Not receiving BYE, outbound calls never hungup

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alex6999
One Hit Wonder
Posts: 1
Joined: 06/11/2015
SIP Device Name: Asterisk
Firmware Version: 11.13.1
ISP Name: VPS
Computer OS: Debian 8
Router: white IP

Not receiving BYE, outbound calls never hungup

Post by alex6999 »

I trying 2 different clients, first - build in android client, outbound calls still active, but remote party hangup
now i setup freephoneline on asterisk

Lets Call from my cell to FLP

Code: Select all

<--- SIP read from UDP:208.65.240.44:5060 --->
INVITE sip:1FPL=NUMBER@208.65.240.44:5060;transport=UDP SIP/2.0
Record-Route: <sip:1FPL=NUMBER@208.65.240.44;lr=on>
Via: SIP/2.0/UDP 208.65.240.44;branch=z9hG4bK3b75.6095c106.0
Via: SIP/2.0/UDP 208.65.240.165:5060;branch=z9hG4bK-524287-1---3f491f14e4014f4d;rport=5060
Via: SIP/2.0/UDP 208.65.240.165:5061;branch=z9hG4bK-fegje7dv3m3rgmyr;rport=5061
Max-Forwards: 69
Record-Route: <sip:208.65.240.165:5060;lr;transport=UDP>
Contact: "Anonymous"<sip:208.65.240.165:5061>
To: <sip:1FPL=NUMBER@208.65.240.165>
From: <sip:CELL=PHONE@208.65.240.165>;tag=lqsmv3evhkcvzkav.o
Call-ID: 4F2F0532@208.72.120.66~o~o
CSeq: 172 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
h323-conf-id: 3704975364-267784677-2432565275-559912524
cisco-GUID: 3704975364-267784677-2432565275-559912524
Content-Length: 148

v=0
o=Sippy 126693072 0 IN IP4 208.65.240.165
s=-
t=0 0
m=audio 33842 RTP/AVP 0 101
c=IN IP4 208.72.121.67
a=rtpmap:101 telephone-event/8000
<------------->
--- (19 headers 7 lines) ---
Sending to 208.65.240.44:5060 (no NAT)
Sending to 208.65.240.44:5060 (no NAT)
Using INVITE request as basis request - 4F2F0532@208.72.120.66~o~o
Found peer 'FPL' for 'CELL=PHONE' from 208.65.240.44:5060
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 208.72.121.67:33842
Looking for 1FPL=NUMBER in incoming (domain 208.65.240.44)
list_route: hop: <sip:1FPL=NUMBER@208.65.240.44;lr=on>
list_route: hop: <sip:208.65.240.165:5060;lr;transport=UDP>

<--- Transmitting (no NAT) to 208.65.240.44:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.65.240.44;branch=z9hG4bK3b75.6095c106.0;received=208.65.240.44
Via: SIP/2.0/UDP 208.65.240.165:5060;branch=z9hG4bK-524287-1---3f491f14e4014f4d;rport=5060
Via: SIP/2.0/UDP 208.65.240.165:5061;branch=z9hG4bK-fegje7dv3m3rgmyr;rport=5061
Record-Route: <sip:1FPL=NUMBER@208.65.240.44;lr=on>
Record-Route: <sip:208.65.240.165:5060;lr;transport=UDP>
From: <sip:CELL=PHONE@208.65.240.165>;tag=lqsmv3evhkcvzkav.o
To: <sip:1FPL=NUMBER@208.65.240.165>
Call-ID: 4F2F0532@208.72.120.66~o~o
CSeq: 172 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1FPL=NUMBER@MY.IP.ADDRESS:5060>
Content-Length: 0


