42-C4-8F
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 275
v=0
o=12895555555 8000 8000 IN IP4 192.168.0.12
s=SIP Call
c=IN IP4 192.168.0.12
t=0 0
m=audio 38854 RTP/AVP 0 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
HT802 --- 2021-10-30 15:01:37.141 SENDING TO voip4.freephoneline.ca:6060
INVITE sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1592846356;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
To: <sip:4165555555@voip4.freephoneline.ca:6060>
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 261 INVITE
Contact: "Lastname" <sip:12895555555@192.168.0.12:13409>
Authorization: Digest username="12895555555", realm="sip-12.FSFEN-wsFongo", nonce="1635606096:71da530fb097783b2a77b85ecc3238b06a153d4c", uri="sip:4165555555@voip4.freephoneline.ca:6060", response="4fc2db5b6a8e6d302a2e95069e7eb00e", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Privacy: none
P-Preferred-Identity: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=E8-2C-6D-69-AC-25
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=C0-74-AD-42-C4-8F
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 275
v=0
o=12895555555 8000 8000 IN IP4 192.168.0.12
s=SIP Call
c=IN IP4 192.168.0.12
t=0 0
m=audio 38854 RTP/AVP 0 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
HT802 --- 2021-10-30 15:01:38.042 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK1592846356;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
Contact: sip:208.85.218.147:5070
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-5EK.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 261 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Server: PortaSIP
Content-Length: 0
HT802 --- 2021-10-30 15:01:38.752 SENDING TO voip4.freephoneline.ca:6060
NOTIFY sip:voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1009956075;rport
From: <sip:12895555555@voip4.freephoneline.ca:6060>;tag=21740758
To: <sip:voip4.freephoneline.ca:6060>
Call-ID: 1069870145-13409-29@BJC.BGI.A.BC
CSeq: 270 NOTIFY
Contact: <sip:12895555555@192.168.0.12:13409>
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
HT802 --- 2021-10-30 15:01:38.768 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1009956075;rport=13409;received=100.100.100.100
From: <sip:12895555555@voip4.freephoneline.ca:6060>;tag=21740758
To: <sip:voip4.freephoneline.ca:6060>;tag=224edcece75971e271cb292defd4983a.98c1
Call-ID: 1069870145-13409-29@BJC.BGI.A.BC
CSeq: 270 NOTIFY
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
HT802 --- 2021-10-30 15:01:44.122 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK195183310;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-8B4.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 260 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm="sip-12.FSFEN-wsFongo",nonce="1635606096:71da530fb097783b2a77b85ecc3238b06a153d4c"
Content-Length: 0
HT802 --- 2021-10-30 15:01:44.129 SENDING TO voip4.freephoneline.ca:6060
ACK sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK195183310;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-8B4.i
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 260 ACK
Content-Length: 0
HT802 --- 2021-10-30 15:01:44.694 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK1592846356;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
Contact: sip:208.85.218.147:5070
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-5EK.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 261 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Content-Type: application/sdp
Server: PortaSIP
H323-credit-time: 14400
Content-Length: 174
v=0
o=PortaSIP 1883377761742262982 1 IN IP4 208.85.218.147
s=-
t=0 0
m=audio 47638 RTP/AVP 0 101
c=IN IP4 208.85.218.147
a=rtpmap:101 telephone-event/8000
a=ptime:20
HT802 --- 2021-10-30 15:01:44.729 SENDING TO 162.213.111.21:6060
ACK sip:208.85.218.147:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK862811350;rport
Route: <sip:4165555555@162.213.111.21:6060;lr=on>
Route: <sip:208.85.218.148;lr;ep>
Route: <sip:208.85.218.145;lr;ep>
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-5EK.i
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 261 ACK
Contact: <sip:12895555555@192.168.0.12:13409>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT802 1.0.31.1
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
HT802 --- 2021-10-30 15:01:48.123 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK195183310;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-8B4.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 260 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm="sip-12.FSFEN-wsFongo",nonce="1635606096:71da530fb097783b2a77b85ecc3238b06a153d4c"
Content-Length: 0
HT802 --- 2021-10-30 15:01:48.131 SENDING TO voip4.freephoneline.ca:6060
ACK sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK195183310;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-8B4.i
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 260 ACK
Content-Length: 0
HT802 --- 2021-10-30 15:01:49.797 SENDING TO 162.213.111.21:6060
BYE sip:208.85.218.147:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK365016651;rport
Route: <sip:4165555555@162.213.111.21:6060;lr=on>
Route: <sip:208.85.218.148;lr;ep>
Route: <sip:208.85.218.145;lr;ep>
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-5EK.i
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 262 BYE
Contact: <sip:12895555555@192.168.0.12:13409>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT802 1.0.31.1
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
HT802 --- 2021-10-30 15:01:49.854 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK365016651;rport=13409
Record-Route: <sip:162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-5EK.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 262 BYE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Server: PortaSIP
Content-Length: 0
HT802 --- 2021-10-30 15:01:52.135 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK195183310;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-8B4.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 260 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm="sip-12.FSFEN-wsFongo",nonce="1635606096:71da530fb097783b2a77b85ecc3238b06a153d4c"
