Random Dropped Calls - SIP Trunks?

The customer is complaining of random dropped calls. I found the call in the log file and will attach some of it below. Is it just me or does it appear the sip trunk is ending the call?

[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:13] Set(“SIP/113-00002f1e”, “OUTNUM=19419151418”) in new stack
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:14] Set(“SIP/113-00002f1e”, “custom=SIP/Outbound ThinQ”) in new stack
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:15] ExecIf(“SIP/113-00002f1e”, “1?Set(DIAL_TRUNK_OPTIONS=M(setmusic^keetons072816)Tt)”) in new stack
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:16] ExecIf(“SIP/113-00002f1e”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^keetons072816)TtM(confirm))”) in new stack
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:17] Macro(“SIP/113-00002f1e”, “dialout-trunk-predial-hook,”) in new stack
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:1] MacroExit(“SIP/113-00002f1e”, “”) in new stack
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:18] GotoIf(“SIP/113-00002f1e”, “0?bypass,1”) in new stack
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:19] ExecIf(“SIP/113-00002f1e”, “1?Set(CONNECTEDLINE(num,i)=19419151418)”) in new stack
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:20] ExecIf(“SIP/113-00002f1e”, “1?Set(CONNECTEDLINE(name,i)=CID:9417472995)”) in new stack
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:21] ExecIf(“SIP/113-00002f1e”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)9417472995)”) in new stack
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:22] GotoIf(“SIP/113-00002f1e”, “0?customtrunk”) in new stack
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:23] Dial(“SIP/113-00002f1e”, “SIP/Outbound ThinQ/19419151418,300,M(setmusic^keetons072816)Tt”) in new stack
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] netsock2.c: Using SIP RTP TOS bits 184
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] netsock2.c: Using SIP RTP CoS mark 5
[2016-12-02 09:21:37] VERBOSE[9135][C-00001bc7] app_dial.c: Called SIP/Outbound ThinQ/19419151418
[2016-12-02 09:21:38] VERBOSE[9135][C-00001bc7] app_dial.c: SIP/Outbound ThinQ-00002f1f is making progress passing it to SIP/113-00002f1e

[2016-12-02 09:21:48] VERBOSE[9135][C-00001bc7] app_dial.c: SIP/Outbound ThinQ-00002f1f answered SIP/113-00002f1e
[2016-12-02 09:21:48] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:1] Set(“SIP/Outbound ThinQ-00002f1f”, “CHANNEL(musicclass)=keetons072816”) in new stack
[2016-12-02 09:21:48] VERBOSE[9142][C-00001bc7] bridge_channel.c: Channel SIP/Outbound ThinQ-00002f1f joined ‘simple_bridge’ basic-bridge
[2016-12-02 09:21:48] VERBOSE[9135][C-00001bc7] bridge_channel.c: Channel SIP/113-00002f1e joined ‘simple_bridge’ basic-bridge
[2016-12-02 09:22:16] NOTICE[5302] chan_sip.c: Disconnecting call ‘SIP/Outbound ThinQ-00002f1f’ for lack of RTP activity in 11 seconds
[2016-12-02 09:22:16] VERBOSE[9142][C-00001bc7] bridge_channel.c: Channel SIP/Outbound ThinQ-00002f1f left ‘simple_bridge’ basic-bridge
[2016-12-02 09:22:16] VERBOSE[9135][C-00001bc7] bridge_channel.c: Channel SIP/113-00002f1e left ‘simple_bridge’ basic-bridge
[2016-12-02 09:22:16] VERBOSE[9135][C-00001bc7] app_macro.c: Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on ‘SIP/113-00002f1e’ in macro ‘dialout-trunk’
[2016-12-02 09:22:16] VERBOSE[9135][C-00001bc7] pbx.c: Spawn extension (from-internal, 9151418, 6) exited non-zero on ‘SIP/113-00002f1e’
[2016-12-02 09:22:16] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:1] Macro(“SIP/113-00002f1e”, “hangupcall”) in new stack
[2016-12-02 09:22:16] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:1] GotoIf(“SIP/113-00002f1e”, “1?theend”) in new stack
[2016-12-02 09:22:16] VERBOSE[9135][C-00001bc7] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2016-12-02 09:22:16] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:3] ExecIf(“SIP/113-00002f1e”, “0?Set(CDR(recordingfile)=)”) in new stack
[2016-12-02 09:22:16] VERBOSE[9135][C-00001bc7] pbx.c: Executing [[email protected]:4] Hangup(“SIP/113-00002f1e”, “”) in new stack
[2016-12-02 09:22:16] VERBOSE[9135][C-00001bc7] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/113-00002f1e’ in macro ‘hangupcall’
[2016-12-02 09:22:16] VERBOSE[9135][C-00001bc7] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/113-00002f1e’

This would be your answer. No RTP activity so trunk is dropping the call.

Possibly being caused by your MAX RTP configuration. This setting is in place to prevent hung channels.

To change this setting, go to the Asterisk SIP Settings under chan sip settings, under the MEDIA & RTP Settings section, you can see that the rtpholdtimeout you can adjust this higher or lower as needed.

Asterisk version?

Asterisk 13.10.0

I’m being told these dropped calls are mid conversation, so it’s hard to believe there is no RTP activity

I may be getting this confused with something else, but aren’t there a couple of RTP issues with Asterisk 13.10?

Hello @mvogel4949,

I am experiencing exactly the same issue with random drop calls exactly after this message of “…left ‘simple_bridge’ basic-bridge…”. Did you finally solve it or found the reason?

Your help will be very important…

Hi @leosoft,

This is a two year old thread.
I’d suggest creating a new topic, make sure to include your system information, proper description of your issue and a call trace of a failed call.

Wow, that was a long time ago. If I remember correctly we adjusted the UDP timeout in the firewall. Increased it from 30s to 60 or 120s