Toooooo early... call droped after 11 seconds now... different log.<br><br>[Mar 17 22:19:17] DEBUG[2783] chan_sip.c: SIP TIMER: Rescheduling retransmission #13781 (6) SIP/2.0 - 1<br>[Mar 17 22:19:17] DEBUG[2783] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 100 ms (Retrans id #13781))<br>
[Mar 17 22:19:20] DEBUG[4459] rtp.c: Got RTCP report of 176 bytes<br>[Mar 17 22:19:21] WARNING[2783] chan_sip.c: Maximum retries exceeded on transmission <a href="mailto:6b24c4992c31e3b264e271ba29f33db4@200.229.195.226">6b24c4992c31e3b264e271ba29f33db4@200.229.195.226</a> for seqno 102 (Critical Response)<br>
[Mar 17 22:19:21] DEBUG[2783] chan_sip.c: Setting SIP_ALREADYGONE on dialog <a href="mailto:6b24c4992c31e3b264e271ba29f33db4@200.229.195.226">6b24c4992c31e3b264e271ba29f33db4@200.229.195.226</a><br>[Mar 17 22:19:21] WARNING[2783] chan_sip.c: Hanging up call <a href="mailto:6b24c4992c31e3b264e271ba29f33db4@200.229.195.226">6b24c4992c31e3b264e271ba29f33db4@200.229.195.226</a> - no reply to our critical packet.<br>
[Mar 17 22:19:21] DEBUG[4459] channel.c: Didn't get a frame from channel: SIP/7977529-081931c0<br>[Mar 17 22:19:21] DEBUG[4459] channel.c: Bridge stops bridging channels SIP/7977529-081931c0 and SIP/242-081910e8<br>[Mar 17 22:19:21] DEBUG[4459] channel.c: Hanging up channel 'SIP/242-081910e8'<br>
[Mar 17 22:19:21] DEBUG[4459] chan_sip.c: Hangup call SIP/242-081910e8, SIP callid <a href="mailto:6fa30fba46a0cd93213baeab361530b8@192.168.20.249">6fa30fba46a0cd93213baeab361530b8@192.168.20.249</a>)<br>[Mar 17 22:19:21] DEBUG[4459] chan_sip.c: Strict routing enforced for session <a href="mailto:6fa30fba46a0cd93213baeab361530b8@192.168.20.249">6fa30fba46a0cd93213baeab361530b8@192.168.20.249</a><br>
[Mar 17 22:19:21] DEBUG[4459] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1<br><br>and not a clue.<br><br>THanks alot<br><br><div class="gmail_quote">On Wed, Mar 17, 2010 at 10:05 PM, Bruno Camargo <span dir="ltr"><<a href="mailto:mustardahc@gmail.com">mustardahc@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Hello Gentleman,<br><br>I guess the problem was the codec.<br><br>I have allowed only g711u for testing purposes and the incoming call endured for 1 minute, until the caller hanged.<br>
<br>Thanks a lot for the support.... but there are tons of questions yet to be answered!<br>
<br>Thanks<div><div></div><div class="h5"><br><br><div class="gmail_quote">On Wed, Mar 17, 2010 at 2:12 PM, Fred Posner <span dir="ltr"><<a href="mailto:fred@teamforrest.com" target="_blank">fred@teamforrest.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div>On Mar 17, 2010, at 1:05 PM, Bruno Camargo wrote:<br>
<br>
> Hi Giorgio,<br>
><br>
> So it means that Asterisk has no native support for g729 ?<br>
><br>
> Thanks<br>
><br>
</div>> --<br>
> BrCaBadT<br>
> --<br>
<br>
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.<br>
<br>
<br>
---fred<br>
<a href="http://qxork.com" target="_blank">http://qxork.com</a><br>
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