Zulu Windows client registers, calls fail audio/rtp


#1

FreePBX 10.13.66-22
Zulu Module 13.0.56.16

The Windows Zulu client registers and I can place outgoing calls. The remote phone rings and I can answer. But audio fails.

8002 TCP is open and forwarded to FreePBX (the client registers). From the logs it looks like the RTP proxying is failing? I’m not expert at reading these. Partial log below.

Appreciate any assistance.
Output of asterisk -x “pjsip show transports”

Transport: 0.0.0.0-tcp tcp 0 0 0.0.0.0:5160
Transport: 0.0.0.0-tls tls 0 0 0.0.0.0:5061
Transport: 0.0.0.0-udp udp 0 0 0.0.0.0:5061
Transport: 0.0.0.0-ws ws 0 0 0.0.0.0:5060
Transport: 0.0.0.0-wss wss 0 0 0.0.0.0:5060

Call log:
[2020-05-01 11:07:21] VERBOSE[13320][C-00000003] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2020-05-01 11:07:21] VERBOSE[13320][C-00000003] res_agi.c: <PJSIP/90199-00000002>AGI Script sangomacrm.agi completed, returning 0
[2020-05-01 11:07:21] VERBOSE[13320][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:25] Set(“PJSIP/90199-00000002”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
[2020-05-01 11:07:21] VERBOSE[13320][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:26] NoOp(“PJSIP/90199-00000002”, “CRM Finished”) in new stack
[2020-05-01 11:07:21] VERBOSE[13320][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:27] GotoIf(“PJSIP/90199-00000002”, “0?bypass,1”) in new stack
[2020-05-01 11:07:21] VERBOSE[13320][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:28] ExecIf(“PJSIP/90199-00000002”, “1?Set(CONNECTEDLINE(num,i)=425yyyxxxx)”) in new stack
[2020-05-01 11:07:21] VERBOSE[13320][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:29] ExecIf(“PJSIP/90199-00000002”, “1?Set(CONNECTEDLINE(name,i)=CID:4256431805)”) in new stack
[2020-05-01 11:07:21] VERBOSE[13320][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:30] ExecIf(“PJSIP/90199-00000002”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)4256431805)”) in new stack
[2020-05-01 11:07:21] VERBOSE[13320][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:31] GotoIf(“PJSIP/90199-00000002”, “0?customtrunk”) in new stack
[2020-05-01 11:07:21] VERBOSE[13320][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:32] Dial(“PJSIP/90199-00000002”, “SIP/FreePBX/425yyyxxxx,300,Tt”) in new stack
[2020-05-01 11:07:21] VERBOSE[13320][C-00000003] netsock2.c: Using SIP RTP TOS bits 184
[2020-05-01 11:07:21] VERBOSE[13320][C-00000003] netsock2.c: Using SIP RTP CoS mark 5
[2020-05-01 11:07:21] VERBOSE[13320][C-00000003] app_dial.c: Called SIP/FreePBX/425yyyxxxx
[2020-05-01 11:07:23] VERBOSE[13320][C-00000003] app_dial.c: SIP/FreePBX-00000004 is making progress passing it to PJSIP/90199-00000002
[2020-05-01 11:07:25] VERBOSE[13320][C-00000003] app_dial.c: SIP/FreePBX-00000004 answered PJSIP/90199-00000002
[2020-05-01 11:07:25] VERBOSE[13359][C-00000003] bridge_channel.c: Channel SIP/FreePBX-00000004 joined ‘simple_bridge’ basic-bridge
[2020-05-01 11:07:25] VERBOSE[13320][C-00000003] bridge_channel.c: Channel PJSIP/90199-00000002 joined ‘simple_bridge’ basic-bridge
[2020-05-01 11:07:53] NOTICE[10287] res_pjsip_sdp_rtp.c: Disconnecting channel ‘PJSIP/90199-00000002’ for lack of RTP activity in 32 seconds
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] bridge_channel.c: Channel PJSIP/90199-00000002 left ‘simple_bridge’ basic-bridge
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] app_macro.c: Spawn extension (macro-dialout-trunk, s, 32) exited non-zero on ‘PJSIP/90199-00000002’ in macro ‘dialout-trunk’
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Spawn extension (from-internal, 425yyyxxxx, 7) exited non-zero on ‘PJSIP/90199-00000002’
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Executing [h@from-internal:1] Macro(“PJSIP/90199-00000002”, “hangupcall”) in new stack
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/90199-00000002”, “1?theend”) in new stack
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2020-05-01 11:07:53] VERBOSE[13359][C-00000003] bridge_channel.c: Channel SIP/FreePBX-00000004 left ‘simple_bridge’ basic-bridge
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/90199-00000002”, “0?Set(CDR(recordingfile)=)”) in new stack
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Executing [s@macro-hangupcall:4] Hangup(“PJSIP/90199-00000002”, “”) in new stack
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘PJSIP/90199-00000002’ in macro ‘hangupcall’
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/90199-00000002’
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] app_stack.c: PJSIP/90199-00000002 Internal Gosub(crm-hangup,s,1) start
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Executing [s@crm-hangup:1] NoOp(“PJSIP/90199-00000002”, “Sending Hangup to CRM”) in new stack
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Executing [s@crm-hangup:2] NoOp(“PJSIP/90199-00000002”, “HANGUP CAUSE: 44”) in new stack
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Executing [s@crm-hangup:3] ExecIf(“PJSIP/90199-00000002”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Executing [s@crm-hangup:4] NoOp(“PJSIP/90199-00000002”, “MASTER CHANNEL: 1588356441.40 = 1588356441.40”) in new stack
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Executing [s@crm-hangup:5] GotoIf(“PJSIP/90199-00000002”, “0?return”) in new stack
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Executing [s@crm-hangup:6] Set(“PJSIP/90199-00000002”, “__CRM_HANGUP=1”) in new stack
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Executing [s@crm-hangup:7] AGI(“PJSIP/90199-00000002”, “sangomacrm.agi”) in new stack
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] res_agi.c: <PJSIP/90199-00000002>AGI Script sangomacrm.agi completed, returning 0
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] pbx.c: Executing [s@crm-hangup:8] Return(“PJSIP/90199-00000002”, “”) in new stack
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/90199-00000002’
[2020-05-01 11:07:53] VERBOSE[13320][C-00000003] app_stack.c: PJSIP/90199-00000002 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=


