No auido when forwarding call to a cell phone

bah12 · March 27, 2025, 2:25pm

I have a Yeahlink set to forward calls to a cell phone. The forward happens but once connected there is no audio in either direction. I’m not seeing anything in the log that shows an error.

25782	[2025-03-27 08:53:23] VERBOSE[3641887][C-000016c4] pbx.c: Spawn extension (trunk-dial-with-exten, 9407265349, 1) exited non-zero on 'PJSIP/985415-00003e55'	
25783	[2025-03-27 08:53:23] VERBOSE[3641887][C-000016c4] app_stack.c: Spawn extension (trunk-dial-with-exten, 9407265349, 1) exited non-zero on 'PJSIP/985415-00003e55'	
26282	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@from-internal:1] Gosub("PJSIP/985415-00003e5a", "macro-user-callerid,s,1(LIMIT,EXTERNAL)") in new stack	
26283	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@from-internal:2] Gosub("PJSIP/985415-00003e5a", "sub-record-check,s,1(out,9407265349,dontcare)") in new stack	
26284	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [out@sub-record-check:1] NoOp("PJSIP/985415-00003e5a", "Outbound Recording Check from 5415 to 9407265349") in new stack	
26285	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [out@sub-record-check:7] Gosub("PJSIP/985415-00003e5a", "recordcheck,1(dontcare,out,9407265349)") in new stack	
26286	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@from-internal:3] ExecIf("PJSIP/985415-00003e5a", "0 ?Set(CHANNEL(accountcode)=)") in new stack	
26287	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@from-internal:4] Set("PJSIP/985415-00003e5a", "_ROUTEID=19") in new stack	
26288	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@from-internal:5] Set("PJSIP/985415-00003e5a", "_ROUTENAME=Nortex_SIP") in new stack	
26289	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@from-internal:6] Set("PJSIP/985415-00003e5a", "EMERGENCYROUTE=YES") in new stack	
26290	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@from-internal:7] Set("PJSIP/985415-00003e5a", "MOHCLASS=default") in new stack	
26291	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@from-internal:8] ExecIf("PJSIP/985415-00003e5a", "1?Set(TRUNKCIDOVERRIDE="Alan Ritchey" <9407263276>)") in new stack	
26292	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@from-internal:9] Set("PJSIP/985415-00003e5a", "_CALLERIDNAMEINTERNAL=Brad Hieb") in new stack	
26293	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@from-internal:10] Set("PJSIP/985415-00003e5a", "_CALLERIDNUMINTERNAL=5415") in new stack	
26294	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@from-internal:11] Set("PJSIP/985415-00003e5a", "_EMAILNOTIFICATION=FALSE") in new stack	
26295	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@from-internal:12] Set("PJSIP/985415-00003e5a", "_NODEST=") in new stack	
26296	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@from-internal:13] Gosub("PJSIP/985415-00003e5a", "macro-dialout-trunk,s,1(4,9407265349,,off)") in new stack	
26297	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [s@macro-dialout-trunk:8] Set("PJSIP/985415-00003e5a", "DIAL_NUMBER=9407265349") in new stack	
26298	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [s@macro-dialout-trunk:18] Set("PJSIP/985415-00003e5a", "OUTNUM=9407265349") in new stack	
26299	[2025-03-27 08:53:44] VERBOSE[3642058][C-000016c6] pbx.c: Executing [s@macro-dialout-trunk:26] Set("PJSIP/985415-00003e5a", "__CRM_DESTINATION=9407265349") in new stack	
26300	[2025-03-27 08:53:45] VERBOSE[3642058][C-000016c6] pbx.c: Executing [s@macro-dialout-trunk:32] ExecIf("PJSIP/985415-00003e5a", "1?Set(CONNECTEDLINE(num,i)=9407265349)") in new stack	
26301	[2025-03-27 08:53:45] VERBOSE[3642058][C-000016c6] pbx.c: Executing [s@macro-dialout-trunk:38] Gosub("PJSIP/985415-00003e5a", "trunk-dial-with-exten,9407265349,1()") in new stack	
26302	[2025-03-27 08:53:45] VERBOSE[3642058][C-000016c6] pbx.c: Executing [9407265349@trunk-dial-with-exten:1] Dial("PJSIP/985415-00003e5a", "PJSIP/9407265349@Nortex_SIP,300,Tb(func-apply-sipheaders^s^1,(4))") in new stack	
26303	[2025-03-27 08:53:45] VERBOSE[3642058][C-000016c6] app_stack.c: Spawn extension (from-pstn, 9407265349, 1) exited non-zero on 'PJSIP/Nortex_SIP-00003e5b'	
26304	[2025-03-27 08:53:45] VERBOSE[3642058][C-000016c6] app_dial.c: Called PJSIP/9407265349@Nortex_SIP	
26305	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:1] Set("PJSIP/Nortex_SIP-00003e5c", "__DIRECTION=INBOUND") in new stack	
26306	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:2] Gosub("PJSIP/Nortex_SIP-00003e5c", "sub-record-check,s,1(in,9407265349,dontcare)") in new stack	
26307	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [in@sub-record-check:1] NoOp("PJSIP/Nortex_SIP-00003e5c", "Inbound Recording Check to 9407265349") in new stack	
26308	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [in@sub-record-check:4] Gosub("PJSIP/Nortex_SIP-00003e5c", "recordcheck,1(dontcare,in,9407265349)") in new stack	
26309	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:3] Set("PJSIP/Nortex_SIP-00003e5c", "CHANNEL(tonezone)=us") in new stack	
26310	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:4] Set("PJSIP/Nortex_SIP-00003e5c", "__FROM_DID=9407265349") in new stack	
26311	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:5] Set("PJSIP/Nortex_SIP-00003e5c", "returnhere=1") in new stack	
26312	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:6] Gosub("PJSIP/Nortex_SIP-00003e5c", "app-blacklist-check,s,1()") in new stack	
26313	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:7] Set("PJSIP/Nortex_SIP-00003e5c", "CDR(did)=9407265349") in new stack	
26314	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:8] GotoIf("PJSIP/Nortex_SIP-00003e5c", "0?") in new stack	
26315	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:9] ExecIf("PJSIP/Nortex_SIP-00003e5c", "0 ?Set(CALLERID(name)=9407263276)") in new stack	
26316	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:10] Set("PJSIP/Nortex_SIP-00003e5c", "__MOHCLASS=") in new stack	
26317	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:11] Set("PJSIP/Nortex_SIP-00003e5c", "__REVERSAL_REJECT=FALSE") in new stack	
26318	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:12] GotoIf("PJSIP/Nortex_SIP-00003e5c", "1?post-reverse-charge") in new stack	
26319	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx_builtins.c: Goto (from-pstn,9407265349,14)	
26320	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:14] NoOp("PJSIP/Nortex_SIP-00003e5c", "") in new stack	
26321	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:15] Set("PJSIP/Nortex_SIP-00003e5c", "__CALLINGNAMEPRES_SV=allowed_not_screened") in new stack	
26322	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:16] Set("PJSIP/Nortex_SIP-00003e5c", "__CALLINGNUMPRES_SV=allowed_not_screened") in new stack	
26323	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:17] Set("PJSIP/Nortex_SIP-00003e5c", "CALLERID(name-pres)=allowed_not_screened") in new stack	
26324	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:18] Set("PJSIP/Nortex_SIP-00003e5c", "CALLERID(num-pres)=allowed_not_screened") in new stack	
26325	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:19] NoOp("PJSIP/Nortex_SIP-00003e5c", "CallerID Entry Point") in new stack	
26326	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:20] Set("PJSIP/Nortex_SIP-00003e5c", "__CRM_DIRECTION=INBOUND") in new stack	
26327	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:21] Set("PJSIP/Nortex_SIP-00003e5c", "__CRM_SOURCE=9407263276") in new stack	
26328	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:22] Set("PJSIP/Nortex_SIP-00003e5c", "__CRM_LINKEDID=1743083625.33547") in new stack	
26329	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:23] AGI("PJSIP/Nortex_SIP-00003e5c", "agi://127.0.0.1/sangomacrm.agi,true") in new stack	
26330	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:24] ExecIf("PJSIP/Nortex_SIP-00003e5c", "1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack	
26331	[2025-03-27 08:53:45] VERBOSE[3642062][C-000016c7] pbx.c: Executing [9407265349@from-pstn:25] Goto("PJSIP/Nortex_SIP-00003e5c", "timeconditions,5,1") in new stack	
27205	[2025-03-27 08:53:57] VERBOSE[3642058][C-000016c6] pbx.c: Spawn extension (trunk-dial-with-exten, 9407265349, 1) exited non-zero on 'PJSIP/985415-00003e5a'	
27206	[2025-03-27 08:53:57] VERBOSE[3642058][C-000016c6] app_stack.c: Spawn extension (trunk-dial-with-exten, 9407265349, 1) exited non-zero on 'PJSIP/985415-00003e5a'

