Hello. Yeastar S20
SIP Registered, phones (Cisco SAP504G) too.
All calls (inbound and outbound) cuts on 186 second.
3m06sec
I'm not able to chek calls once again, there is no one in the office at this time.
May be someone have an idea about the menu parameter that can couse this issue.
IPPBX*CLI> == Parsing '/etc/asterisk/pjsip_wizard.conf': Found
IPPBX*CLI>
IPPBX*CLI> -- Channel PJSIP/trunk-RT-endpoint-00000011 left 'simple_bridge' basic-bridge <c3af900d-a1b3-4456-bd91-aa517a01f001>
IPPBX*CLI>
IPPBX*CLI> -- Channel PJSIP/403-00000012 left 'simple_bridge' basic-bridge <c3af900d-a1b3-4456-bd91-aa517a01f001>
IPPBX*CLI>
IPPBX*CLI> == Spawn extension (macro-realstexten, s, 19) exited non-zero on 'PJSIP/trunk-RT-endpoint-00000011' in macro 'realstexten'
IPPBX*CLI>
IPPBX*CLI> == Spawn extension (macro-stdexten-fromoutside, s, 4) exited non-zero on 'PJSIP/trunk-RT-endpoint-00000011' in macro 'stdexten-fromoutside'
IPPBX*CLI>
IPPBX*CLI> == Spawn extension (ivr_6500, 403, 1) exited non-zero on 'PJSIP/trunk-RT-endpoint-00000011'
IPPBX*CLI>
IPPBX*CLI> -- Executing [h@ivr_6500:1] Hangup("PJSIP/trunk-RT-endpoint-00000011", "") in new stack
IPPBX*CLI>
IPPBX*CLI> == Spawn extension (ivr_6500, h, 1) exited non-zero on 'PJSIP/trunk-RT-endpoint-00000011'
IPPBX*CLI>
Maybe it help?
This sounds like an issue related to Session-Timer.
Try change the session timer settings and see if that help.
If not, post a pcap logs for the calls here.
Hello Oscar, no help, I've already tried to to so.
My log:
== Parsing '/etc/asterisk/pjsip.conf': Found
== Parsing '/etc/asterisk/pjsip_auth.conf': Found
== Parsing '/etc/asterisk/pjsip_wizard.conf': Found
<--- Received SIP request (535 bytes) from UDP:213.228.77.123:5060 --->
INFO sip:5850034@10.24.160.140:5060 SIP/2.0
From: <sip:4959701933@ngn-sip.sinor.ru;user=phone>;tag=5924
To: "SIPLineUser SIPLineUser"<sip:5850034@ngn-sip.sinor.ru>;tag=afe137be-bdfd-4833-ad52-d28151daad83
Call-ID: ce993e45066439ad781ea16c844ecee4b9a73599@213.228.77.123
CSeq: 82575 INFO
Via: SIP/2.0/UDP 213.228.77.123:5060;branch=z9hG4bK-29e4c93-3a5b1f8f-5c844799
User-Agent: Nortel SESM 17.0.12.20
Max-Forwards: 20
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin
Content-Length: 0
<--- Received SIP request (535 bytes) from UDP:213.228.77.123:5060 --->
INFO sip:5850034@10.24.160.140:5060 SIP/2.0
From: <sip:4959701933@ngn-sip.sinor.ru;user=phone>;tag=5924
To: "SIPLineUser SIPLineUser"<sip:5850034@ngn-sip.sinor.ru>;tag=afe137be-bdfd-4833-ad52-d28151daad83
Call-ID: ce993e45066439ad781ea16c844ecee4b9a73599@213.228.77.123
CSeq: 82575 INFO
Via: SIP/2.0/UDP 213.228.77.123:5060;branch=z9hG4bK-29e4c93-3a5b1f8f-5c844799
User-Agent: Nortel SESM 17.0.12.20
Max-Forwards: 20
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin
Content-Length: 0
<--- Received SIP request (533 bytes) from UDP:213.228.77.123:5060 --->
BYE sip:5850034@10.24.160.140:5060 SIP/2.0
From: <sip:4959701933@ngn-sip.sinor.ru;user=phone>;tag=5924
To: "SIPLineUser SIPLineUser"<sip:5850034@ngn-sip.sinor.ru>;tag=afe137be-bdfd-4833-ad52-d28151daad83
Call-ID: ce993e45066439ad781ea16c844ecee4b9a73599@213.228.77.123
CSeq: 82576 BYE
Via: SIP/2.0/UDP 213.228.77.123:5060;branch=z9hG4bK-29e4cb3-3a5b9c96-3ce8e55a
User-Agent: Nortel SESM 17.0.12.20
Max-Forwards: 20
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin
Content-Length: 0
<--- Transmitting SIP response (419 bytes) to UDP:213.228.77.123:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.228.77.123:5060;received=213.228.77.123;branch=z9hG4bK-29e4cb3-3a5b9c96-3ce8e55a
Call-ID: ce993e45066439ad781ea16c844ecee4b9a73599@213.228.77.123
From: <sip:4959701933@ngn-sip.sinor.ru;user=phone>;tag=5924
To: "SIPLineUser SIPLineUser" <sip:5850034@ngn-sip.sinor.ru>;tag=afe137be-bdfd-4833-ad52-d28151daad83
CSeq: 82576 BYE
Server: Yeastar S20-30.5.0.30
Content-Length: 0
-- Channel PJSIP/trunk-RT-endpoint-00000007 left 'simple_bridge' basic-bridge <ee3a67e6-5e3a-48c3-8225-be128549236e>
-- Channel PJSIP/401-00000008 left 'simple_bridge' basic-bridge <ee3a67e6-5e3a-48c3-8225-be128549236e>
<--- Transmitting SIP request (435 bytes) to UDP:192.168.200.152:5060 --->