Content-Length: 0
HT802 --- 2021-10-30 15:01:52.142 SENDING TO voip4.freephoneline.ca:6060
ACK sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK195183310;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-8B4.i
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 260 ACK
Content-Length: 0
HT802 --- 2021-10-30 15:01:56.125 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK195183310;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-8B4.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 260 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm="sip-12.FSFEN-wsFongo",nonce="1635606096:71da530fb097783b2a77b85ecc3238b06a153d4c"
Content-Length: 0
HT802 --- 2021-10-30 15:01:56.132 SENDING TO voip4.freephoneline.ca:6060
ACK sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK195183310;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-8B4.i
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 260 ACK
Content-Length: 0
However, I'm still having the issue where if I call from FPL voip and the phone starts ringing on the other side, if I hang up, the phone keeps ringing on the other side. If the other side does pick up, there is no sound and it's just stays like that for what seems like 11 minutes. My main concern here is that my family members are calling Europe and if no one picks up after a few rings, they hang up. However the voicemail on the other side picks up the call after hanging up, and for 11 minutes we get charged every time for the line staying open. Here is the call log where we hang up before the other side picks up:
- Code: Select all
HT802 --- 2021-10-30 15:00:16.067 SENDING TO voip4.freephoneline.ca:6060
INVITE sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1262813457;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 INVITE
Contact: "Lastname" <sip:12895555555@192.168.0.12:13409>
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Privacy: none
P-Preferred-Identity: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=E8-2C-6D-69-AC-25
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=C0-74-AD-42-C4-8F
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 275
v=0
o=12895555555 8000 8000 IN IP4 192.168.0.12
s=SIP Call
c=IN IP4 192.168.0.12
t=0 0
m=audio 39654 RTP/AVP 0 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
HT802 --- 2021-10-30 15:00:16.103 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK1262813457;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm="sip-12.FSFEN-wsFongo",nonce="1635606016:e1dafbe08091b8666c474965af9947227cf52748"
Content-Length: 0
HT802 --- 2021-10-30 15:00:16.112 SENDING TO voip4.freephoneline.ca:6060
ACK sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1262813457;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 ACK
Content-Length: 0
HT802 --- 2021-10-30 15:00:16.121 SENDING TO voip4.freephoneline.ca:6060
INVITE sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK655153266;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 INVITE
Contact: "Lastname" <sip:12895555555@192.168.0.12:13409>
Authorization: Digest username="12895555555", realm="sip-12.FSFEN-wsFongo", nonce="1635606016:e1dafbe08091b8666c474965af9947227cf52748", uri="sip:4165555555@voip4.freephoneline.ca:6060", response="ceb8612088d6add0805b0700eccc108b", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Privacy: none
P-Preferred-Identity: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=E8-2C-6D-69-AC-25
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=C0-74-AD-42-C4-8F
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 275
v=0
o=12895555555 8000 8000 IN IP4 192.168.0.12
s=SIP Call
c=IN IP4 192.168.0.12
t=0 0
m=audio 39654 RTP/AVP 0 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
HT802 --- 2021-10-30 15:00:16.619 SENDING TO voip4.freephoneline.ca:6060
INVITE sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK655153266;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 INVITE
Contact: "Lastname" <sip:12895555555@192.168.0.12:13409>
Authorization: Digest username="12895555555", realm="sip-12.FSFEN-wsFongo", nonce="1635606016:e1dafbe08091b8666c474965af9947227cf52748", uri="sip:4165555555@voip4.freephoneline.ca:6060", response="ceb8612088d6add0805b0700eccc108b", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Privacy: none
P-Preferred-Identity: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=E8-2C-6D-69-AC-25
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=C0-74-AD-42-C4-8F
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 275
v=0
o=12895555555 8000 8000 IN IP4 192.168.0.12
s=SIP Call
c=IN IP4 192.168.0.12
t=0 0
m=audio 39654 RTP/AVP 0 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
HT802 --- 2021-10-30 15:00:17.188 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK655153266;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
Contact: sip:208.85.218.147:5070
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-zhQ.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Server: PortaSIP
Content-Length: 0
HT802 --- 2021-10-30 15:00:18.619 SENDING TO voip4.freephoneline.ca:6060
NOTIFY sip:voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1804093993;rport
From: <sip:12895555555@voip4.freephoneline.ca:6060>;tag=585552616
To: <sip:voip4.freephoneline.ca:6060>
Call-ID: 1094899600-13409-24@BJC.BGI.A.BC
CSeq: 220 NOTIFY
Contact: <sip:12895555555@192.168.0.12:13409>
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
HT802 --- 2021-10-30 15:00:18.634 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1804093993;rport=13409;received=100.100.100.100
From: <sip:12895555555@voip4.freephoneline.ca:6060>;tag=585552616
To: <sip:voip4.freephoneline.ca:6060>;tag=224edcece75971e271cb292defd4983a.0221
Call-ID: 1094899600-13409-24@BJC.BGI.A.BC
CSeq: 220 NOTIFY
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
HT802 --- 2021-10-30 15:00:23.701 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK1262813457;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm="sip-12.FSFEN-wsFongo",nonce="1635606016:e1dafbe08091b8666c474965af9947227cf52748"
Content-Length: 0
HT802 --- 2021-10-30 15:00:23.707 SENDING TO voip4.freephoneline.ca:6060