<------------>
Audio is at 18202
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 208.65.240.44:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.65.240.44;branch=z9hG4bK3b75.6095c106.0;received=208.65.240.44
Via: SIP/2.0/UDP 208.65.240.165:5060;branch=z9hG4bK-524287-1---3f491f14e4014f4d;rport=5060
Via: SIP/2.0/UDP 208.65.240.165:5061;branch=z9hG4bK-fegje7dv3m3rgmyr;rport=5061
Record-Route: <sip:1FPL=NUMBER@208.65.240.44;lr=on>
Record-Route: <sip:208.65.240.165:5060;lr;transport=UDP>
From: <sip:CELL=PHONE@208.65.240.165>;tag=lqsmv3evhkcvzkav.o
To: <sip:1FPL=NUMBER@208.65.240.165>;tag=as0a652845
Call-ID: 4F2F0532@208.72.120.66~o~o
CSeq: 172 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1FPL=NUMBER@MY.IP.ADDRESS:5060>
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 1473614804 1473614804 IN IP4 MY.IP.ADDRESS
s=Asterisk PBX 11.13.1~dfsg-2+b1
c=IN IP4 MY.IP.ADDRESS
t=0 0
m=audio 18202 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:208.65.240.44:5060 --->
ACK sip:1FPL=NUMBER@208.65.240.44:5060;transport=UDP SIP/2.0
Record-Route: <sip:1FPL=NUMBER@208.65.240.44;lr=on>
Via: SIP/2.0/UDP 208.65.240.44;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 208.65.240.165:5060;branch=z9hG4bK-524287-1---e4d03701ba148538;rport=5060
Via: SIP/2.0/UDP 208.65.240.165:5061;rport=5061;branch=z9hG4bK-n43bsoycl2sdpnf2
Max-Forwards: 69
Route: <sip:1FPL=NUMBER@208.65.240.44;lr>
To: <sip:1FPL=NUMBER@208.65.240.165>;tag=as0a652845
From: <sip:CELL=PHONE@208.65.240.165>;tag=lqsmv3evhkcvzkav.o
Call-ID: 4F2F0532@208.72.120.66~o~o
CSeq: 172 ACK
User-Agent: Sippy
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Now lets hung up my cell phone

Code: Select all

<--- SIP read from UDP:208.65.240.44:5060 --->
BYE sip:1FPL=NUMBER@208.65.240.44:5060;transport=UDP SIP/2.0
Record-Route: <sip:1FPL=NUMBER@208.65.240.44;lr=on>
Via: SIP/2.0/UDP 208.65.240.44;branch=z9hG4bK4b75.58ab27a5.0
Via: SIP/2.0/UDP 208.65.240.165:5060;branch=z9hG4bK-524287-1---1848b64239f9a938;rport=5060
Via: SIP/2.0/UDP 208.65.240.165:5061;branch=z9hG4bK-asxc6cj75yotbszl;rport=5061
Max-Forwards: 69
Route: <sip:1FPL=NUMBER@208.65.240.44;lr>
Contact: "Anonymous"<sip:208.65.240.165:5061>
To: <sip:1FPL=NUMBER@208.65.240.165>;tag=as0a652845
From: <sip:CELL=PHONE@208.65.240.165>;tag=lqsmv3evhkcvzkav.o
Call-ID: 4F2F0532@208.72.120.66~o~o
CSeq: 173 BYE
User-Agent: Sippy
h323-conf-id: 3704975364-267784677-2432565275-559912524
cisco-GUID: 3704975364-267784677-2432565275-559912524
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 208.65.240.44:5060 (no NAT)
Scheduling destruction of SIP dialog '4F2F0532@208.72.120.66~o~o' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 208.65.240.44:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.65.240.44;branch=z9hG4bK4b75.58ab27a5.0;received=208.65.240.44
Via: SIP/2.0/UDP 208.65.240.165:5060;branch=z9hG4bK-524287-1---1848b64239f9a938;rport=5060
Via: SIP/2.0/UDP 208.65.240.165:5061;branch=z9hG4bK-asxc6cj75yotbszl;rport=5061
Record-Route: <sip:1FPL=NUMBER@208.65.240.44;lr=on>
From: <sip:CELL=PHONE@208.65.240.165>;tag=lqsmv3evhkcvzkav.o
To: <sip:1FPL=NUMBER@208.65.240.165>;tag=as0a652845
Call-ID: 4F2F0532@208.72.120.66~o~o
CSeq: 173 BYE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Everything work fine, now let try to call from asterisk extention via FPL to my cell

I dont know why debug started from middle of SIP packet, I used "sip debug peer FPL"

