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 [[email protected]:1] ExecIf(“SIP/level3-00010478”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
pbx.c: Executing [[email protected]:2] NoOp(“SIP/level3-00010478”, “Applying SIP Headers to channel SIP/level3-00010478”) in new stack
pbx.c: Executing [[email protected]:3] Set(“SIP/level3-00010478”, “TECH=SIP”) in new stack
pbx.c: Executing [[email protected]:4] Set(“SIP/level3-00010478”, “SIPHEADERKEYS=”) in new stack
pbx.c: Executing [[email protected]:5] While(“SIP/level3-00010478”, “0”) in new stack
app_while.c: Jumping to priority 13
pbx.c: Executing [[email protected]: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 [[email protected]:1] Macro(“PJSIP/5331-00000150”, “hangupcall”) in new stack
pbx.c: Executing [[email protected]:1] GotoIf(“PJSIP/5331-00000150”, “1?theend”) in new stack
pbx_builtins.c: Goto (macro-hangupcall,s,3)
pbx.c: Executing [[email protected]:3] ExecIf(“PJSIP/5331-00000150”, “0?Set(CDR(recordingfile)=)”) in new stack
pbx.c: Executing [[email protected]:4] NoOp(“PJSIP/5331-00000150”, "SIP/level3-00010478 montior file= ") in new stack
pbx.c: Executing [[email protected]:5] GotoIf(“PJSIP/5331-00000150”, “1?skipagi”) in new stack
pbx_builtins.c: Goto (macro-hangupcall,s,7)
pbx.c: Executing [[email protected]: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 [[email protected]:1] NoOp(“PJSIP/5331-00000150”, “Sending Hangup to CRM”) in new stack
pbx.c: Executing [[email protected]:2] NoOp(“PJSIP/5331-00000150”, “HANGUP CAUSE: 127”) in new stack
pbx.c: Executing [[email protected]:3] ExecIf(“PJSIP/5331-00000150”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
pbx.c: Executing [[email protected]:4] NoOp(“PJSIP/5331-00000150”, “MASTER CHANNEL: 1593033876.183038 = 1593033876.183038”) in new stack
pbx.c: Executing [[email protected]:5] GotoIf(“PJSIP/5331-00000150”, “0?return”) in new stack
pbx.c: Executing [[email protected]:6] Set(“PJSIP/5331-00000150”, “__CRM_HANGUP=1”) in new stack
pbx.c: Executing [[email protected]: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 [[email protected]rm-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 [email protected]:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response