ACK sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1262813457;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 ACK
Content-Length: 0
HT802 --- 2021-10-30 15:00:27.703 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK1262813457;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm="sip-12.FSFEN-wsFongo",nonce="1635606016:e1dafbe08091b8666c474965af9947227cf52748"
Content-Length: 0
HT802 --- 2021-10-30 15:00:27.709 SENDING TO voip4.freephoneline.ca:6060
ACK sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1262813457;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 ACK
Content-Length: 0
HT802 --- 2021-10-30 15:00:28.492 SENDING TO voip4.freephoneline.ca:6060
CANCEL sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK655153266;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 CANCEL
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Content-Length: 0
HT802 --- 2021-10-30 15:00:28.507 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 200 canceling
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK655153266;rport=13409;received=100.100.100.100
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=d7837ce6bbd631122d10546eb75bb4cf-4821
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 CANCEL
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
HT802 --- 2021-10-30 15:00:31.716 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK1262813457;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm="sip-12.FSFEN-wsFongo",nonce="1635606016:e1dafbe08091b8666c474965af9947227cf52748"
Content-Length: 0
HT802 --- 2021-10-30 15:00:31.723 SENDING TO voip4.freephoneline.ca:6060
ACK sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1262813457;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 ACK
Content-Length: 0
HT802 --- 2021-10-30 15:00:33.821 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK655153266;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
Contact: sip:208.85.218.147:5070
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-zhQ.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Content-Type: application/sdp
Server: PortaSIP
H323-credit-time: 14400
Content-Length: 174
v=0
o=PortaSIP 3000972168979591419 1 IN IP4 208.85.218.147
s=-
t=0 0
m=audio 45874 RTP/AVP 0 101
c=IN IP4 208.85.218.147
a=rtpmap:101 telephone-event/8000
a=ptime:20
HT802 --- 2021-10-30 15:00:33.834 SENDING TO 162.213.111.21:6060
ACK sip:208.85.218.147:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1133103701;rport
Route: <sip:4165555555@162.213.111.21:6060;lr=on>
Route: <sip:208.85.218.148;lr;ep>
Route: <sip:208.85.218.145;lr;ep>
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-zhQ.i
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 ACK
Contact: <sip:12895555555@192.168.0.12:13409>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT802 1.0.31.1
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
HT802 --- 2021-10-30 15:00:35.715 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK1262813457;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm="sip-12.FSFEN-wsFongo",nonce="1635606016:e1dafbe08091b8666c474965af9947227cf52748"
Content-Length: 0
HT802 --- 2021-10-30 15:00:35.721 SENDING TO voip4.freephoneline.ca:6060
ACK sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1262813457;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 ACK
Content-Length: 0
I've also included my Grandstream HT802 and Smart/RG configuration screenshots for reference:
- newaira
- Just Passing Thru
- Posts: 4
- Joined: 10/29/2021
- SIP Device Name: Grandstream HT802
- Firmware Version: 1.0.31.1
- ISP Name: Acanac Cable
- Computer OS: Windows
- Router: SMART/RG SR808ac
Re: Call not ending after hanging up
by Liptonbrisk » 10/31/2021
newaira wrote:So I switched to voip4.freephoneline.ca:6060 and when I hang up the voip, it ends the call successfully. Here is the SIP log for such a call (I've removed my phone numbers and IPs):
- Code: Select all
HT802 --- 2021-10-30 15:01:49.797 SENDING TO 162.213.111.21:6060
BYE sip:208.85.218.147:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK365016651;rport
Route: <sip:4165555555@162.213.111.21:6060;lr=on>
Route: <sip:208.85.218.148;lr;ep>
Route: <sip:208.85.218.145;lr;ep>
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-5EK.i
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 262 BYE
Contact: <sip:12895555555@192.168.0.12:13409>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT802 1.0.31.1
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
HT802 --- 2021-10-30 15:01:49.854 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK365016651;rport=13409
Record-Route: <sip:162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=ri7M8w+y0BGS-5EK.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=878506654
Call-ID: 559900536-13409-28@BJC.BGI.A.BC
CSeq: 262 BYE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Server: PortaSIP
Content-Length: 0
Yes, that sequence is what you want. BYE is sent. 200 OK is received. Call ends.
I thought that was what iceonu was referring to, but it wasn't.
It's a different sequence that occurs when hanging up before the party at the other end answers. CANCEL is sent instead. The response is supposed to be 200 OK, and then 487 is supposed to be sent back.
However, I'm still having the issue where if I call from FPL voip and the phone starts ringing on the other side, if I hang up, the phone keeps ringing on the other side. If the other side does pick up, there is no sound and it's just stays like that for what seems like 11 minutes. My main concern here is that my family members are calling Europe and if no one picks up after a few rings, they hang up. However the voicemail on the other side picks up the call after hanging up, and for 11 minutes we get charged every time for the line staying open. Here is the call log where we hang up before the other side picks up:
- Code: Select all
HT802 --- 2021-10-30 15:00:17.188 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK655153266;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
Contact: sip:208.85.218.147:5070
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-zhQ.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Server: PortaSIP
Content-Length: 0
Okay, so ringing starts here.
