most likely it's because during the time you are dong artificial ringback the other side is not doing RTP right.<br><br>When the call is answered we flush the rtp buffer and your missing audio is probably flushed with it.<br>
so you can choose to have a 3 second delay or erase the 3 seconds as it does now.<br><br>This is a known problem with sonus which has been proven to build up an audio delay during the time<br>you are waiting for the call to answer. I'm sure you prefer the way it is to a large audio delay.<br>
<br><br><br><div class="gmail_quote">On Wed, Dec 3, 2008 at 10:46 PM, Angel Carpintero <span dir="ltr"><<a href="mailto:ack@telefonica.net">ack@telefonica.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
No TDM , all is SIP :<br>
<br>
<br>
PSTN ---> Sip Proxy_A --> FS ( brigde ) ringback/transfer_ringback<br>
-> Sip Proxy_B --> PSTN<br>
<br>
<br>
In logfile i think you can get some details about Media Gateways<br>
( Sonus ) PSTN inbound / outbound is provided by Level3.<br>
<br>
I can get a capture of a call if you want, in capture the audio is not<br>
missing, issue with :<br>
<br>
- rtp buffer ?<br>
- Sonus ?<br>
<br>
Let me know anything you need so i can provide a log or create a new<br>
scenario.<br>
<br>
<br>
Thanks,<br>
<br>
El mié, 03-12-2008 a las 22:12 -0600, Anthony Minessale escribió:<br>
<div><div></div><div class="Wj3C7c">> what does PSTN represent?<br>
><br>
> I know what the PSTN is but how are you reaching it?<br>
> is it TDM, SIP etc... what gateway type other details.<br>
><br>
><br>
> On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero <<a href="mailto:ack@telefonica.net">ack@telefonica.net</a>><br>
> wrote:<br>
> Hi guys,<br>
><br>
> I've a strange issue with FS , version svn -r10584 ,<br>
> when FS bridges a call first 3 seconds of audio are missing ,<br>
> looks that<br>
> only happens on PSTN calls and using ringback or<br>
> transfer_ringback. This<br>
> only happens in calls from PSTN , not from VOIP. Some<br>
> scenarios i tried<br>
> to isolate this issue :<br>
><br>
><br>
> - Issue<br>
><br>
> PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN<br>
><br>
> - Good setting bypass_media before run bridge but i need rtp<br>
> in FS path<br>
><br>
> PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN<br>
><br>
> - Good<br>
><br>
> PSTN --> FS ( brigde ) WITHOUT ringback/transfer_ringback -><br>
> PSTN<br>
><br>
> - Always good<br>
><br>
> VOIP --> FS ( brigde ) -> PSTN<br>
><br>
><br>
> Dialplan has nothing wrong ( i guess ):<br>
><br>
> <extension name="Transfers"><br>
> <condition field="destination_number"<br>
> expression="^1??XXXXXXXXXX$"><br>
> <action application="answer"/><br>
> <action application="speak" data="cepstral|Allison-8kHz|<br>
> blah"/><br>
> <action application="set"<br>
> data="hangup_after_bridge=false"/><br>
> <action application="set" data="playback_terminators=#"/><br>
> <action application="set" data="ringback=$${us-ring}"/><br>
> <action application="set" data="transfer_ringback=<br>
> $${hold_music}"/><br>
> <action application="set" data="effective_caller_id_name=<br>
> ${caller_id_name}"/><br>
> <action application="set"<br>
> data="effective_caller_id_number=<br>
> ${caller_id_number}"/><br>
> <action application="set" data="originate_timeout=30"/><br>
> <action application="set" data="call_timeout=30"/><br>
> <action application="bridge"<br>
> data="sofia/default/18008226235@PSTN_GW"/><br>
> <action application="speak" data="cepstral|Allison-8kHz|<br>
> Transfer<br>
> finished"/><br>
> <action application="hangup"/><br>
> </condition><br>
> </extension><br>
><br>
><br>
><br>
> Any ideas ?<br>
><br>
> Attached log of FS ( F8 from console ).<br>
><br>
><br>
> Thanks in advance !<br>
><br>
> --<br>
> Angel Carpintero<br>
> ack ( at ) telefonica ( dot ) net<br>
><br>
> Key fingerprint = 3FD3 9C90 149E 7824 CECD 6BCF AC2C CA61<br>
> 6EF1 B90D<br>
><br>
><br>
><br>
</div></div><div class="Ih2E3d">> --<br>
> Anthony Minessale II<br>
><br>
> FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
> ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
><br>
> AIM: anthm<br>
> <a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>
> GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
> IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br>
><br>
> FreeSWITCH Developer Conference<br>
> <a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br>
> <a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
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> pstn:213-799-1400<br>
<br>
</div>--<br>
<div><div></div><div class="Wj3C7c">Angel Carpintero<br>
ack ( at ) telefonica ( dot ) net<br>
<br>
Key fingerprint = 3FD3 9C90 149E 7824 CECD 6BCF AC2C CA61 6EF1 B90D<br>
<br>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
<br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
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