Outbound external calls drop after 5 minutes exactly

hady144 · December 16, 2023, 3:10pm

No issues with incoming internal or external calls but When I dial external call, during the External call at minute 5 I hear ( number is not answering ) then disconnected

Sip Debug at minute 5 as follow:

<------------>
[2023-12-16 14:57:36] WARNING[1088][C-00000021]: channel.c:4968 ast_prod: Prodding channel ‘SIP/608-0000003b’ failed
== Spawn extension (macro-dialout-trunk, s-NOANSWER, 4) exited non-zero on ‘SIP/608-0000003b’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 1819700, 12) exited non-zero on ‘SIP/608-0000003b’
– Executing [h@from-internal:1] Macro(“SIP/608-0000003b”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/608-0000003b”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/608-0000003b”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] Hangup(“SIP/608-0000003b”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/608-0000003b’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/608-0000003b’
– SIP/608-0000003b Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“SIP/608-0000003b”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“SIP/608-0000003b”, “HANGUP CAUSE: 34”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“SIP/608-0000003b”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“SIP/608-0000003b”, “MASTER CHANNEL: 1702738354.66 = 1702738354.66”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“SIP/608-0000003b”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“SIP/608-0000003b”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“SIP/608-0000003b”, “agi://127.0.0.1/sangomacrm.agi”) in new stack
– <SIP/608-0000003b>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“SIP/608-0000003b”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/608-0000003b’
– SIP/608-0000003b Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/608-0000003b

david55 · December 16, 2023, 3:20pm

The log shows the aftermath of the failure, not the actual failure. It looks like you set a five minute limit on answering and the limit was reached.

You’ve used screen scrape, so most of the timing information has been lost.

You should provide the full log contents for the complete call, as text, and, for this forum, on pastebin.freepbx.org or another similar service.

You may need to enable protocol logging, with “sip set debug on”>

chan_sip is no longer maintained by anyone.

hady144 · December 16, 2023, 8:12pm

83 Session Progress
Via: SIP/2.0/UDP 192.168.2.16:33154;branch=z9hG4bK1164867633;received=10.242.26.82;rport=33154
From: sip:[email protected];tag=786440204
To: sip:[email protected];tag=as6f78263a
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-15.0.37(16.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 1535935873 1535935878 IN IP4 10.242.13.10
s=Asterisk PBX 16.17.0
c=IN IP4 10.242.13.10
t=0 0
m=audio 19590 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<------------>
– Executing [s-NOANSWER@macro-dialout-trunk:4] Congestion(“SIP/608-0000003b”, “20”) in new stack

<— Reliably Transmitting (NAT) to 10.242.26.82:33154 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.2.16:33154;branch=z9hG4bK1164867633;received=10.242.26.82;rport=33154
From: sip:[email protected];tag=786440204
To: sip:[email protected];tag=as6f78263a
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-15.0.37(16.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

<------------>
[2023-12-16 14:57:36] WARNING[1088][C-00000021]: channel.c:4968 ast_prod: Prodding channel ‘SIP/608-0000003b’ failed
== Spawn extension (macro-dialout-trunk, s-NOANSWER, 4) exited non-zero on ‘SIP/608-0000003b’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 1819700, 12) exited non-zero on ‘SIP/608-0000003b’
– Executing [h@from-internal:1] Macro(“SIP/608-0000003b”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/608-0000003b”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/608-0000003b”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] Hangup(“SIP/608-0000003b”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/608-0000003b’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/608-0000003b’
– SIP/608-0000003b Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“SIP/608-0000003b”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“SIP/608-0000003b”, “HANGUP CAUSE: 34”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“SIP/608-0000003b”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“SIP/608-0000003b”, “MASTER CHANNEL: 1702738354.66 = 1702738354.66”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“SIP/608-0000003b”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“SIP/608-0000003b”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“SIP/608-0000003b”, “agi://127.0.0.1/sangomacrm.agi”) in new stack
– <SIP/608-0000003b>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“SIP/608-0000003b”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/608-0000003b’
– SIP/608-0000003b Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/608-0000003b

