[Freeswitch-users] missing 3 seconds of audio on bridge calls

[Angel Carpintero]

No TDM , all is SIP :


PSTN ---> Sip Proxy_A --> FS ( brigde ) ringback/transfer_ringback 
-> Sip Proxy_B --> PSTN 


In logfile i think you can get some details about Media Gateways
( Sonus ) PSTN inbound / outbound is provided by Level3. 

I can get a capture of a call if you want, in capture the audio is not 
missing, issue with :

- rtp buffer ?
- Sonus ? 

Let me know anything you need so i can provide a log or create a new
scenario.


Thanks,

El mié, 03-12-2008 a las 22:12 -0600, Anthony Minessale escribió:
> what does PSTN represent? 
> 
> I know what the PSTN is but how are you reaching it?
> is it TDM, SIP etc... what gateway type other details.
> 
> 
> On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero <ack at telefonica.net>
> wrote:
>         Hi guys,
>         
>          I've a strange issue with FS , version svn -r10584 ,
>         when FS bridges a call first 3 seconds of audio are missing ,
>         looks that
>         only happens on PSTN calls and using ringback or
>         transfer_ringback. This
>         only happens in calls from PSTN , not from VOIP. Some
>         scenarios i tried
>         to isolate this issue :
>         
>         
>         - Issue
>         
>         PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
>         
>         - Good setting bypass_media before run bridge but i need rtp
>         in FS path
>         
>         PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
>         
>         - Good
>         
>         PSTN --> FS ( brigde ) WITHOUT ringback/transfer_ringback ->
>         PSTN
>         
>         - Always good
>         
>         VOIP --> FS ( brigde ) -> PSTN
>         
>         
>         Dialplan has nothing wrong ( i guess ):
>         
>         <extension name="Transfers">
>            <condition field="destination_number"
>         expression="^1??XXXXXXXXXX$">
>              <action application="answer"/>
>              <action application="speak" data="cepstral|Allison-8kHz|
>         blah"/>
>              <action application="set"
>         data="hangup_after_bridge=false"/>
>              <action application="set" data="playback_terminators=#"/>
>              <action application="set" data="ringback=$${us-ring}"/>
>              <action application="set" data="transfer_ringback=
>         $${hold_music}"/>
>              <action application="set" data="effective_caller_id_name=
>         ${caller_id_name}"/>
>              <action application="set"
>         data="effective_caller_id_number=
>         ${caller_id_number}"/>
>              <action application="set" data="originate_timeout=30"/>
>              <action application="set" data="call_timeout=30"/>
>              <action application="bridge"
>         data="sofia/default/18008226235 at PSTN_GW"/>
>              <action application="speak" data="cepstral|Allison-8kHz|
>         Transfer
>         finished"/>
>              <action application="hangup"/>
>            </condition>
>          </extension>
>         
>         
>         
>         Any ideas ?
>         
>         Attached log of FS ( F8 from console ).
>         
>         
>         Thanks in advance !
>         
>         --
>         Angel Carpintero
>         ack ( at ) telefonica ( dot ) net
>         
>         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61
>         6EF1 B90D
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
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> 
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> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400

-- 
Angel Carpintero
ack ( at ) telefonica ( dot ) net

Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D

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