SPA3000 cuts incoming call after 15-30 sec
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- Quiet One
- Posts: 38
- Joined: 11/17/2011
- SIP Device Name: oBi200
- ISP Name: Carrytel
- Computer OS: w10
- Router: Asus AC87U
SPA3000 cuts incoming call after 15-30 sec
I got perfectly "working" SPA3000 however there is one problem that I can't get rid of
Sometimes incoming call (only incoming) gets cut after 15-30 sec.
Looks like this will happen only if you use the VOIP number and after first call received from PSTN
All calls received after the first disconnection to PSTN are fine.
Next call received to VOIP registered provider will be cut but after that all calls to VOIP will work fine.
Any ideas what could be the problem?
The ATA is temporary in DMZ so this shouldn't be firewall problem
Thanks
Sometimes incoming call (only incoming) gets cut after 15-30 sec.
Looks like this will happen only if you use the VOIP number and after first call received from PSTN
All calls received after the first disconnection to PSTN are fine.
Next call received to VOIP registered provider will be cut but after that all calls to VOIP will work fine.
Any ideas what could be the problem?
The ATA is temporary in DMZ so this shouldn't be firewall problem
Thanks
- Funkytown
- Technical Support
- Posts: 460
- Joined: 04/01/2010
- SIP Device Name: Cisco SPA112
- Firmware Version: 1.4.1 (SR5) Oct 14 2
- ISP Name: Cik Telecom
- Computer OS: Windows 10
- Router: Zyxel EMG2926
- Smartphone Model: LG G8 Thinq
- Android Version: Android Q
Re: SPA3000 cuts incoming call after 15-30 sec
I am sorry for the delay in responding, you can always try my suggestions here http://forum.fongo.com/viewtopic.php?f=6&t=7944
Good Luck
Good Luck
-
- Quiet One
- Posts: 38
- Joined: 11/17/2011
- SIP Device Name: oBi200
- ISP Name: Carrytel
- Computer OS: w10
- Router: Asus AC87U
Re: SPA3000 cuts incoming call after 15-30 sec
Thanks , will do
-
- Quiet One
- Posts: 38
- Joined: 11/17/2011
- SIP Device Name: oBi200
- ISP Name: Carrytel
- Computer OS: w10
- Router: Asus AC87U
Re: SPA3000 cuts incoming call after 15-30 sec
1. First suggestion is to check inside the modem/router for a setting "keep alive" - Enable this, it's basically telling the devices connected to the modem/router that there still here and alive and not to worry.
Enabled (10sec)
2. Second suggestion you might have to upgrade the modem/router firmware.
Latest
3. Third suggestion is to reduce the firewalls to minimum in the modem/router.
DMZ enabled
4. Fourth suggestion check your modem/router for a SIP ALG setting - Disable this.
Disabled but shouldn't matter since DMZ enabled
Enabled (10sec)
2. Second suggestion you might have to upgrade the modem/router firmware.
Latest
3. Third suggestion is to reduce the firewalls to minimum in the modem/router.
DMZ enabled
4. Fourth suggestion check your modem/router for a SIP ALG setting - Disable this.
Disabled but shouldn't matter since DMZ enabled
- Funkytown
- Technical Support
- Posts: 460
- Joined: 04/01/2010
- SIP Device Name: Cisco SPA112
- Firmware Version: 1.4.1 (SR5) Oct 14 2
- ISP Name: Cik Telecom
- Computer OS: Windows 10
- Router: Zyxel EMG2926
- Smartphone Model: LG G8 Thinq
- Android Version: Android Q
Re: SPA3000 cuts incoming call after 15-30 sec
Try (60sec) this should work.lifeisfun wrote:Enabled (10sec)
-
- Quiet One
- Posts: 38
- Joined: 11/17/2011
- SIP Device Name: oBi200
- ISP Name: Carrytel
- Computer OS: w10
- Router: Asus AC87U
Re: SPA3000 cuts incoming call after 15-30 sec
Unfortunately no change, will try to use sip debug.
-
- Quiet One
- Posts: 38
- Joined: 11/17/2011
- SIP Device Name: oBi200
- ISP Name: Carrytel
- Computer OS: w10
- Router: Asus AC87U
Re: SPA3000 cuts incoming call after 15-30 sec
Any ideas why it cut off the call ?