BYE sip:401@192.168.200.152:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.150:5060;rport;branch=z9hG4bKPj437b466a-4a73-4a48-a47f-49ed050505d6
From: <sip:4959701933@10.24.160.140>;tag=b286ebcd-38bf-48a1-861a-046451a38b74
To: <sip:401@192.168.200.152>;tag=3d1e64fc6db3c3bci0
Call-ID: cd6673ce-d2ec-4d33-8b0a-388376bcc8ab
CSeq: 5537 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Yeastar S20-30.5.0.30
Content-Length: 0
== Spawn extension (macro-realstexten, s, 19) exited non-zero on 'PJSIP/trunk-RT-endpoint-00000007' in macro 'realstexten'
<--- Received SIP response (358 bytes) from UDP:192.168.200.152:5060 --->
SIP/2.0 200 OK
To: <sip:401@192.168.200.152>;tag=3d1e64fc6db3c3bci0
From: <sip:4959701933@10.24.160.140>;tag=b286ebcd-38bf-48a1-861a-046451a38b74
Call-ID: cd6673ce-d2ec-4d33-8b0a-388376bcc8ab
CSeq: 5537 BYE
Via: SIP/2.0/UDP 192.168.200.150:5060;branch=z9hG4bKPj437b466a-4a73-4a48-a47f-49ed050505d6
Server: Cisco/SPA504G-7.5.6(XU)
Content-Length: 0
== Spawn extension (macro-stdexten-fromoutside, s, 4) exited non-zero on 'PJSIP/trunk-RT-endpoint-00000007' in macro 'stdexten-fromoutside'
== Spawn extension (ivr_6500, 401, 1) exited non-zero on 'PJSIP/trunk-RT-endpoint-00000007'
-- Executing [h@ivr_6500:1] Hangup("PJSIP/trunk-RT-endpoint-00000007", "") in new stack
== Spawn extension (ivr_6500, h, 1) exited non-zero on 'PJSIP/trunk-RT-endpoint-00000007'
<--- Transmitting SIP request (439 bytes) to UDP:192.168.200.151:5060 --->
OPTIONS sip:400@192.168.200.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.150:5060;rport;branch=z9hG4bKPjc9aeefda-5c74-4812-91be-013bb095e92a
From: <sip:unknown@10.24.160.140>;tag=7a6c7b9e-5cae-4e12-8fcd-59bc85a1f512
To: <sip:400@192.168.200.151>
Contact: <sip:unknown@192.168.200.150:5060>
Call-ID: d4d7352c-4276-4f0e-bad1-b6b7037d44b0
CSeq: 32476 OPTIONS
Max-Forwards: 70
User-Agent: Yeastar S20-30.5.0.30
Content-Length: 0
<--- Received SIP response (451 bytes) from UDP:192.168.200.151:5060 --->
SIP/2.0 200 OK
To: <sip:400@192.168.200.151>;tag=b6c4c0ec387b13fi0
From: <sip:unknown@10.24.160.140>;tag=7a6c7b9e-5cae-4e12-8fcd-59bc85a1f512
Call-ID: d4d7352c-4276-4f0e-bad1-b6b7037d44b0
CSeq: 32476 OPTIONS
Via: SIP/2.0/UDP 192.168.200.150:5060;branch=z9hG4bKPjc9aeefda-5c74-4812-91be-013bb095e92a
Server: Cisco/SPA504G-7.5.6(XU)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
IPPBX*CLI>
From part of the SIP information, seems like the SIP trunk provider is trying to send SIP INFO message and expect Yeatar S20 to response but S20 doesn't response then it hangup the call.
Also, it would be better if you can capture the SIP logs through web GUI. It might be much clear for the whole SIP call flow.
If possible try register the SIP account in your IP phone and get a working pcap logs for comparing what would be the reason.
Don't know how to attach files in here. Cloudlink:
https://yadi.sk/d/zAI_AMzw3Qdmgv
We unplug the WAN cable, put it back, make a call, still 186 seconds.
Hello Sergey,
I believe this is an compatible issue between the Yeastar S PBX and your SIP trunk provider.
Your SIP trunk provider are trying to send SIP INFO message after the call establish. Normally, there is not such kind of SIP INFO message.
The worse thing is the SIP trunk provider seems expected to receive response from Yeastar S PBX. But as that is not a normal SIP message of a SIP call flow, the Yeastar S PBX doesnt response.
Then the SIP trunk provider disconnect the call. See below the call flow:
The ways to fix this issue:
1. Ask the SIP trunk provider to disable SIP INFO message;
2. Ask Yeastar to be compatible with the SIP INFO message by submitting a ticket;
I would suggest you to try both ways at the same time.
One question:
If you register the SIP account on an IP phone, will it work normally? If so, I suggest you to provide SIP logs for the workings one(register SIP account on the IP phone) for Yeastar reference. This will shorter the time they fix the issue.
Hello Oscar!
Thank you for response.
We've tried to setup sip phones directly, but no luck, there where no response from provider for registration. If we will sucseed, I'll sent you the logs.