#2

Changed the extension from ChanSIP to PJSIP. Same result.


(Angel Velasquez) #3

Did you try adding a STUN server?


(Shahin Nazir) #4

Hi @FreerPBXer
I can see you are using too ols PBX system. Pls first try to upgrade your PBX system minimum to FreePBX-14
https://wiki.freepbx.org/display/PPS/Upgrading+from+FreePBX+10.13.66+to+SNG7

Also i can recommended to you use Asterisk 16.9 last version.


#5

Did that about 15 minutes ago and it seems to be working. Hadn’t posted back yet because I wanted to do some additional testing, but a STUN server does seem to have fixed it.

Thank you.


#6

Thank you, Snazir. We aren’t 100% comfortable with the upgrade path yet. There have been quite a few problems reported with the “restore older system backup to new system” path. So we are waiting a bit longer before trying that for clients.


#7

Angel, that does seem to have resolved the issue, thank you.

In this deployment we have a mix of internal clients, external clients, and external clients connected via Windows SSTP VPN. I reviewed this page: https://wiki.freepbx.org/pages/viewpage.action?pageId=110003971

Is there a concise guide for configuring STUN, ICE Hosts, and ICE Blacklists? I seemed to have to add the local subnet to the ICE Blacklist for LAN-side clients to work, but that broke external clients (internet and VPN).

For now I removed the ICE Blacklist and added the FreePBX server as an ICE Host. External clients are working, but I’ve not re-tested internal.


(system) closed #8

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