Stewart1 · March 27, 2025, 6:42pm

If the PBX is behind a NAT, in your router/firewall, confirm that the RTP port range (default is UDP 10000-20000) is forwarded from any IP to the private address of the PBX.

mcelsie · March 27, 2025, 8:51pm

Hello @bah12 - Grab a pcap using the command below or if you have system admin pro then use the packet capture feature. Start the capture then make the call in question, verify there is no audio and then end the call. Once the call is over then stop the trace and DM this to me. To do this put the file up on dropbox/google drive or something like this and send me a link.

Command: tcpdump -i any -s 0 -w sip-trace.pcap

bah12 · March 27, 2025, 9:03pm

Will do tomorrow I’ve left for the day.

bah12 · March 31, 2025, 2:44pm

I sent you the pcap, and the ports are forwarded but not from any IP just the fqdn of the trunks (2 sip stations trunks and one local ISP). I did change it to Any but no luck.

trunk1.freepbx.com
trunk2.freepbx.com

bah12 · April 1, 2025, 1:43pm

Thank you for the help, the “Force Answer” fixed the issue. For others facing this, below is a more detailed answer from the dm.

Thank you for this file, I looked it over and I can see the PBX is getting RTP but not sending any RTP out, this is why there is no audio.

To see the RTP packets I used the filter “udp.port == 17430 || udp.port == 14518”. The ports here are from our SDPs sent in each leg of the call. Now when you run this filter you can see only inbound RTP and nothing outbound. But we also see all packets are comfort noise and not a typical RTP audio packet.

Comfort noise is sent rather than sending an RTP packet that has silence in it. The purpose is the comfort noise packet should have a smaller payload and save bandwidth during times of silence on the call.

Now the strange part here are no normal packets, all of them are comfort noise. This seems to be creating an issue preventing Asterisk from forwarding the RTP. This could be a bug.

Since it is very weird behaviour to send comfort noise only packets, then I would say its probably best to address that rather then file an Asterisk bug.

A good work around to this would be to edit the inbound route in FreePBX and go to the Advanced tab, then set “Force Answer” to YES.

This will make the PBX pick up the call and it should start sending a ring back tone when forwarding the call. This should change up the media flow enough to resolve the issue.

system · April 8, 2025, 1:44pm

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