[Freeswitch-users] PCAP help and many call failures PLEASE HELP

[mayamatakeshi]

On Thu, Feb 24, 2022 at 12:27 AM Sean Devoy <sdevoy at 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 at 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 at 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 at 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=$%7boutbound_caller_id_number%7d at 64.136.173.30>REDACTED/>
>
>       <action application="set" data="
> sip_from_user=${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 at 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
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