Trouble with Outbound SIP


(John) #1

Our Telco provider sends us calls first over PRIs and if that fails or if they are full, then via SIP. During a PRI outage yesterday we noticed that none of the SIP calls were coming in or out.

I’ve been doing some test calls from the SIP trunks and have found an odd behavior. If I dial my cell phone from the PBX, my cell phone rings and I can answer but my phone on the PBX shows “Trying” on the display. If I hang up the cell phone it rings again, but again the call never completes.

it’s like the signaling starts but never completes.

netsock2.c: Using SIP RTP TOS bits 184

netsock2.c: Using SIP RTP CoS mark 5

app_stack.c: SIP/level3-00010478 Internal Gosub(func-apply-sipheaders,s,1(1)) start

pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf(“SIP/level3-00010478”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack

pbx.c: Executing [s@func-apply-sipheaders:2] NoOp(“SIP/level3-00010478”, “Applying SIP Headers to channel SIP/level3-00010478”) in new stack

pbx.c: Executing [s@func-apply-sipheaders:3] Set(“SIP/level3-00010478”, “TECH=SIP”) in new stack

pbx.c: Executing [s@func-apply-sipheaders:4] Set(“SIP/level3-00010478”, “SIPHEADERKEYS=”) in new stack

pbx.c: Executing [s@func-apply-sipheaders:5] While(“SIP/level3-00010478”, “0”) in new stack

app_while.c: Jumping to priority 13

pbx.c: Executing [s@func-apply-sipheaders:14] Return(“SIP/level3-00010478”, “”) in new stack

app_stack.c: Spawn extension (from-trunk-sip-level3, 82145551212, 1) exited non-zero on ‘SIP/level3-00010478’

app_stack.c: SIP/level3-00010478 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=

app_dial.c: Called SIP/level3/2145551212

app_macro.c: Spawn extension (macro-dialout-trunk, s, 33) exited non-zero on ‘PJSIP/5331-00000150’ in macro ‘dialout-trunk’

pbx.c: Spawn extension (from-internal, 82145551212, 6) exited non-zero on ‘PJSIP/5331-00000150’

pbx.c: Executing [h@from-internal:1] Macro(“PJSIP/5331-00000150”, “hangupcall”) in new stack

pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/5331-00000150”, “1?theend”) in new stack

pbx_builtins.c: Goto (macro-hangupcall,s,3)

pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/5331-00000150”, “0?Set(CDR(recordingfile)=)”) in new stack

pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/5331-00000150”, "SIP/level3-00010478 montior file= ") in new stack

pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/5331-00000150”, “1?skipagi”) in new stack

pbx_builtins.c: Goto (macro-hangupcall,s,7)

pbx.c: Executing [s@macro-hangupcall:7] Hangup(“PJSIP/5331-00000150”, “”) in new stack

app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/5331-00000150’ in macro ‘hangupcall’

pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/5331-00000150’

app_stack.c: PJSIP/5331-00000150 Internal Gosub(crm-hangup,s,1) start

pbx.c: Executing [s@crm-hangup:1] NoOp(“PJSIP/5331-00000150”, “Sending Hangup to CRM”) in new stack

pbx.c: Executing [s@crm-hangup:2] NoOp(“PJSIP/5331-00000150”, “HANGUP CAUSE: 127”) in new stack

pbx.c: Executing [s@crm-hangup:3] ExecIf(“PJSIP/5331-00000150”, “0?Set(__CRM_VOICEMAIL=)”) in new stack

pbx.c: Executing [s@crm-hangup:4] NoOp(“PJSIP/5331-00000150”, “MASTER CHANNEL: 1593033876.183038 = 1593033876.183038”) in new stack

pbx.c: Executing [s@crm-hangup:5] GotoIf(“PJSIP/5331-00000150”, “0?return”) in new stack

pbx.c: Executing [s@crm-hangup:6] Set(“PJSIP/5331-00000150”, “__CRM_HANGUP=1”) in new stack

pbx.c: Executing [s@crm-hangup:7] AGI(“PJSIP/5331-00000150”, “sangomacrm.agi”) in new stack

res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi

res_agi.c: <PJSIP/5331-00000150>AGI Script sangomacrm.agi completed, returning 0

pbx.c: Executing [s@crm-hangup:8] Return(“PJSIP/5331-00000150”, “”) in new stack

app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/5331-00000150’

app_stack.c: PJSIP/5331-00000150 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

[2020-06-24 21:25:08] WARNING[16234] chan_sip.c: Retransmission timeout reached on transmission 1fbe12da0b0ada3640b4fdd14674f453@12.156.39.41:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 31999ms with no response


#2

Is your PBX behind a NAT device?


(John) #3

no it’s got a public ip directly on it.


#4

Can you show us a sip packet trace of a test call? Do you remember the last time things were known to be working over this trunk?


(system) closed #5

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