Thanks
[0:5060]<<ip:5060
ACK sip:2222222222@ip:5060 SIP/2.0
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-d8754z-6941bd0134481612-1---d8754z-;rport
Via: SIP/2.0/UDP ip:5061;rport=5061;branch=z9hG4bK-b7ixozcd7tsgxh5m
Max-Forwards: 69
To: <sip:222222222222@ip>;tag=3ba38fee197cfa3di0
From: "222222222222"<sip:22222222222@ip>;tag=mge3kqcby2k5avmo.o
Call-ID: 11111111111@192.168.1.236~o
CSeq: 232 ACK
User-Agent: Sippy
Content-Length: 0
CC:Connected
[0:0]ENC INIT 0
[0:0]RTP Tx Up (pt=0->43d4085d:16452)
[0:0]RTCP Tx Up
[0:0]RTP Rx 1st PKT @16428(3)
[0:0]DEC INIT 0
[0]On Hook
[0:0]AUD Rel Call
[0:5060]->ip:5060
BYE sip:ip:5061 SIP/2.0
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-e9dc9a22;rport
From: <sip:22222222222@ip>;tag=3ba38fee197cfa3di0
To: "3438832130" <sip:222222222@ip>;tag=mge3kqcby2k5avmo.o
Call-ID: 111111111111@192.168.1.236~o
CSeq: 101 BYE
Max-Forwards: 70
Route: <sip:ip:5060;lr>, <sip:ip:5060;lr;transport=UDP>
User-Agent: Sipura/SPA3000-3.1.7(GWc)
Content-Length: 0
[0:5060]<<ip:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-e9dc9a22;rport=5060
To: "22222222222"<sip:22222222222@ip>;tag=mge3kqcby2k5avmo.o
From: <sip:222222222222@ip>;tag=3ba38fee197cfa3di0
Call-ID: 11111111111@192.168.1.236~o
CSeq: 101 BYE
Server: Sippy
Content-Length: 0
DLG Terminated
Sess Terminated
RSE_DEBUG: reference domain:voip.provider.ca
[0:5060]->ip:5060
NOTIFY sip:voip.provider.ca SIP/2.0
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-8b6266c0;rport
From: 222222222 <sip:2222222222@voip.freephoneline.ca>;tag=19a51be41bfe29d1o0
To: <sip:voip.freephoneline.ca>
Call-ID: 11111111111@10.0.0.130
CSeq: 6831 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA3000-3.1.7(GWc)
Content-Length: 0
[0:5060]<<ip:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-8b6266c0;rport
From: 2222222222 <sip:222222222@voip.provider.ca>;tag=19a51be41bfe29d1o0
To: <sip:voip.provider.ca>;tag=deadbeef
Call-ID: 1111111111111111@10.0.0.130
CSeq: 6831 NOTIFY
Content-Length: 0
RSE_DEBUG: unref domain, voip.provider.ca
RSE_DEBUG: last unref for domain voip.provider.ca
CC:Clean Up
--- OBJ POOL STAT ---
OP:RTPRXB = 96 ( 96 192) OP:RTPREB = 40 ( 40 48)
OP:RTPTXB = 64 ( 64 108) OP:TIMEOU = 110 (120 40)
OP:SIPCOR = 0 ( 1 28) OP:SIPCTS = 32 ( 32 564)
OP:SIPSTS = 32 ( 32 3452) OP:SIPAUS = 2 ( 8 588)
OP:SIPDLG = 10 ( 10 140) OP:SIPSES = 12 ( 12 7920)
OP:SIPREG = 2 ( 4 252) OP:SIPLIN = 0 ( 2 128)
OP:STUNTS = 16 ( 16 68)
RSE_DEBUG: reference domain:voip.provider.ca
[0:5060]->ip:5060
NOTIFY sip:voip.provider.ca SIP/2.0
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-509ee45d;rport
From: 2222222222 <sip:2222222222@voip.provider.ca>;tag=19a51be41bfe29d1o0
To: <sip:voip.provider.ca>
Call-ID: 111111111111@10.0.0.130
CSeq: 6832 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA3000-3.1.7(GWc)
Content-Length: 0