- Code: Select all
HT802 --- 2021-10-30 15:00:28.492 SENDING TO voip4.freephoneline.ca:6060
CANCEL sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK655153266;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 CANCEL
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Content-Length: 0
HT802 --- 2021-10-30 15:00:28.507 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 200 canceling
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK655153266;rport=13409;received=100.100.100.100
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=d7837ce6bbd631122d10546eb75bb4cf-4821
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 CANCEL
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
You hang up here. CANCEL is sent. CANCEL occurs when the call is hung up before the other side answers.
I don't see a proper response to this or 487 coming back.
I can reproduce the problem. SIP device does not indicate call is active, but FPL keeps it open.
This is what should happen:
Source: https://thanhloi2603.wordpress.com/2017 ... -messages/
I going to try contacting someone about this. . .
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
-
Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up [before other end answe
by Liptonbrisk » 10/31/2021
- Testing to Fongo Mobile and Telus Mobility carrier numbers (I doubt it matters what number I call)
I can reproduce this problem using an OBi202. I've tried all 3 FPL proxy servers: voip.freephoneline.ca, voip2.freephoneline.ca, and voip4.freephoneline.ca:6060.
I can also reproduce this problem on Acrobits Groundwire SIP app on both Wi-Fi and 5G.
However, the Freephoneline Desktop app works properly for cancelling calls while on the the same LAN as the OBi202.
It should be noted that the FPL desktop app uses 208.85.218.148, which is voip3.freephoneline.ca. We're not allowed to use voip3.freephoneline.ca without risk of having our accounts suspended:
https://support.freephoneline.ca/hc/en- ... redentials.
"Use of Fongo SIP Servers that are not listed in this document will result in your account being suspended."
I can reproduce this problem using an OBi202. I've tried all 3 FPL proxy servers: voip.freephoneline.ca, voip2.freephoneline.ca, and voip4.freephoneline.ca:6060.
I can also reproduce this problem on Acrobits Groundwire SIP app on both Wi-Fi and 5G.
However, the Freephoneline Desktop app works properly for cancelling calls while on the the same LAN as the OBi202.
It should be noted that the FPL desktop app uses 208.85.218.148, which is voip3.freephoneline.ca. We're not allowed to use voip3.freephoneline.ca without risk of having our accounts suspended:
https://support.freephoneline.ca/hc/en- ... redentials.
"Use of Fongo SIP Servers that are not listed in this document will result in your account being suspended."
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
-
Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up [before other end answe
by Liptonbrisk » 10/31/2021
Liptonbrisk wrote:
However, the Freephoneline Desktop app works properly for cancelling calls while on the the same LAN as the OBi202.
It should be noted that the FPL desktop app uses 208.85.218.148, which is voip3.freephoneline.ca. We're not allowed to use voip3.freephoneline.ca without risk of having our accounts suspended:
https://support.freephoneline.ca/hc/en- ... redentials.
"Use of Fongo SIP Servers that are not listed in this document will result in your account being suspended."
Welp. . .
We've all been blocked from using our VoIP Unlock Key Credentials with voip3.freephoneline.ca:
I get "Register Failed: 403 Forbidden (server=208.85.218.148:5060; retry in 120s)".
That's news to me.
I wanted to see if I could reproduce this issue on my SIP devices using voip3.freephoneline.ca.
Unfortunately, I have no permitted method of testing voip3.freephoneline.ca other than to use the FPL desktop application. Well, there is another way, but I don't want to end up getting one of my accounts banned.
Anyway, the FPL desktop application works properly for cancelling calls before they're answered, but that's likely because it's not using voip.freephoneline.ca or voip2.freephoneline.ca.
I suspect this issue needs to be addressed by the third party switch vendor that Fongo uses.
I see "Server: PortaSIP" in the logs.
https://www.portaone.com/resources/PortaSIP.pdf
I guess https://www.portaone.com/ is Fongo's switch vendor.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
-
Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up [before other end answe
by Liptonbrisk » 10/31/2021
There's not much more I can do at this point.
If you're losing World Credits over this situation, choose "World Credits Inquiry" for the final issue type.
The escalation process is located at https://support.fongo.com/hc/en-us/arti ... -Complaint
If you're losing World Credits over this situation, choose "World Credits Inquiry" for the final issue type.
The escalation process is located at https://support.fongo.com/hc/en-us/arti ... -Complaint
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
-
Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up [before other end answe
by Liptonbrisk » 11/01/2021
Liptonbrisk wrote:
I suspect this issue needs to be addressed by the third party switch vendor that Fongo uses.
I see "Server: PortaSIP" in the logs.
https://www.portaone.com/resources/PortaSIP.pdf
I guess https://www.portaone.com/ is Fongo's switch vendor.
Just to update, the problem has been reported to the switch vendor. I don't know what the turn around time is for a fix.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
-
Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up [before other end answe
by newaira » 11/02/2021
Liptonbrisk wrote:Just to update, the problem has been reported to the switch vendor. I don't know what the turn around time is for a fix.
Thank you for really digging deep into this issue! I'll submit a ticket to FPL and hopefully it will be addressed...
