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Losing sound on incoming/outgoing calls


(Uncleanon) #1

Hello,

I have a freepbx which is connected to a goip gsm gateway. Everything works as expected except that sometimes as people are in a call - irrespective of whether incoming (i.e gsm -> goip -> freepbx -> ip phone) or outgoing people on the internal phones stop hearing the other party. Sometimes this is intermittent other times the call has to be disconnected and either of the parties redial. I would like to debug the issue. Here is the asterisk log for one such outgoing call:

/var/log/asterisk/full:[2019-02-08 10:57:07] VERBOSE[23562][C-0000011b] app_stack.c: Spawn extension (from-trunk, 08xxxxx, 1) exited non-zero on 'SIP/goip-1-0000023b'
/var/log/asterisk/full:[2019-02-08 10:57:07] VERBOSE[23562][C-0000011b] app_stack.c: SIP/goip-1-0000023b Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
/var/log/asterisk/full:[2019-02-08 10:57:07] VERBOSE[23562][C-0000011b] app_dial.c: Called SIP/goip-1/208xxxxx
/var/log/asterisk/full:[2019-02-08 10:57:07] VERBOSE[23562][C-0000011b] app_dial.c: SIP/goip-1-0000023b is ringing
/var/log/asterisk/full:[2019-02-08 10:57:13] VERBOSE[23562][C-0000011b] app_dial.c: SIP/goip-1-0000023b is making progress passing it to SIP/1101-0000023a
/var/log/asterisk/full:[2019-02-08 10:57:21] VERBOSE[23562][C-0000011b] app_dial.c: SIP/goip-1-0000023b answered SIP/1101-0000023a
/var/log/asterisk/full:[2019-02-08 10:57:21] VERBOSE[23564][C-0000011b] bridge_channel.c: Channel SIP/goip-1-0000023b joined 'simple_bridge' basic-bridge <1611b223-63ba-49c4-9595-bf32437c1804>
/var/log/asterisk/full:[2019-02-08 10:57:21] VERBOSE[23562][C-0000011b] bridge_channel.c: Channel SIP/1101-0000023a joined 'simple_bridge' basic-bridge <1611b223-63ba-49c4-9595-bf32437c1804>
/var/log/asterisk/full:[2019-02-08 11:01:49] VERBOSE[23562][C-0000011b] bridge_channel.c: Channel SIP/1101-0000023a left 'simple_bridge' basic-bridge <1611b223-63ba-49c4-9595-bf32437c1804>
/var/log/asterisk/full:[2019-02-08 11:01:49] VERBOSE[23562][C-0000011b] app_macro.c: Spawn extension (macro-dialout-trunk, s, 25) exited non-zero on 'SIP/1101-0000023a' in macro 'dialout-trunk'
/var/log/asterisk/full:[2019-02-08 11:01:49] VERBOSE[23562][C-0000011b] pbx.c: Spawn extension (from-internal, 08xxxxx, 6) exited non-zero on 'SIP/1101-0000023a'
/var/log/asterisk/full:[2019-02-08 11:01:49] VERBOSE[23562][C-0000011b] pbx.c: Executing [h@from-internal:1] Macro("SIP/1101-0000023a", "hangupcall") in new stack
/var/log/asterisk/full:[2019-02-08 11:01:49] VERBOSE[23562][C-0000011b] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/1101-0000023a", "1?theend") in new stack
/var/log/asterisk/full:[2019-02-08 11:01:49] VERBOSE[23562][C-0000011b] pbx_builtins.c: Goto (macro-hangupcall,s,3)
/var/log/asterisk/full:[2019-02-08 11:01:49] VERBOSE[23564][C-0000011b] bridge_channel.c: Channel SIP/goip-1-0000023b left 'simple_bridge' basic-bridge <1611b223-63ba-49c4-9595-bf32437c1804>

So here the call starts at 10.57, people are conversing for about 4 minutes and in the last 20 seconds the person on the ip phone stopped hearing the person that dialled the gsm number. What I have ascertained so far:

  1. It doesn’t seem to be a lack of resource on the Freepbx machine - it’s a vm with 1gb of ram and 2gb of swap, with usually 200mb of free ram. There are 9 ip phones connected to it and one trunk - the GOIP. Looking at the CPU/MEMORY graphs around the times when a conversation experience the issue doesn’t show any speaks, the load avg is usually 0.06. The vm has 2 cores

  2. I don’t think this is a networking issue - I’ve enabled SIP Quality and the average latency to the phones is 30ms (they are old cisco 7941) and to the goip - 2ms. In order to fully exclude that I intend on connecting the goip directly to the vm since I have free network interface on the physical server where the vm resides. Currently the connection happens via switch.

  3. I thought it could be due to weak GSM signal, go ip shows RSSI of 23-25. I also had a 1 hour long gsm conversation with someone who was in the same room as the goip, on the same mobile operator with no drops (we were using actual physical mobile phones for that). So I doubt this could be, though it seems most likely.

I would like to ask for advice how to proceed to trouble shoot it? If it is a networking problem should I be seeing something in freepbx’s log that rtp streams or dropping or something else ?


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