search for: chan_sip

Hello, CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ... grep -e "Error" -e "eventually" p-console chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Er...

...9021653-dc67 of format slin since our native format has changed to ulaw Oct 17 11:10:27 DEBUG[3088] rtp.c: Difference is 23528, ms is 2961 Oct 17 11:10:30 DEBUG[3088] rtp.c: Difference is 22416, ms is 2822 Oct 17 11:10:31 DEBUG[3088] rtp.c: Difference is 1920, ms is 260 Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Allocating new SIP dialog for 13587bb10611325917636b6a211c0182@192.168.0.10 - REGISTER (No RTP) Oct 17 11:10:35 DEBUG[3088] acl.c: ##### Testing 80.87.16.11 with 192.168.0.0 Oct 17 11:10:35 DEBUG[3088] chan_sip.c: Target address 80.87.16.11 is not local, substituting externip Oct 17 11:10:35...

...carrier. They get my registration, but they see that my call is interrupted before they can complete the connection. My Asterisk log shows that the call times out after the time (45s) specified in my dialplan Dial() command. What is wrong? [from /var/log/asterisk/full]: Jan 30 23:40:35 DEBUG[6245] chan_sip.c: Stopping retransmission on '24154c0d430e550821bda73c155cf573@82.165.187.196' of Request 102: Match Found Jan 30 23:40:44 DEBUG[6268] manager.c: Manager received command 'Command' Jan 30 23:40:44 DEBUG[6268] manager.c: Manager received command 'Command' Jan 30 23:40:44 DEB...

...'hassip' (on reload) at line 35. [May 27 15:52:33] WARNING[2306] chan_dahdi.c: Ignoring any changes to 'hasiax' (on reload) at line 39. [May 27 15:52:33] WARNING[2306] chan_dahdi.c: Ignoring any changes to 'hasmanager' (on reload) at line 47. [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users [May 27 15:52:33] WARNING[2306] chan_sip.c: !!! will...

Hello all, I downloaded and built the Asterisk v1.4.9 from the Debian Unstable repository on my Debian Etch GNU/Linux but when I checked the logs, I got some error messages from the chan_sip.c. You can find the logs below. # pwd /usr/src/debian/ # apt-get build-dep asterisk # exit $ cd /usr/src/debian/asterisk-1.4.9~dfsg/ $ debuild -us -uc ... ... ... - - - < s n i p > - - - Generating docs /usr/src/debian/asterisk-1.4.9~dfsg/apps/app_skel.c:19Warning: Unsupported xml/html tag...

...<sip:PPC998202@194.208.44.44:34560;user=phone> Call-ID: e62dffffcd4dffff@194.183.145.211 CSeq: 100 REGISTER Expires: 3600 User-Agent: Grandstream HT496 1.0.2.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 Jan 25 22:26:07 DEBUG[41042]: chan_sip.c:3318 parse_request: Header 0: REGISTER sip:194.208.44.43 SIP/2.0 (34) Jan 25 22:26:07 DEBUG[41042]: chan_sip.c:3318 parse_request: Header 1: Via: SIP/2.0/UDP 194.208.44.44:34560;branch=PPC998202.z9hG4bK97220000265fffff,SIP/2.0/UDP 194.183.145.211:5064;branch=z9hG4bK97220000265fffff (141) Jan 25 2...

...bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then 1 days. I need your help .... *CLI log:- * [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '"4411" <sip:4411 at 204.152.194.246>' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:20] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '"4411" <sip:4411 at 204...

....255 appended to acl for peer [Jul 12 19:08:58] DEBUG[11552] acl.c: ##### Testing 127.0.0.1 with 0.0.0.0 [Jul 12 19:08:58] DEBUG[11552] acl.c: ##### Testing 127.0.0.1 with 127.0.0.1 [Jul 12 19:08:58] DEBUG[11552] manager.c: Manager received command 'Originate' [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Asked to create a SIP channel with formats: 0x40 (slin) [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Setting NAT on RTP to Off [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: *** Our native formats...