- newaira
- Just Passing Thru
- Posts: 4
- Joined: 10/29/2021
- SIP Device Name: Grandstream HT802
- Firmware Version: 1.0.31.1
- ISP Name: Acanac Cable
- Computer OS: Windows
- Router: SMART/RG SR808ac
Re: Call not ending after hanging up [before other end answe
by Liptonbrisk » 11/02/2021
newaira wrote:
Thank you for really digging deep into this issue!
You're welcome!
I'll submit a ticket to FPL and hopefully it will be addressed...
I'm not sure there's much they can do at this point since the issue has already been reported to the switch vendor. However, if you have lost World Credits as a result of this problem, that is something they can address.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
-
Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up [before other end answe
by iceonu » 11/04/2021
Thanks Libtonbrisk for all your replies. I haven't had a chance to check the forum until now. I implemented the changes to my settings per your suggestions. Still having the same issue. Whats interesting is I don't believe its related to the Grandstream device. I setup my old Linksys PAP2T and discovered the same thing was happening. So far I've only been dialling out to a Rogers #, so don't know if this problem occurs calling other providers. I'll attach a PCAP call flow.
- iceonu
- Just Passing Thru
- Posts: 13
- Joined: 06/04/2013
- SIP Device Name: Grandstream GXP2135
- ISP Name: Telus
- Computer OS: OS11.6.1
- Router: Ubiquiti UDM-Pro
Re: Call not ending after hanging up [before other end answe
by iceonu » 11/04/2021
Call flow.pdf- Call Flow
- (26.66 KiB) Downloaded 159 times
- iceonu
- Just Passing Thru
- Posts: 13
- Joined: 06/04/2013
- SIP Device Name: Grandstream GXP2135
- ISP Name: Telus
- Computer OS: OS11.6.1
- Router: Ubiquiti UDM-Pro
Re: Call not ending after hanging up [before other end answe
by Liptonbrisk » 11/04/2021
iceonu wrote: Whats interesting is I don't believe its related to the Grandstream device.
It's not. It's an issue that only FPL's switch vendor, https://www.portaone.com/, can address, and the issue has been reported to them. When you originally posted, I thought you were hanging up after the other end answered. That process sends BYE--not CANCEL. If you hang up after the other end answers (BYE), the call will disconnect properly. As such, I provided steps that I thought would help you based on that working scenario.
CANCEL (hanging up before the other end connects) is not working properly. I changed the subject heading of this thread for clarification. The instructions in my initial response will not help resolve this issue.
I've confirmed the issue here:
viewtopic.php?f=8&t=20261#p79211
and
viewtopic.php?f=8&t=20261#p79212
and
viewtopic.php?f=8&t=20261#p79214
and
viewtopic.php?f=8&t=20261#p79216
So far I've only been dialling out to a Rogers #, so don't know if this problem occurs calling other providers.
It happens with everything, except when using the FPL desktop application (only because it uses voip3.freephoneline.ca, which VoIP unlock key users can't use). Unfortunately, I have no clue when the issue may be addressed. There's nothing further I can do.
If you happen to be losing World Credits over this situation, choose "World Credits Inquiry" for the final issue type when submitting a ticket: https://support.fongo.com/hc/en-us/requests/new.
The escalation process is located at https://support.fongo.com/hc/en-us/arti ... -Complaint.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
-
Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up [before other end answe
by iceonu » 11/04/2021
Looking at the capture closer it reveals that the CANCEL message from the VOIP phone includes a different Call-ID header than the INVITE did. 
And this is a problem because :
...and Ive quoted from: https://andrewjprokop.wordpress.com/201 ... ia-header/
"The branch parameter must be unique across space and time for all requests sent by a user agent. The exceptions to this rule are CANCEL and ACK for non-2xx responses. A CANCEL will have the same branch parameter as the request it cancels. An ACK for a non-2xx response will have the same branch parameter as the INVITE whose response it acknowledges".
If using the FPL desktop app, Im going to guess the Call-ID header for the invite and cancel match.
And this is a problem because : ...and Ive quoted from: https://andrewjprokop.wordpress.com/201 ... ia-header/
"The branch parameter must be unique across space and time for all requests sent by a user agent. The exceptions to this rule are CANCEL and ACK for non-2xx responses. A CANCEL will have the same branch parameter as the request it cancels. An ACK for a non-2xx response will have the same branch parameter as the INVITE whose response it acknowledges".
If using the FPL desktop app, Im going to guess the Call-ID header for the invite and cancel match.
- iceonu
- Just Passing Thru
- Posts: 13
- Joined: 06/04/2013
- SIP Device Name: Grandstream GXP2135
- ISP Name: Telus
- Computer OS: OS11.6.1
- Router: Ubiquiti UDM-Pro
Re: Call not ending after hanging up [before other end answe
by Liptonbrisk » 11/05/2021
iceonu wrote: Looking at the capture closer it reveals that the CANCEL message from the VOIP phone includes a different Call-ID header than the INVITE did
It doesn't in my syslogs.
CALL-ID remains the same for the INVITE and CANCEL requests.
newaira's appears to be edited (632916890-13409-23@BJC.BGI.A.BC).
However, the same random, generated number for CALL-ID appears in my logs for both INVITE and CANCEL, and that's the same for newaira as well.