Code: Select all

<--- SIP read from UDP:208.65.240.44:5060 --->
Audio is at 19676
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 208.65.240.44:5060:
INVITE sip:CELL=PHONE@voip.freephoneline.ca SIP/2.0
Via: SIP/2.0/UDP MY.IP.ADDRESS:5060;branch=z9hG4bK2474b410
Max-Forwards: 70
From: <sip:522@MY.IP.ADDRESS>;tag=as07b000af
To: <sip:CELL=PHONE@voip.freephoneline.ca>
Contact: <sip:522@MY.IP.ADDRESS:5060>
Call-ID: 5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Date: Thu, 11 Jun 2015 05:11:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 1597017536 1597017536 IN IP4 MY.IP.ADDRESS
s=Asterisk PBX 11.13.1~dfsg-2+b1
c=IN IP4 MY.IP.ADDRESS
t=0 0
m=audio 19676 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:208.65.240.44:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP MY.IP.ADDRESS:5060;rport=5060;branch=z9hG4bK2474b410
Record-Route: <sip:208.65.240.165:5060;transport=udp;lr>
Record-Route: <sip:CELL=PHONE@208.65.240.44;lr>
To: <sip:CELL=PHONE@voip.freephoneline.ca>
From: <sip:522@MY.IP.ADDRESS>;tag=as07b000af
Call-ID: 5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060
CSeq: 102 INVITE
Server: Sippy
WWW-Authenticate: Digest realm="208.65.240.165",nonce="893262c632891b8eb9baf9fa06759f9dbe89"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 208.65.240.44:5060:
ACK sip:CELL=PHONE@voip.freephoneline.ca SIP/2.0
Via: SIP/2.0/UDP MY.IP.ADDRESS:5060;branch=z9hG4bK2474b410
Max-Forwards: 70
From: <sip:522@MY.IP.ADDRESS>;tag=as07b000af
To: <sip:CELL=PHONE@voip.freephoneline.ca>
Contact: <sip:522@MY.IP.ADDRESS:5060>
Call-ID: 5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Content-Length: 0


