Ok, that makes more sense. Try this new patch and let me know how it goes, once you confirm it works I will post it in my blog with a better name.
http://moythreads.com/testasync.diff Moy On Wed, Apr 15, 2009 at 11:52 AM, <cyr2...@gmail.com> wrote: > Hi Moy, > You are right. I failed applying the patch. In fact, I applied it but I > didn't "make install" so I started a wrong asterisk. I apologize, it was my > mistake. This time I made sure twice before getting the logs and this time > the log message you said appears, but it doesn't work either as you can see: > I'm copying the whole log from the originate action to the hangup: > > ===================================================================== > [Apr 15 13:01:22] DEBUG[25752]: manager.c:2108 process_message: Manager > received command 'originate' > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:15874 sip_request_call: Asked to > create a SIP channel with formats: 0x40 (slin) > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:4508 sip_alloc: Allocating new SIP > dialog for (No Call-ID) - INVITE (With RTP) > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:2740 do_setnat: Setting NAT on RTP > to Off > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:2745 do_setnat: Setting NAT on > VRTP to Off > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:2998 sip_call: Outgoing Call for > 501 > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:3013 sip_call: Our T38 capability > (0), joint T38 capability (0) > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:6423 add_sdp: ** Our capability: > 0x38000e (gsm|ulaw|alaw|h263|h263p|h264) Video flag: False > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:6424 add_sdp: ** Our prefcodec: > 0x40 (slin) > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:6439 add_sdp: This call needs > video offers! > [Apr 15 13:01:22] DEBUG[30207]: chan_sip.c:2215 __sip_semi_ack: (Provisional) > Stopping retransmission (but retaining packet) on > '5c2607ce7cda26537726b6a4323a3...@10.0.5.20' Request 102: Found > [Apr 15 13:01:22] DEBUG[30207]: chan_sip.c:2215 __sip_semi_ack: (Provisional) > Stopping retransmission (but retaining packet) on > '5c2607ce7cda26537726b6a4323a3...@10.0.5.20' Request 102: Found > [Apr 15 13:01:22] DEBUG[30207]: chan_sip.c:2157 __sip_ack: Acked pending > invite 102 > [Apr 15 13:01:22] DEBUG[30207]: chan_sip.c:2174 __sip_ack: Stopping > retransmission on '5c2607ce7cda26537726b6a4323a3...@10.0.5.20' of Request > 102: Match Not Found > [Apr 15 13:01:22] DEBUG[30207]: chan_sip.c:5470 process_sdp: We're settling > with these formats: 0x100008 (alaw|h263p) > [Apr 15 13:01:22] DEBUG[30207]: chan_sip.c:8236 build_route: build_route: > Contact hop: <sip:5...@10.0.2.151:5060;user=phone> > [Apr 15 13:01:22] > Channel SIP/501-0828df48 was answered. > [Apr 15 13:01:22] DEBUG[26934]: pbx.c:1831 pbx_extension_helper: Launching > 'NoOp' > [Apr 15 13:01:22] -- Executing [...@sip_sercom:1] > NoOp("SIP/501-0828df48", "entrada numeracion del 8 801") in new stack > [Apr 15 13:01:22] DEBUG[26934]: pbx.c:1831 pbx_extension_helper: Launching > 'AGI' > [Apr 15 13:01:22] -- Executing [...@sip_sercom:2] AGI("SIP/501-0828df48", > "agi:async") in new stack > [Apr 15 13:01:51] DEBUG[25752]: manager.c:2108 process_message: Manager > received command 'AGI' > [Apr 15 13:01:51] -- Playing 'es/demo-congrats' (escape_digits=1) > (sample_offset 0) > [Apr 15 13:01:51] DEBUG[26934]: rtp.c:2753 ast_rtp_write: Ooh, format changed > from unknown to alaw > [Apr 15 13:01:51] DEBUG[26934]: rtp.c:2770 ast_rtp_write: Created smoother: > format: 8 ms: 20 len: 160 > [Apr 15 13:01:51] DEBUG[26934]: channel.c:1793 ast_settimeout: Scheduling > timer at 160 sample intervals > [Apr 15 13:02:00] DEBUG[25752]: manager.c:2108 process_message: Manager > received command 'Redirect' > [Apr 15 13:02:00] DEBUG[25752]: channel.c:1378 ast_softhangup_nolock: > Soft-Hanging up channel 'SIP/501-0828df48' > [Apr 15 13:02:00] DEBUG[26934]: channel.c:1793 ast_settimeout: Scheduling > timer at 0 sample intervals > [Apr 15 13:02:00] DEBUG[26934]: res_agi.c:488 launch_asyncagi: > launch_asyncagi returned (0x2) for chan SIP/501-0828df48 > [Apr 15 13:02:00] DEBUG[26934]: pbx.c:2448 __ast_pbx_run: Extension 801, > priority 0 returned normally even though call was hung up > [Apr 15 13:02:00] DEBUG[26934]: channel.c:1378 ast_softhangup_nolock: > Soft-Hanging up channel 'SIP/501-0828df48' > [Apr 15 13:02:00] DEBUG[26934]: channel.c:1477 ast_hangup: Hanging up channel > 'SIP/501-0828df48' > [Apr 15 13:02:00] DEBUG[26934]: chan_sip.c:3485 sip_hangup: Hangup call > SIP/501-0828df48, SIP callid 5c2607ce7cda26537726b6a4323a3...@10.0.5.20) > ===================================================================== > > As it seemed the execution was exiting by a line of code without a log, I did > a bit modification to res_agi.c (some additional log line) and I was able to > find out the execution was exiting in the line 437 with the res variable > containing a -1: > > if (cmd) { > res = agi_handle_command(chan, &async_agi, > cmd->cmd_buffer); > if ((res < 0) || (res == AST_PBX_KEEPALIVE)) { > free_agi_cmd(cmd); > break; > > In order to discard any version issues, I installed a new one from scratch > and then applied the async-agi patch only, getting the same results. By the > way, I was also able to install an asterisk 1.6.0.9 with the same > configuration and dial plan like the 1.4.18 one and it worked fine. > > I hope this can be useful. > > Regards > Jose > > > -- Moises Silva wrote : > > I really think you did not recompile and reinstall after applying the > new patch. I don't see any code path where the message > > [Apr 13 12:03:57] DEBUG[2755]: res_agi.c:464 launch_asyncagi: No frame > read on channel SIP/501-08279028, going out ... > > Is displayed but then > > ast_log(LOG_DEBUG, "launch_asyncagi returned (0x%X) for chan %s\n", > returnstatus, chan->name); > > is NOT displayed. In fact, there is no way you can get out of > launch_asyncagi without displaying that message. I tested this with > 1.4.18 version exactly. > > The fact that works for some people and not for others may be due to > different asterisk versions and/or dial plan specific issues. > > Please make sure the patch was correctly applied, once that is done we > can try some other things. > > > -- > This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com > http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/11929418.html > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- "I do not agree with what you have to say, but I’ll defend to the death your right to say it." Voltaire _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