The random CALL-ID numbers differ for NOTIFY and INVITE. That doesn't matter.
A CANCEL will have the same branch parameter as the request it cancels.
Branch parameter, in the example that newaira provided, is z9hG4bK655153266 for the CANCEL request. It's also z9hG4bK655153266 for the INVITE.
They remain the same for me as well.
There's no 487 received, which is required to end the SIP session.
viewtopic.php?f=8&t=20261&p=79224#p79210
- Code: Select all
HT802 --- 2021-10-30 15:00:16.121 SENDING TO voip4.freephoneline.ca:6060
INVITE sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK655153266;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 INVITE
Contact: "Lastname" <sip:12895555555@192.168.0.12:13409>
Authorization: Digest username="12895555555", realm="sip-12.FSFEN-wsFongo", nonce="1635606016:e1dafbe08091b8666c474965af9947227cf52748", uri="sip:4165555555@voip4.freephoneline.ca:6060", response="ceb8612088d6add0805b0700eccc108b", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Privacy: none
P-Preferred-Identity: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=E8-2C-6D-69-AC-25
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=C0-74-AD-42-C4-8F
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 275
v=0
o=12895555555 8000 8000 IN IP4 192.168.0.12
s=SIP Call
c=IN IP4 192.168.0.12
t=0 0
m=audio 39654 RTP/AVP 0 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
HT802 --- 2021-10-30 15:00:16.619 SENDING TO voip4.freephoneline.ca:6060
INVITE sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK655153266;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 INVITE
Contact: "Lastname" <sip:12895555555@192.168.0.12:13409>
Authorization: Digest username="12895555555", realm="sip-12.FSFEN-wsFongo", nonce="1635606016:e1dafbe08091b8666c474965af9947227cf52748", uri="sip:4165555555@voip4.freephoneline.ca:6060", response="ceb8612088d6add0805b0700eccc108b", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Privacy: none
P-Preferred-Identity: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=E8-2C-6D-69-AC-25
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=C0-74-AD-42-C4-8F
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 275
v=0
o=12895555555 8000 8000 IN IP4 192.168.0.12
s=SIP Call
c=IN IP4 192.168.0.12
t=0 0
m=audio 39654 RTP/AVP 0 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
HT802 --- 2021-10-30 15:00:17.188 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK655153266;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
Contact: sip:208.85.218.147:5070
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-zhQ.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Server: PortaSIP
Content-Length: 0
HT802 --- 2021-10-30 15:00:18.619 SENDING TO voip4.freephoneline.ca:6060
NOTIFY sip:voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1804093993;rport
From: <sip:12895555555@voip4.freephoneline.ca:6060>;tag=585552616
To: <sip:voip4.freephoneline.ca:6060>
Call-ID: 1094899600-13409-24@BJC.BGI.A.BC
CSeq: 220 NOTIFY
Contact: <sip:12895555555@192.168.0.12:13409>
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
HT802 --- 2021-10-30 15:00:18.634 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1804093993;rport=13409;received=100.100.100.100
From: <sip:12895555555@voip4.freephoneline.ca:6060>;tag=585552616
To: <sip:voip4.freephoneline.ca:6060>;tag=224edcece75971e271cb292defd4983a.0221
Call-ID: 1094899600-13409-24@BJC.BGI.A.BC
CSeq: 220 NOTIFY
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
HT802 --- 2021-10-30 15:00:23.701 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK1262813457;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm="sip-12.FSFEN-wsFongo",nonce="1635606016:e1dafbe08091b8666c474965af9947227cf52748"
Content-Length: 0
HT802 --- 2021-10-30 15:00:23.707 SENDING TO voip4.freephoneline.ca:6060
ACK sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1262813457;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 ACK
Content-Length: 0
HT802 --- 2021-10-30 15:00:27.703 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK1262813457;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm="sip-12.FSFEN-wsFongo",nonce="1635606016:e1dafbe08091b8666c474965af9947227cf52748"
Content-Length: 0
HT802 --- 2021-10-30 15:00:27.709 SENDING TO voip4.freephoneline.ca:6060
ACK sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1262813457;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 ACK
Content-Length: 0
HT802 --- 2021-10-30 15:00:28.492 SENDING TO voip4.freephoneline.ca:6060
CANCEL sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK655153266;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 CANCEL
Max-Forwards: 70
User-Agent: Grandstream HT802 1.0.31.1
Content-Length: 0
HT802 --- 2021-10-30 15:00:28.507 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 200 canceling
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK655153266;rport=13409;received=100.100.100.100
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=d7837ce6bbd631122d10546eb75bb4cf-4821
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 CANCEL
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
HT802 --- 2021-10-30 15:00:31.716 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK1262813457;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: PortaSIP
WWW-Authenticate: Digest realm="sip-12.FSFEN-wsFongo",nonce="1635606016:e1dafbe08091b8666c474965af9947227cf52748"
Content-Length: 0
HT802 --- 2021-10-30 15:00:31.723 SENDING TO voip4.freephoneline.ca:6060
ACK sip:4165555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:13409;branch=z9hG4bK1262813457;rport
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-4a9.i
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 210 ACK
Content-Length: 0
HT802 --- 2021-10-30 15:00:33.821 RECEIVING FROM 162.213.111.21:6060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.12:13409;received=100.100.100.100;branch=z9hG4bK655153266;rport=13409
Record-Route: <sip:208.85.218.145;lr;ep>
Record-Route: <sip:208.85.218.148;lr;ep>
Record-Route: <sip:4165555555@162.213.111.21:6060;lr=on>
Contact: sip:208.85.218.147:5070
To: <sip:4165555555@voip4.freephoneline.ca:6060>;tag=oyvO8gu53RqS-zhQ.i
From: "Lastname" <sip:12895555555@voip4.freephoneline.ca:6060>;tag=641450642
Call-ID: 632916890-13409-23@BJC.BGI.A.BC
CSeq: 211 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Content-Type: application/sdp
Server: PortaSIP
H323-credit-time: 14400
Content-Length: 174
v=0
o=PortaSIP 3000972168979591419 1 IN IP4 208.85.218.147
s=-
t=0 0
m=audio 45874 RTP/AVP 0 101
c=IN IP4 208.85.218.147
a=rtpmap:101 telephone-event/8000
a=ptime:20
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
-
Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up [before other end answe
by Liptonbrisk » 11/05/2021
Just to update, FPL's switch used for VoIP unlock key users is in "maintenance mode, preparing for user migration to a newer version of the software and new hardware. To facilitate this the [switch] vendor added another proxy in the path." There is no known date yet for the migration or maintenance to be completed.
