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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400