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

4 Dec 2008

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

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