Toooooo early... call droped after 11 seconds now... different log. [Mar 17 22:19:17] DEBUG[2783] chan_sip.c: SIP TIMER: Rescheduling retransmission #13781 (6) SIP/2.0 - 1 [Mar 17 22:19:17] DEBUG[2783] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 100 ms (Retrans id #13781)) [Mar 17 22:19:20] DEBUG[4459] rtp.c: Got RTCP report of 176 bytes [Mar 17 22:19:21] WARNING[2783] chan_sip.c: Maximum retries exceeded on transmission [email protected] for seqno 102 (Critical Response) [Mar 17 22:19:21] DEBUG[2783] chan_sip.c: Setting SIP_ALREADYGONE on dialog [email protected] [Mar 17 22:19:21] WARNING[2783] chan_sip.c: Hanging up call [email protected] - no reply to our critical packet. [Mar 17 22:19:21] DEBUG[4459] channel.c: Didn't get a frame from channel: SIP/7977529-081931c0 [Mar 17 22:19:21] DEBUG[4459] channel.c: Bridge stops bridging channels SIP/7977529-081931c0 and SIP/242-081910e8 [Mar 17 22:19:21] DEBUG[4459] channel.c: Hanging up channel 'SIP/242-081910e8' [Mar 17 22:19:21] DEBUG[4459] chan_sip.c: Hangup call SIP/242-081910e8, SIP callid [email protected]) [Mar 17 22:19:21] DEBUG[4459] chan_sip.c: Strict routing enforced for session [email protected] [Mar 17 22:19:21] DEBUG[4459] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
and not a clue. THanks alot On Wed, Mar 17, 2010 at 10:05 PM, Bruno Camargo <[email protected]>wrote: > Hello Gentleman, > > I guess the problem was the codec. > > I have allowed only g711u for testing purposes and the incoming call > endured for 1 minute, until the caller hanged. > > Thanks a lot for the support.... but there are tons of questions yet to be > answered! > > Thanks > > > On Wed, Mar 17, 2010 at 2:12 PM, Fred Posner <[email protected]> wrote: > >> On Mar 17, 2010, at 1:05 PM, Bruno Camargo wrote: >> >> > Hi Giorgio, >> > >> > So it means that Asterisk has no native support for g729 ? >> > >> > Thanks >> > >> > -- >> > BrCaBadT >> > -- >> >> Depends on your definition of support. It supports passthrough... but if >> you're using it locally on a bridge on transcoding, you'll need to purchase >> a license. The codec itself is non-G729 compliant. >> >> >> ---fred >> http://qxork.com >> >> >> >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > BrCaBadT > -- BrCaBadT
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