---
Audio is at 19676
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 208.65.240.44:5060:
INVITE sip:CELL=PHONE@voip.freephoneline.ca SIP/2.0
Via: SIP/2.0/UDP MY.IP.ADDRESS:5060;branch=z9hG4bK448ae0b1
Max-Forwards: 70
From: <sip:522@MY.IP.ADDRESS>;tag=as07b000af
To: <sip:CELL=PHONE@voip.freephoneline.ca>
Contact: <sip:522@MY.IP.ADDRESS:5060>
Call-ID: 5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Authorization: Digest username="1FPL=NUMBER", realm="208.65.240.165", algorithm=MD5, uri="sip:CELL=PHONE@voip.freephoneline.ca", nonce="893262c632891b8eb9baf9fa06759f9dbe89", response="13c47a869a475ef2db5983e3eb0cfd11"
Date: Thu, 11 Jun 2015 05:11:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 1597017536 1597017537 IN IP4 MY.IP.ADDRESS
s=Asterisk PBX 11.13.1~dfsg-2+b1
c=IN IP4 MY.IP.ADDRESS
t=0 0
m=audio 19676 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #1 (no NAT) to 208.65.240.44:5060:
INVITE sip:CELL=PHONE@voip.freephoneline.ca SIP/2.0
Via: SIP/2.0/UDP MY.IP.ADDRESS:5060;branch=z9hG4bK448ae0b1
Max-Forwards: 70
From: <sip:522@MY.IP.ADDRESS>;tag=as07b000af
To: <sip:CELL=PHONE@voip.freephoneline.ca>
Contact: <sip:522@MY.IP.ADDRESS:5060>
Call-ID: 5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Authorization: Digest username="1FPL=NUMBER", realm="208.65.240.165", algorithm=MD5, uri="sip:CELL=PHONE@voip.freephoneline.ca", nonce="893262c632891b8eb9baf9fa06759f9dbe89", response="13c47a869a475ef2db5983e3eb0cfd11"
Date: Thu, 11 Jun 2015 05:11:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 1597017536 1597017537 IN IP4 MY.IP.ADDRESS
s=Asterisk PBX 11.13.1~dfsg-2+b1
c=IN IP4 MY.IP.ADDRESS
t=0 0
m=audio 19676 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #2 (no NAT) to 208.65.240.44:5060:
INVITE sip:CELL=PHONE@voip.freephoneline.ca SIP/2.0
Via: SIP/2.0/UDP MY.IP.ADDRESS:5060;branch=z9hG4bK448ae0b1
Max-Forwards: 70
From: <sip:522@MY.IP.ADDRESS>;tag=as07b000af
To: <sip:CELL=PHONE@voip.freephoneline.ca>
Contact: <sip:522@MY.IP.ADDRESS:5060>
Call-ID: 5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Authorization: Digest username="1FPL=NUMBER", realm="208.65.240.165", algorithm=MD5, uri="sip:CELL=PHONE@voip.freephoneline.ca", nonce="893262c632891b8eb9baf9fa06759f9dbe89", response="13c47a869a475ef2db5983e3eb0cfd11"
Date: Thu, 11 Jun 2015 05:11:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 1597017536 1597017537 IN IP4 MY.IP.ADDRESS
s=Asterisk PBX 11.13.1~dfsg-2+b1
c=IN IP4 MY.IP.ADDRESS
t=0 0
m=audio 19676 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #3 (no NAT) to 208.65.240.44:5060:
INVITE sip:CELL=PHONE@voip.freephoneline.ca SIP/2.0
Via: SIP/2.0/UDP MY.IP.ADDRESS:5060;branch=z9hG4bK448ae0b1
Max-Forwards: 70
From: <sip:522@MY.IP.ADDRESS>;tag=as07b000af
To: <sip:CELL=PHONE@voip.freephoneline.ca>
Contact: <sip:522@MY.IP.ADDRESS:5060>
Call-ID: 5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Authorization: Digest username="1FPL=NUMBER", realm="208.65.240.165", algorithm=MD5, uri="sip:CELL=PHONE@voip.freephoneline.ca", nonce="893262c632891b8eb9baf9fa06759f9dbe89", response="13c47a869a475ef2db5983e3eb0cfd11"
Date: Thu, 11 Jun 2015 05:11:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 1597017536 1597017537 IN IP4 MY.IP.ADDRESS
s=Asterisk PBX 11.13.1~dfsg-2+b1
c=IN IP4 MY.IP.ADDRESS
t=0 0
m=audio 19676 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:208.65.240.44:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP MY.IP.ADDRESS:5060;rport=5060;branch=z9hG4bK448ae0b1
Record-Route: <sip:208.65.240.165:5060;transport=udp;lr>
Record-Route: <sip:CELL=PHONE@208.65.240.44;lr>
Contact: "Anonymous"<sip:208.65.240.165:5061>
To: <sip:CELL=PHONE@voip.freephoneline.ca>;tag=rwdxshhilhaacgoi.i
From: <sip:522@MY.IP.ADDRESS>;tag=as07b000af
Call-ID: 5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060
CSeq: 103 INVITE
Content-Type: application/sdp
Server: Sippy
Content-Length: 161

v=0
o=Sippy 76179408 1 IN IP4 208.65.240.165
s=-
t=0 0
m=audio 11764 RTP/AVP 0 101
c=IN IP4 204.244.196.120
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (12 headers 8 lines) ---
list_route: hop: <sip:CELL=PHONE@208.65.240.44;lr>
list_route: hop: <sip:208.65.240.165:5060;transport=udp;lr>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 204.244.196.120:11764

<--- SIP read from UDP:208.65.240.44:5060 --->

<------------->

<--- SIP read from UDP:208.65.240.44:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP MY.IP.ADDRESS:5060;rport=5060;branch=z9hG4bK448ae0b1
Record-Route: <sip:208.65.240.165:5060;transport=udp;lr>
Record-Route: <sip:CELL=PHONE@208.65.240.44;lr>
Contact: "Anonymous"<sip:208.65.240.165:5061>
To: <sip:CELL=PHONE@voip.freephoneline.ca>;tag=rwdxshhilhaacgoi.i
From: <sip:522@MY.IP.ADDRESS>;tag=as07b000af
Call-ID: 5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060
CSeq: 103 INVITE
Content-Type: application/sdp
Server: Sippy
H323-credit-time: 14400
Content-Length: 161