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c version 1.249 (the version the patch was written for) and the latest as of today 1.258. Both work great. Open ports 5060 and your RTP range (found...

...y? Can this be backported to the 1.4 branch? This could be another good reason to upgrade to 1.6.0.16 after I do some good testing... I'm having this issue on asterisk 1.4.22, this is a quick grep for ERROR of the last lines of /var/log/asterisk/messages file: [Sep 23 10:51:45] ERROR[26172] chan_sip.c: We could NOT get the channel lock for SIP/TRUNK_SWITCH4PRI-ad66bcf8! [Sep 23 10:51:45] ERROR[26172] chan_sip.c: SIP transaction failed: 57d7bbb010e5b40105f0fe5b4364420b at 192.168.130.25 [Sep 23 10:51:45] ERROR[26172] chan_sip.c: We could NOT get the channel lock for SIP/TRUNK_SWITCH4PRI-ad66...

Calling my home numbers has always worked. Till now. The Asterisk CLI show the following : [Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to '<sip:092xx90xx at 85.xx.xx.xx>;tag=as5b139383' And after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 re...

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:6991 handle_request: Check for res for 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:1605 update_user_counter: is not a local user 2004-08-09 17:36:2...

Hello list when trying to set up webRTC communications with sipjs client package (tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file the following : DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 99.88.77.66... OK. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:9 IN IP4 0.0.0.0... UNSUPPORTED OR FAILED. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=candidate:4275385644 1 udp 2122260...

Hello list, I have nat=no and qualify=no in my sip peer definition and still my CLI is flooded with : [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (24ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerp...

Good evening from the CVS of the 2005/05/14 it's impossible to build asterisk* on a redhat 7.3 i get this at compile time chan_sip.c: In function `build_user': chan_sip.c:10007: parse error before `struct' chan_sip.c:10029: `userflags' undeclared (first use in this function) chan_sip.c:10029: (Each undeclared identifier is reported only once chan_sip.c:10029: for each function it appears in.) chan_sip.c:10029: `mas...

Say, I just picked this up on my messages! There are a whole host of these requests! Anyone know whow there people are? Is there a way to report them? Any suggestions as to how to block them? [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '"912" <sip:1 at 41.1.1.1>' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '"912" <sip:1 at 41.1.1.1>' failed for '184.106.217.112' - Wrong password [Aug...

...erforming firewall/routing roles. Outgoing and incoming calls working perfectly from the ATA (linksys pap2t) but not from asterisk, because it hangs up after 10 seconds. Some LOGS: [Mar 16 15:11:12] DEBUG[13311] acl.c: ##### Testing 192.168.20.113 with 192.168.20.0 [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 0: OPTIONS sip:241 at 192.168.20.113:15956;rinstance=542e2865b2c6abe1 SIP/2.0 (71) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.20.249:5060;branch=z9hG4bK29f64fe1;rport (65) [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 2: From: "asterisk" &l...

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 am attempting to update my Asterisk installation from 1.0 to the latest stable version. When I use CVS checkout, I am receiving the following messages on chan_sip.c: RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v retrieving revision 1.510.2.25 retrieving revision 1.510.2.27 Merging differences between 1.510.2.25 and 1.510.2.27 into chan_sip.c M asterisk/channels/chan_sip.c Then, when I "make install" the compilation errors out on chan_sip...

...null)", ...): Temporary failure in name resolution [Mar 21 09:57:04] WARNING[6748] acl.c: Unable to lookup 'pbx2.server.com' ----------------------------------------------------------------THEN ------------------------------------------------------------- [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer 'YYYY' is now UNREACHABLE! Last qualify: 10 [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer 'C1045' is now UNREACHABLE! Last qualify: 140 [Mar 21 09:57:04] NOTICE[6748] chan_sip.c: Peer 'XXXX' is now UNREACHABLE! Last qualify: 33 [Mar 21 09:57:15] NOTICE[6748] cha...