[Freeswitch-users] PCAP help and many call failures PLEASE HELP
David Villasmil
david.villasmil.work at gmail.com
Thu Feb 24 00:33:27 UTC 2022
try adding
<action application="set" data="ignore_early_media=true"/>
right before the bridge.
Regards,
David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
On Wed, Feb 23, 2022 at 10:46 PM mayamatakeshi <mayamatakeshi at gmail.com>
wrote:
>
>
> 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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>
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