Consequently, there's nothing users can do but wait for a fix.
It seems logical to conclude the other proxy is presenting an issue with respect to 487 not being received.
Consequently, there's nothing users can do but wait for a fix.
It seems logical to conclude the other proxy is presenting an issue with respect to 487 not being received.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
-
Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up [before other end answe
by iceonu » 11/05/2021
this capture was what I was seeing and then lead me to think that the SBC (session border controller) is messing with it.
Is this issue affecting all FPL users?
INVITE sip:5555555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.235:5060;branch=z9hG4bK990824362;rport
From: "VoIP phone" <sip:5555555555@voip4.freephoneline.ca:6060>;tag=1079506320
To: <sip:5555555555@voip4.freephoneline.ca:6060>
Call-ID: 76328505-5060-3367@BJC.BGI.D.CDF
CSeq: 33660 INVITE
Contact: "VoIP phone" <sip:5555555555@192.168.3.235:5060>
Max-Forwards: 70
User-Agent: Grandstream GXP2135 1.0.11.39
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 267
SIP/2.0 200 canceling
Via: SIP/2.0/UDP 192.168.3.235:5060;branch=z9hG4bK1476503250;rport=5060;received=***
From: "VoIP phone" <sip:5555555555@voip4.freephoneline.ca:6060>;tag=1079506320
To: <sip:555555555@voip4.freephoneline.ca:6060>;tag=d7837ce6bbd631122d10546eb75bb4cf-8522
Call-ID: 76328505-5060-3367@BJC.BGI.D.CDF
CSeq: 33661 CANCEL
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
v=0
o=14032751718 8000 8000 IN IP4 192.168.3.235
s=SIP Call
c=IN IP4 192.168.3.235
t=0 0
m=audio 5004 RTP/AVP 0 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Is this issue affecting all FPL users?
INVITE sip:5555555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.235:5060;branch=z9hG4bK990824362;rport
From: "VoIP phone" <sip:5555555555@voip4.freephoneline.ca:6060>;tag=1079506320
To: <sip:5555555555@voip4.freephoneline.ca:6060>
Call-ID: 76328505-5060-3367@BJC.BGI.D.CDF
CSeq: 33660 INVITE
Contact: "VoIP phone" <sip:5555555555@192.168.3.235:5060>
Max-Forwards: 70
User-Agent: Grandstream GXP2135 1.0.11.39
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 267
SIP/2.0 200 canceling
Via: SIP/2.0/UDP 192.168.3.235:5060;branch=z9hG4bK1476503250;rport=5060;received=***
From: "VoIP phone" <sip:5555555555@voip4.freephoneline.ca:6060>;tag=1079506320
To: <sip:555555555@voip4.freephoneline.ca:6060>;tag=d7837ce6bbd631122d10546eb75bb4cf-8522
Call-ID: 76328505-5060-3367@BJC.BGI.D.CDF
CSeq: 33661 CANCEL
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
v=0
o=14032751718 8000 8000 IN IP4 192.168.3.235
s=SIP Call
c=IN IP4 192.168.3.235
t=0 0
m=audio 5004 RTP/AVP 0 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
- iceonu
- Just Passing Thru
- Posts: 13
- Joined: 06/04/2013
- SIP Device Name: Grandstream GXP2135
- ISP Name: Telus
- Computer OS: OS11.6.1
- Router: Ubiquiti UDM-Pro
Re: Call not ending after hanging up [before other end answe
by Liptonbrisk » 11/05/2021
iceonu wrote:this capture was what I was seeing and then lead me to think
You haven't provided the corresponding INVITE with "200 canceling" below. The easiest thing to look at is that CSeq differs, and they should match.
