search for: initreq

...Apr 11 15:32:48] VERBOSE[9226] app_dial.c: -- DAHDI/35-1 is making progress passing it to SIP/2052-000006fa [Apr 11 15:32:48] DEBUG[6893] chan_sip.c: Allocating new SIP dialog for 656de8c01fcfde12371cfaa41a6cc357 at 127.0.1.1 - OPTIONS (No RTP) [Apr 11 15:32:48] DEBUG[6893] chan_sip.c: Initializing initreq for method OPTIONS - callid 0ca5e0f16cc3027a450c5ce920189bc5 at 192.168.100.238 [Apr 11 15:32:48] DEBUG[6893] chan_sip.c: Stopping retransmission on ' 0ca5e0f16cc3027a450c5ce920189bc5 at 192.168.100.238' of Request 102: Match Found [Apr 11 15:32:49] DEBUG[30773] rtp.c: Got RTCP report of 76...

Minor update - use the following: > if (strcasecmp(data, > "x-Asterisk-Request-URI-pseudo-header")==0) > { > ast_copy_string(buf, p->initreq.rlPart2, len); > -----Original Message----- > From: Steve Langstaff > Sent: 23 October 2006 09:58 > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [asterisk-users] SIP_HEADER function; what names > are available? > > Looking at the...

...-3c26701519b8-5xxapzoav2u4 Fromtag: <no from tag> Totag: <no to tag> [Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35505 at RedEdelca [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no NAT) [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672b3f35a-dpd0zv11yegl [Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis request - 3c2672b3f35a-dpd0zv11yegl [...] [Feb 11 10:44:13] DEBUG[4659] chan_sip.c: INVITE part of call transfer. Replaces [pickup-3c26701519b8-5xxapzoav2u4] [Feb 11 10:44:...

...Apr 11 15:32:48] VERBOSE[9226] app_dial.c: -- DAHDI/35-1 is making progress passing it to SIP/2052-000006fa [Apr 11 15:32:48] DEBUG[6893] chan_sip.c: Allocating new SIP dialog for 656de8c01fcfde12371cfaa41a6cc357 at 127.0.1.1 - OPTIONS (No RTP) [Apr 11 15:32:48] DEBUG[6893] chan_sip.c: Initializing initreq for method OPTIONS - callid 0ca5e0f16cc3027a450c5ce920189bc5 at 192.168.100.238 [Apr 11 15:32:48] DEBUG[6893] chan_sip.c: Stopping retransmission on '0ca5e0f16cc3027a450c5ce920189bc5 at 192.168.100.238' of Request 102: Match Found [Apr 11 15:32:49] DEBUG[30773] rtp.c: Got RTCP report of 76...

...} else { /* Incoming call, not up */ const char *res; if (p->hangupcause && (res = hangup_cause2sip(p->hangupcause))) transmit_response_reliable(p, res, &p->initreq); else transmit_response_reliable(p, "603 Declined", &p->initreq); p->invitestate = INV_TERMINATED; Obviously this doesn't include cases where the URI is not found, the c...

...te: hci0 corrupted SCO packet" entries in kernel logs. Can anybody please help? Tks ++++++ 13:37:17 chan_sip.c: Allocating new SIP dialog for 42eb60ff04309999607e7eb97cc86c69 at 192.168.0.204 - OPTIONS (No RTP) 13:37:17 acl.c: Found IP address for this socket 13:37:17 chan_sip.c: Initializing initreq for method OPTIONS - callid 5bae8a561541036e45990a137366cb14 at 192.168.0.204 13:37:17 chan_sip.c: Trying to put 'OPTIONS si' onto UDP socket destined for 192.168.0.84:27928 13:37:17 chan_sip.c: Stopping retransmission on ' 5bae8a561541036e45990a137366cb14 at 192.168.0.204' of Reque...

...that changed SSRC for DTMF. (Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82) * Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. (Closes issue #18051. Reported by eeman. Patched, tested by twilson) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.39 Thank you for your continued support of Asterisk!

...that changed SSRC for DTMF. (Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82) * Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. (Closes issue #18051. Reported by eeman. Patched, tested by twilson) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.16 Thank you for your continued support of Asterisk!

...that changed SSRC for DTMF. (Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82) * Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. (Closes issue #18051. Reported by eeman. Patched, tested by twilson) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.39 Thank you for your continued support of Asterisk!

...that changed SSRC for DTMF. (Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82) * Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. (Closes issue #18051. Reported by eeman. Patched, tested by twilson) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.16 Thank you for your continued support of Asterisk!

Running asterisk 1.2.7 with latest zaptel on centos4.4. with Aastra 55i phones. Local outbound calling works fine, but ATT requires clients enter 7 digit code for long distance. All calls with 7 digit code are lost within 20 seconds of the call. This is the message I?m getting: Apr 19 12:38:16 WARNING[9615]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission

...** Our prefcodec: (alaw) [Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: -- Done with adding codecs to SDP [Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw|gsm) [Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Initializing initreq for method INVITE - callid 7568eb9e7c148e535166a89702423c3e at yyy.yyy.yyy.yy:5060 [Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for xxx.xxx.xxx.xxx:45731 [Jan 6 11:38:29] VERBOSE[5383][C-000473c5] app_dial.c: Called SIP/4120089...

...000/2 a=rtpmap:9 G722/8000/1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=setup:actpass a=ssrc:1181629171 cname:{dc854b06-da58-45b3-8185-bbc6a57746c0} <-------------> --- (13 headers 31 lines) --- [May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:25444 handle_request_invite: Initializing initreq for method INVITE - callid cf2990ba- 3f12-3d9e-adb6-52889c414ed3 Using INVITE request as basis request - cf2990ba-3f12-3d9e-adb6-52889c414ed3 Found peer 'vr1a882' for 'vr1a882' from 2.2.2.2:8558 [May 19 16:47:43] DEBUG[14160][C-00000007]: rtp_engine.c:421 ast_rtp_in...

Hello you mean while placing a video call ? What info am I looking for in the debug output ? Kind regards. J. On 21-04-17 12:28, Marcelo Terres wrote: > Did you try to activate DEBUG and set the verbosity to a higher level > (100?) to check what Asterisk tells you about? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at

I'm attempting to find where my extra long DTMF Tones are coming from. I'm dialing from my sip handset through my proxy to my Asterisk box which is my PSTN Gateway. I'm pressing 4 to select a menu and everything is fine. [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin '4' received on SIP/trunk-0a02dee0 [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin passthrough

I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100