v=0
o=Sippy 76179408 1 IN IP4 208.65.240.165
s=-
t=0 0
m=audio 11764 RTP/AVP 0 101
c=IN IP4 204.244.196.120
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (13 headers 8 lines) ---
list_route: hop: <sip:CELL=PHONE@208.65.240.44;lr>
list_route: hop: <sip:208.65.240.165:5060;transport=udp;lr>
Transmitting (no NAT) to 208.65.240.44:5060:
ACK sip:208.65.240.165:5061 SIP/2.0
Via: SIP/2.0/UDP MY.IP.ADDRESS:5060;branch=z9hG4bK0dbaecaa
Route: <sip:CELL=PHONE@208.65.240.44;lr>,<sip:208.65.240.165:5060;transport=udp;lr>
Max-Forwards: 70
From: <sip:522@MY.IP.ADDRESS>;tag=as07b000af
To: <sip:CELL=PHONE@voip.freephoneline.ca>;tag=rwdxshhilhaacgoi.i
Contact: <sip:522@MY.IP.ADDRESS:5060>
Call-ID: 5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Content-Length: 0


---

Now.... lets hangup cell phone......

Code: Select all

nothing happines, asterisk still thinking call in progress
and I think it still sending and STILL RECEIVING RPT packets from FPL
in case of non-receive PRT it will show retranmission warning like in one-way audio behind NAT scenario
i am not sure about it

Lets hungup extention

Code: Select all

Scheduling destruction of SIP dialog '5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:CELL=PHONE@208.65.240.44;lr> for address/port to send to
set_destination: set destination to 208.65.240.44:5060
Reliably Transmitting (no NAT) to 208.65.240.44:5060:
BYE sip:208.65.240.165:5061 SIP/2.0
Via: SIP/2.0/UDP MY.IP.ADDRESS:5060;branch=z9hG4bK406f2509
Route: <sip:CELL=PHONE@208.65.240.44;lr>,<sip:208.65.240.165:5060;transport=udp;lr>
Max-Forwards: 70
From: <sip:522@MY.IP.ADDRESS>;tag=as07b000af
To: <sip:CELL=PHONE@voip.freephoneline.ca>;tag=rwdxshhilhaacgoi.i
Call-ID: 5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060
CSeq: 104 BYE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Authorization: Digest username="1FPL=NUMBER", realm="208.65.240.165", algorithm=MD5, uri="sip:208.65.240.165:5061", nonce="893262c632891b8eb9baf9fa06759f9dbe89", response="2debd0a7bcdc47fcb482ac6e582298f5"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:208.65.240.44:5060 --->
SIP/2.0 481 Call Leg Does Not Exist
Via: SIP/2.0/UDP MY.IP.ADDRESS:5060;rport=5060;branch=z9hG4bK406f2509
Record-Route: <sip:208.65.240.44;lr>
To: <sip:CELL=PHONE@voip.freephoneline.ca>;tag=rwdxshhilhaacgoi.i
From: <sip:522@MY.IP.ADDRESS>;tag=as07b000af
Call-ID: 5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060
CSeq: 104 BYE
Server: Sippy
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '5cb5d2176a4ed31e2528e40879de9a88@MY.IP.ADDRESS:5060' Method: INVITE
Thats right, 481 Call Leg Does Not Exist, but FPL didnt tell it to my server.
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Mango
Tried and True
Posts: 411
Joined: 08/14/2014
SIP Device Name: OBi110
Firmware Version: 1.3.0 (Build: 2824)
ISP Name: Telus
Computer OS: Windows 7
Router: Toastman Tomato

Re: Not receiving BYE, outbound calls never hungup

Post by Mango »

I don't fully understand your first line. Are you saying that you see the same behaviour with your Android phone as with Asterisk?

Try to use sip set debug on (instead of peer) to see if the packet is arriving, but Asterisk is not recognizing it. If the packet arrives, you need to correct your sip.conf. You can probably skip this step, if you see the same behaviour with Android as with Asterisk as that would tend to eliminate Asterisk configuration problems.

If the packet never arrives, and if your Asterisk is behind NAT, try to set externip=your.public.ip.address from within the [general] context of sip.conf. Do a sip reload from the console and test. Don't worry if you have a dynamic IP - this is just to confirm my guess about what is causing the problem.

Let me know how things go.
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