- Code: Select all
INVITE sip:5555555555@voip4.freephoneline.ca:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.235:5060;branch=z9hG4bK990824362;rport
From: "VoIP phone" <sip:5555555555@voip4.freephoneline.ca:6060>;tag=1079506320
To: <sip:5555555555@voip4.freephoneline.ca:6060>
Call-ID: 76328505-5060-3367@BJC.BGI.D.CDF
CSeq: 33660 INVITE
Contact: "VoIP phone" <sip:5555555555@192.168.3.235:5060>
Max-Forwards: 70
User-Agent: Grandstream GXP2135 1.0.11.39
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 267
SIP/2.0 200 canceling
Via: SIP/2.0/UDP 192.168.3.235:5060;branch=z9hG4bK1476503250;rport=5060;received=***
From: "VoIP phone" <sip:5555555555@voip4.freephoneline.ca:6060>;tag=1079506320
To: <sip:555555555@voip4.freephoneline.ca:6060>;tag=d7837ce6bbd631122d10546eb75bb4cf-8522
Call-ID: 76328505-5060-3367@BJC.BGI.D.CDF
CSeq: 33661 CANCEL
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
In the example you provided, before "SIP/2.0 200 canceling", you should see a section including "CANCEL sip:5555555555@voip4.freephoneline.ca:6060 SIP/2.0" with CSeq 33661.
Before that CANCEL you should see "INVITE sip:5555555555@voip4.freephoneline.ca:6060" (you likely scrolled up too far from where you found "200 canceling") with CSeq 33361.
That's the INVITE that matters.
And if you scroll up a little further in your log, you should also see why the INVITE with CSeq 33600 doesn't matter because it was later challenged by "SIP/2.0 401 Unauthorized" with "CSeq: 33600 INVITE".
Regardless, the problem is we can only see what we send and receive.
We have no idea whether 487 is sent to us. All we know is that we don't receive it.
Edit: I’ve been informed that proxy is sending 481 instead, and that the problem is also affecting Fongo Home Phone users.
Is this issue affecting all FPL users?
Based on the accounts I've tested, I believe it's affecting all VoIP unlock key users. I’ve been informed that it’s also affecting Fongo Home Phone users.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up [before other end answe
by Liptonbrisk » 11/05/2021
For others reading, FPL's switch used for VoIP unlock key users is in "maintenance mode, preparing for user migration to a newer version of the software and new hardware. To facilitate this the [switch] vendor added another proxy in the path." There is no known date yet for the migration or maintenance to be completed.
Consequently, there's nothing users can do but wait for a fix.
Based on SIP logs, the third party switch vendor seems to be https://www.portaone.com/.
It seems logical to conclude the other proxy is presenting an issue with respect to 487 not being received after CANCEL is sent.
For those inadvertently losing World Credits over this situation, choose "World Credits Inquiry" for the final issue type in a ticket request.
The escalation process is located at https://support.fongo.com/hc/en-us/arti ... -Complaint
Consequently, there's nothing users can do but wait for a fix.
Based on SIP logs, the third party switch vendor seems to be https://www.portaone.com/.
It seems logical to conclude the other proxy is presenting an issue with respect to 487 not being received after CANCEL is sent.
For those inadvertently losing World Credits over this situation, choose "World Credits Inquiry" for the final issue type in a ticket request.
The escalation process is located at https://support.fongo.com/hc/en-us/arti ... -Complaint
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up [before other end answe
by iceonu » 11/06/2021
Thanks for sharing all your knowledge and info, hopefully we will see this fix happen sooner than later. FPL VoIPUnlock key is great service to have.
- iceonu
- Just Passing Thru
- Posts: 13
- Joined: 06/04/2013
- SIP Device Name: Grandstream GXP2135
- ISP Name: Telus
- Computer OS: OS11.6.1
- Router: Ubiquiti UDM-Pro
Re: Call not ending after hanging up [before other end answe
by Liptonbrisk » 11/06/2021
iceonu wrote:Thanks for sharing all your knowledge and info,
You’re welcome!
hopefully we will see this fix happen sooner than later. FPL VoIPUnlock key is great service to have.
I agree, particularly because my parents and in-laws are VoIP unlock key customers.
Please do not send me emails; I do not work for nor represent Freephoneline or Fongo. Post questions on the forums so that others may learn from responses or assist you. Thank you. If you have an issue with your account or have a billing issue, submit a ticket here: https://support.fongo.com/hc/en-us/requests/new. Visit http://status.fongo.com/ to check FPL/Fongo service status. Freephoneline setup guides can be found at viewforum.php?f=15.
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Liptonbrisk - Technical Support
- Posts: 2661
- Joined: 04/26/2010
- SIP Device Name: Obihai 202/2182, Groundwire
- Firmware Version: various
- ISP Name: gateway/hub in bridge mode
- Computer OS: Windows 64 bit
- Router: Asuswrt-Merlin & others
Re: Call not ending after hanging up [before other end answe
by iceonu » 11/30/2021
Issue looks to be resolved. 487 termination request now being sent. 
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- iceonu
- Just Passing Thru
- Posts: 13
- Joined: 06/04/2013
- SIP Device Name: Grandstream GXP2135
- ISP Name: Telus
- Computer OS: OS11.6.1
- Router: Ubiquiti UDM-Pro
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