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[ack at telefonica.net]

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


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[anthony.minessale at g...]

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 (ack@telefonica.net)> wrote:

[ack at telefonica.net]

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:

[anthony.minessale at g...]

most likely it's because during the time you are dong artificial ringback the other side is not doing RTP right.

When the call is answered we flush the rtp buffer and your missing audio is probably flushed with it.
so you can choose to have a 3 second delay or erase the 3 seconds as it does now.

This is a known problem with sonus which has been proven to build up an audio delay during the time
you are waiting for the call to answer. I'm sure you prefer the way it is to a large audio delay.



On Wed, Dec 3, 2008 at 10:46 PM, Angel Carpintero <ack@telefonica.net (ack@telefonica.net)> wrote: