this is what i got on the full asterisk log at the time the call got dropped<br><br> 4134 [Jan 19 10:41:54] DEBUG[11136] chan_dahdi.c: Write returned -1 (Resource temporarily unavailable) on channel 4<br> 4135 [Jan 19 10:41:54] DEBUG[11136] chan_dahdi.c: Write returned -1 (Resource temporarily unavailable) on channel 4<br>
4136 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=51)<br> 4137 [Jan 19 10:41:54] DEBUG[11136] chan_dahdi.c: Write returned -1 (Resource temporarily unavailable) on channel 4<br>
4138 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=51)<br> 4139 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=51)<br>
4140 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=51)<br> 4141 [Jan 19 10:41:54] DEBUG[11129] rtp.c: Got RTCP report of 176 bytes<br> 4142 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=51)<br>
4143 [Jan 19 10:41:54] DEBUG[11132] audiohook.c: Audiohook 0xb6c52ef4 has stale audio in its factories. Flushing them both<br> 4144 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=51)<br>
4145 [Jan 19 10:41:54] DEBUG[11132] rtp.c: Got RTCP report of 208 bytes<br> 4146 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=51)<br> 4147 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=51)<br>
4148 [Jan 19 10:41:54] DEBUG[11129] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=51)<br> 4149 [Jan 19 10:41:54] DEBUG[11129] audiohook.c: Failed to get 160 samples from write factory 0xb6c53968<br>
4150 [Jan 19 10:41:54] DEBUG[6547] chan_sip.c: = No match Their Call ID: OWE1NDNkODE1OWIwMjI3Zjk0NjFlMmZlN2U0YjEzNmM. Their Tag 007b5a64 Our tag: as4de61d80<br> 4151 [Jan 19 10:41:54] DEBUG[6547] chan_sip.c: = Found Their Call ID: NDI5ODhlNDgxYzNhODM4ZTQ3YTg3ZGNiZTkwZWU0Zjg. Their Ta g 5573a07e Our tag: as355bacb9<br>
4152 [Jan 19 10:41:54] DEBUG[6547] chan_sip.c: **** Received BYE (8) - Command in SIP BYE<br> 4153 [Jan 19 10:41:54] DEBUG[6547] chan_sip.c: Setting SIP_ALREADYGONE on dialog NDI5ODhlNDgxYzNhODM4ZTQ3YTg3ZGNiZTkwZWU0Zj g.<br>
4154 [Jan 19 10:41:54] DEBUG[6547] chan_sip.c: Received bye, issuing owner hangup<br> 4155 [Jan 19 10:41:54] DEBUG[11132] channel.c: Didn't get a frame from channel: SIP/1011-0000007b<br> 4156 [Jan 19 10:41:54] DEBUG[11132] chan_dahdi.c: Requested indication 20 on channel DAHDI/5-1<br>
4157 [Jan 19 10:41:54] DEBUG[11132] channel.c: Bridge stops bridging channels SIP/1011-0000007b and DAHDI/5-1<br> 4158 [Jan 19 10:41:54] DEBUG[11132] pbx.c: Launching 'Macro'<br> 4159 [Jan 19 10:41:54] VERBOSE[11132] logger.c: -- Executing [h@macro-dialout-trunk:1] Macro("SIP/1011-0000007b", "hang upcall|") in new stack<br>
4160 [Jan 19 10:41:54] DEBUG[11132] pbx.c: Expression result is '1'<br> 4161 [Jan 19 10:41:54] DEBUG[11132] pbx.c: Launching 'GotoIf'<br> 4162 [Jan 19 10:41:54] VERBOSE[11132] logger.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1011-0000007b", "1?skip rg") in new stack<br>
4163 [Jan 19 10:41:54] VERBOSE[11132] logger.c: -- Goto (macro-hangupcall,s,4)<br><br><div class="gmail_quote">2010/1/19 Moises Silva <span dir="ltr"><<a href="mailto:moises.silva@gmail.com">moises.silva@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class="im">On Tue, Jan 19, 2010 at 11:46 AM, blackgecko <span dir="ltr"><<a href="mailto:blackgecko@gmail.com" target="_blank">blackgecko@gmail.com</a>></span> wrote:<br>
<div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
ive done test and the calls get hanged up after 30 seconds, i dont know if this helps to identify what the problem can be.<br></blockquote></div><br></div>The protocol file does not show any problems. You need to start debugging this at a higher layer (Asterisk) it seems to me Asterisk decides to hangup the call, and openr2 just go ahead and hangup the call using R2 signaling, so Asterisk logs are needed with full debug enabled in order to see where the hangup comes from.<br>
<font color="#888888">
<br>-- <br>Moises Silva<br>Senior Software Engineer<br>Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada<br>t. 1 905 474 1990 x 128 | e. <a href="mailto:moy@sangoma.com" target="_blank">moy@sangoma.com</a><br>
</font><br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-r2 mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-r2" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-r2</a><br></blockquote></div><br>