Thanks again Anthony ! You fixed the issue with DTMF i had reported : http://jira.freeswitch.org/browse/FSCORE-251 Chris Danielson added to Wiki a nice page collecting these issues with Sonus : http://wiki.freeswitch.org/wiki/RTP_Issues Cheers, El mié, 10-12-2008 a las 03:10 +0100, Angel Carpintero escribió:
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 <ack@telefonica.net> 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 <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 > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@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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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org 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
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