Thanks
[0:5060]<<ip:5060
ACK sip:2222222222@ip:5060 SIP/2.0
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-d8754z-6941bd0134481612-1---d8754z-;rport
Via: SIP/2.0/UDP ip:5061;rport=5061;branch=z9hG4bK-b7ixozcd7tsgxh5m
Max-Forwards: 69
To: <sip:222222222222@ip>;tag=3ba38fee197cfa3di0
From: "222222222222"<sip:22222222222@ip>;tag=mge3kqcby2k5avmo.o
Call-ID: 11111111111@192.168.1.236~o
CSeq: 232 ACK
User-Agent: Sippy
Content-Length: 0
CC:Connected
[0:0]ENC INIT 0
[0:0]RTP Tx Up (pt=0->43d4085d:16452)
[0:0]RTCP Tx Up
[0:0]RTP Rx 1st PKT @16428(3)
[0:0]DEC INIT 0
[0]On Hook
[0:0]AUD Rel Call
[0:5060]->ip:5060
BYE sip:ip:5061 SIP/2.0
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-e9dc9a22;rport
From: <sip:22222222222@ip>;tag=3ba38fee197cfa3di0
To: "3438832130" <sip:222222222@ip>;tag=mge3kqcby2k5avmo.o
Call-ID: 111111111111@192.168.1.236~o
CSeq: 101 BYE
Max-Forwards: 70
Route: <sip:ip:5060;lr>, <sip:ip:5060;lr;transport=UDP>
User-Agent: Sipura/SPA3000-3.1.7(GWc)
Content-Length: 0
[0:5060]<<ip:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-e9dc9a22;rport=5060
To: "22222222222"<sip:22222222222@ip>;tag=mge3kqcby2k5avmo.o
From: <sip:222222222222@ip>;tag=3ba38fee197cfa3di0
Call-ID: 11111111111@192.168.1.236~o
CSeq: 101 BYE
Server: Sippy
Content-Length: 0
DLG Terminated
Sess Terminated
RSE_DEBUG: reference domain:voip.provider.ca
[0:5060]->ip:5060
NOTIFY sip:voip.provider.ca SIP/2.0
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-8b6266c0;rport
From: 222222222 <sip:2222222222@voip.freephoneline.ca>;tag=19a51be41bfe29d1o0
To: <sip:voip.freephoneline.ca>
Call-ID: 11111111111@10.0.0.130
CSeq: 6831 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA3000-3.1.7(GWc)
Content-Length: 0
[0:5060]<<ip:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-8b6266c0;rport
From: 2222222222 <sip:222222222@voip.provider.ca>;tag=19a51be41bfe29d1o0
To: <sip:voip.provider.ca>;tag=deadbeef
Call-ID: 1111111111111111@10.0.0.130
CSeq: 6831 NOTIFY
Content-Length: 0
RSE_DEBUG: unref domain, voip.provider.ca
RSE_DEBUG: last unref for domain voip.provider.ca
CC:Clean Up
--- OBJ POOL STAT ---
OP:RTPRXB = 96 ( 96 192) OP:RTPREB = 40 ( 40 48)
OP:RTPTXB = 64 ( 64 108) OP:TIMEOU = 110 (120 40)
OP:SIPCOR = 0 ( 1 28) OP:SIPCTS = 32 ( 32 564)
OP:SIPSTS = 32 ( 32 3452) OP:SIPAUS = 2 ( 8 588)
OP:SIPDLG = 10 ( 10 140) OP:SIPSES = 12 ( 12 7920)
OP:SIPREG = 2 ( 4 252) OP:SIPLIN = 0 ( 2 128)
OP:STUNTS = 16 ( 16 68)
RSE_DEBUG: reference domain:voip.provider.ca
[0:5060]->ip:5060
NOTIFY sip:voip.provider.ca SIP/2.0
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-509ee45d;rport
From: 2222222222 <sip:2222222222@voip.provider.ca>;tag=19a51be41bfe29d1o0
To: <sip:voip.provider.ca>
Call-ID: 111111111111@10.0.0.130
CSeq: 6832 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA3000-3.1.7(GWc)
Content-Length: 0