<— SIP read from UDP:10.242.26.82:33154 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.16:33154;branch=z9hG4bK1164867633;rport
From: sip:[email protected];tag=786440204
To: sip:[email protected];tag=as6f78263a
Call-ID: [email protected]
CSeq: 21 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: ACK
Reliably Transmitting (NAT) to 10.242.26.82:33154:
OPTIONS sip:[email protected]:33154 SIP/2.0
Via: SIP/2.0/UDP 10.242.13.10:5060;branch=z9hG4bK15bd49f4;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as4e788339
To: sip:[email protected]:33154
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.37(16.17.0)
Date: Sat, 16 Dec 2023 14:57:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

<— SIP read from UDP:10.242.26.82:33154 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.242.13.10:5060;branch=z9hG4bK15bd49f4;rport=5060
From: “Unknown” sip:[email protected];tag=as4e788339
To: sip:[email protected]:33154;tag=636673885
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:10.242.26.82:33154 —>

<------------->
Reliably Transmitting (NAT) to 10.242.26.82:33154:
OPTIONS sip:[email protected]:33154 SIP/2.0
Via: SIP/2.0/UDP 10.242.13.10:5060;branch=z9hG4bK7520d63b;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as7b30c239
To: sip:[email protected]:33154
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.37(16.17.0)
Date: Sat, 16 Dec 2023 14:58:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

Retransmitting #1 (NAT) to 10.242.26.82:33154:
OPTIONS sip:[email protected]:33154 SIP/2.0
Via: SIP/2.0/UDP 10.242.13.10:5060;branch=z9hG4bK7520d63b;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as7b30c239
To: sip:[email protected]:33154
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.37(16.17.0)
Date: Sat, 16 Dec 2023 14:58:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

<— SIP read from UDP:10.242.26.82:33154 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.242.13.10:5060;branch=z9hG4bK7520d63b;rport=5060
From: “Unknown” sip:[email protected];tag=as7b30c239
To: sip:[email protected]:33154;tag=402745258
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:10.242.26.82:33154 —>

<------------->
Reliably Transmitting (NAT) to 10.242.26.82:33154:
OPTIONS sip:[email protected]:33154 SIP/2.0
Via: SIP/2.0/UDP 10.242.13.10:5060;branch=z9hG4bK34d5ca0b;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as166921f2
To: sip:[email protected]:33154
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.37(16.17.0)
Date: Sat, 16 Dec 2023 14:59:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

<— SIP read from UDP:10.242.26.82:33154 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.242.13.10:5060;branch=z9hG4bK34d5ca0b;rport=5060
From: “Unknown” sip:[email protected];tag=as166921f2
To: sip:[email protected]:33154;tag=167462989
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
freepbxCLI> <------------>
freepbxCLI> [2023-12-16 14:57:36] WARNING[1088][C-00000021]: channel.c:4968 ast_prod: Prodding channel ‘SIP/608-0000003b’ failed
freepbxCLI> == Spawn extension (macro-dialout-trunk, s-NOANSWER, 4) exited non-zero on ‘SIP/608-0000003b’ in macro ‘dialout-trunk’
freepbxCLI> == Spawn extension (from-internal, 1819700, 12) exited non-zero on ‘SIP/608-0000003b’
freepbxCLI> – Executing [h@from-internal:1] Macro(“SIP/608-0000003b”, “hangupcall”) in new stack
freepbxCLI> – Executing [s@macro-hangupcall:1] GotoIf(“SIP/608-0000003b”, "1

david55 · December 16, 2023, 8:15pm

The log still doesn’t start until after the call has already failed.

hady144 · December 16, 2023, 8:18pm

thanks for reply
can you please tell me how to monitor the log on the way that you mean? what are steps or commands to do

Thanks

david55 · December 16, 2023, 8:23pm

Enable the full log through the GUI. Access the resulting log from /var/log/asterisk/full, using your favourite file transfer tools. It might also be available through the GUI.

Cause code 34 is congestion, but the log shows that Dial indicated no answer.

hady144 · December 16, 2023, 8:46pm

https://pastebin.freepbx.org/view/cff47e09

system · January 15, 2024, 8:47pm

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