Thanks Anthony , you did a great work ! this is fixed in svn r10691. Some notes for people using Sonus and L3 as was my case :
in var.xml in some scenario you may need : <X-PRE-PROCESS cmd="set" data="send_silence_when_idle=400"/> in sip_profiles/internal.xml : <param name="rtp-rewrite-timestamps" value="true"/> might help for some people with rtp issues : <param name="rtp-timer-name" value="none"/> If you have issues with DTMF and timestamps add also : <param name="pass-rfc2833" value="true"/> I've a little issues with DTMF from VOIP , i i'll figure out can could be the issue , from PSTN all works like a charm :) Cheers, El jue, 04-12-2008 a las 09:34 -0600, Anthony Minessale escribió: > 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 <[EMAIL PROTECTED]> > wrote: > 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 > <[EMAIL PROTECTED]> > > 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/[EMAIL PROTECTED]"/> > > <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:[EMAIL PROTECTED] > > GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:[EMAIL PROTECTED] > > iax:[EMAIL PROTECTED]/888 > > googletalk:[EMAIL PROTECTED] > > pstn:213-799-1400 > > > -- > > Angel Carpintero > ack ( at ) telefonica ( dot ) net > > Key fingerprint = 3FD3 9C90 149E 7824 CECD 6BCF AC2C CA61 > 6EF1 B90D > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:[EMAIL PROTECTED] > GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[EMAIL PROTECTED] > iax:[EMAIL PROTECTED]/888 > googletalk:[EMAIL PROTECTED] > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Angel Carpintero ack ( at ) telefonica ( dot ) net Key fingerprint = 3FD3 9C90 149E 7824 CECD 6BCF AC2C CA61 6EF1 B90D "No basta saber, hay que aplicar lo que se sabe; no basta querer hacerlas cosas, hay que hacerlas". "Knowing is not enough; we must apply. Willing is not enough; we must do" Johann Wolfgang von Goethe
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