On Thu, Feb 24, 2022 at 12:27 AM Sean Devoy <sdevoy@bizfocused.com> wrote:
I have debug level FS logs. There are so many lines it is difficult to follow a single call.
Do you have the logs related to the capture file? If yes, then search for the Call-IDs of the troubled call: Leg1: Call-ID: f0544e98-378154a6@192.168.2.111 Leg2: Call-ID: 26e3398f-0e97-123b-3b83-16f3d912a996 For example, I did a test call with Call-ID: 090abe39-188c-4f06-b371-e8ffe3d862f8 So I grepped the freeswitch.log file with: [root@lab002 ~]$ grep 090abe39-188c-4f06-b371-e8ffe3d862f8 /usr/local/freeswitch/log/freeswitch.log 4b860aaf-b481-4e26-8886-37321c2fc3f3 2022-02-24 06:49:17.131915 [INFO] sofia.c:10414 sofia/external_in/123412341234@test.com receiving invite from 127.0.0.1:5060 version: 1.10.7-dev git dd24113 2021-08-25 17:37:19Z 64bit call-id: 090abe39-188c-4f06-b371-e8ffe3d862f8 The 4b860aaf-b481-4e26-8886-37321c2fc3f3 above is the channel uuid (the unique identifier of the channel in freeswitch). The freeswitch log file will include that channel uuid in all log lines related to that channel. So you can grep your freeswitch.log file for it. If no clue about the problem is found, do the same for the other leg.
This happens are about 50% of ALL calls.
Here is the dial plan:
<extension name="ten_or eleven_digiti_dial" >
<condition field="destination_number" expression=" ^(?:1|\+1)?([2-9]\d\d[2-9]\d{6})$">
<action application="set" data="ringback=${us-ring}"/>
<action application="set" data=" effective_caller_id_number=${outbound_caller_id_number}"/>
<action application="set" data="ani=${outbound_caller_id_number}"/>
<action application="set" data=" origination_caller_id_number=${outbound_caller_id_number}"/>
<action application="set" data="effective_caller_id_name=REDACTED"/>
<action application="set" data= sip_from_uri=${outbound_caller_id_number}@ <sip_from_uri=$%7boutbound_caller_id_number%7d@64.136.173.30>REDACTED/>
<action application="set" data=" sip_from_user=${outbound_caller_id_number}"/>
<action application="set" data="sip_from_host=REDACTED"/>
<action application="set" data="sip_h_X-CID=${sip_call_id}"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="sip_h_X-CID=${sip_call_id}"/>
<action application="bridge_export" data=" hold_music=/etc/freeswitch/recordings/REDACTED-moh.wav"/>
<action application="bridge" data= {sip_from_uri=${outbound_caller_id_number}@REDACTED,sip_from_user=${outbound_caller_id_number},sip_from_host=REDACTED}sofia/gateway/voip-innovations-outbound/1$1 <%7bsip_from_uri=$%7boutbound_caller_id_number%7d@64.136.173.30,sip_from_user=$%7boutbound_caller_id_number%7d,sip_from_host=64.136.173.30%7dsofia/gateway/voip-innovations-outbound/1$1> />
</condition>
</extension>
OK. So there is nothing setting an answer timeout (so it would default to 60 seconds). The variable controlling it would be this one: https://freeswitch.org/confluence/display/FREESWITCH/call_timeout