| Summary: | ASTERISK-10331: NOTIFY contains invalid To header | ||
| Reporter: | goestelecom (goestelecom) | Labels: | |
| Date Opened: | 2007-09-18 14:57:44 | Date Closed: | 2007-11-09 10:34:34.000-0600 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/Subscriptions |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) invalid_notify.txt | |
| Description: | In response to a SUBSCRIBE request the NOTIFY that is generated contains a To header that starts: "To: <sip:sip:..." ****** ADDITIONAL INFORMATION ****** Some of the names have been changed to protect the innocent. 1.1.1.28:5060 is Asterisk 2.2.2.253 is the Firewall/NAT 192.168.1.190 is the SPA9000 generating the SUBSCRIBE # U 1.1.1.28:5060 -> 2.2.2.253:1071 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.190:5060;branch=z9hG4bK-55230554;received=2.2.2.253. From: "" <sip:2125551212@example.com>;tag=6895d61f9ec38e11. To: "" <sip:2125551212@example.com>;tag=as0b9c6e4f. Call-ID: df4669b7-cdf64f59@192.168.1.190. CSeq: 59282 SUBSCRIBE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Expires: 60. Contact: <sip:8400@1.1.1.28:5060>;expires=60. Content-Length: 0. . # U 1.1.1.28:5060 -> 2.2.2.253:1071 NOTIFY sip:2125551212@192.168.1.190:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.28:5060;branch=z9hG4bK545a76d8;rport. Max-Forwards: 70. From: "asterisk" <sip:asterisk@example.com>;tag=as65a60c16. To: <sip:sip:2125551212@192.168.1.190:5060>;tag=6895d61f9ec38e11. Contact: <sip:asterisk@1.1.1.28:5060>. Call-ID: df4669b7-cdf64f59@192.168.1.190. CSeq: 104 NOTIFY. User-Agent: Asterisk PBX. Event: message-summary. Content-Type: application/simple-message-summary. Subscription-State: active. Content-Length: 87. . Messages-Waiting: yes. Message-Account: sip:101@example.com. Voice-Message: 4/0. | ||
| Comments: | By: Joshua C. Colp (jcolp) 2007-09-19 08:22:09 Please attach a sip debug from Asterisk with the entire dialog (SUBSCRIBE and NOTIFY). By: mlegas (mlegas) 2007-09-25 04:55:09 this happens too on my production system on 1.4.11 By: ibercom (ibercom) 2007-10-11 12:36:54 This happens too on my production system on 1.4.11. When Thomson ST2030 receive the wrong NOTIFY it reply: <------------> Reliably Transmitting (no NAT) to 172.xx.xx.xx:5060: NOTIFY sip:20012@172.xx.xx.xx:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.ss.ss.ss:5060;branch=z9hG4bK008b7684;rport From: "asterisk" <sip:asterisk@domain:5060>;tag=as0f1fe753 To: <sip:sip:20012@172.xx.xx.xx:5060;user=phone>;tag=c0a80101-2255 Contact: <sip:asterisk@172.ss.ss.ss> Call-ID: 446f-c0a80101-d-2@172.xx.xx.xx CSeq: 121 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 87 Messages-Waiting: yes Message-Account: sip:*99@domain:5060 Voice-Message: 2/0 (0/0) --- <--- SIP read from 172.xx.xx.xx:5060 ---> SIP/2.0 481 CallLeg/Transaction Does Not Exist Via: SIP/2.0/UDP 172.ss.ss.ss:5060;branch=z9hG4bK008b7684;rport From: "asterisk"<sip:asterisk@domain:5060>;tag=as0f1fe753 To: <sip:sip@172.xx.xx.xx:5060;user=phone>;tag=c0a80101-2255 Call-ID: 446f-c0a80101-d-2@172.xx.xx.xx CSeq: 121 NOTIFY Content-Length: 0 <-------------> And Asterisk log: [Oct 11 19:44:50] WARNING[21869] chan_sip.c: Remote host can't match request NOTIFY to call '446f-c0a80101-d-2@172.19.11.226'. Giving up. By: Digium Subversion (svnbot) 2007-11-07 20:24:51.000-0600 Repository: asterisk Revision: 89101 U branches/1.4/channels/chan_sip.c ------------------------------------------------------------------------ r89101 | file | 2007-11-07 20:24:50 -0600 (Wed, 07 Nov 2007) | 4 lines Do not add a sip: to the beginning of the To URI unless needed. (closes issue ASTERISK-10331) Reported by: goestelecom ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2007-11-07 20:26:18.000-0600 Repository: asterisk Revision: 89102 _U trunk/ U trunk/channels/chan_sip.c ------------------------------------------------------------------------ r89102 | file | 2007-11-07 20:26:18 -0600 (Wed, 07 Nov 2007) | 12 lines Merged revisions 89101 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4 lines Do not add a sip: to the beginning of the To URI unless needed. (closes issue ASTERISK-10331) Reported by: goestelecom ........ ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2007-11-09 10:34:34.000-0600 Repository: asterisk Revision: 89131 _U team/file/t38fun/ U team/file/t38fun/apps/app_queue.c U team/file/t38fun/apps/app_voicemail.c U team/file/t38fun/cdr/cdr_tds.c U team/file/t38fun/channels/chan_sip.c U team/file/t38fun/codecs/codec_zap.c U team/file/t38fun/configs/extensions.ael.sample U team/file/t38fun/configs/res_odbc.conf.sample U team/file/t38fun/doc/valgrind.txt U team/file/t38fun/include/asterisk/lock.h U team/file/t38fun/main/config.c U team/file/t38fun/main/manager.c U team/file/t38fun/main/say.c U team/file/t38fun/main/srv.c U team/file/t38fun/main/tdd.c U team/file/t38fun/pbx/pbx_ael.c U team/file/t38fun/res/res_jabber.c U team/file/t38fun/res/res_musiconhold.c ------------------------------------------------------------------------ r89131 | file | 2007-11-09 10:34:31 -0600 (Fri, 09 Nov 2007) | 168 lines Merged revisions 89032,89036-89037,89042,89045-89046,89053,89079,89085,89088,89090,89093,89095,89097,89099,89101,89103,89105,89111,89115,89119,89125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4 lines Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable. (closes issue ASTERISK-10551) Reported by: pj ........ r89036 | murf | 2007-11-06 13:52:50 -0400 (Tue, 06 Nov 2007) | 1 line closes issue ASTERISK-8547 - where the [catname](!) and [catname](othercat1,othercat2,...) notation gets dropped across a ConfigUpdate (or any other thing that would cause a config file to be written). While I was at it, I also cleaned up some of the destroy routines to free up comments, which was not being done. Made sure the new struct I introduced is also cleaned up properly at destruction time. My code handles multiple template inclusions. Many thanks to ssokol for his patch, which, while not literally used in the final merge, served as a foundation for the fix. ........ r89037 | russell | 2007-11-06 14:20:07 -0400 (Tue, 06 Nov 2007) | 11 lines If someone were to delete the files used by an existing MOH class, and then issue a reload, further use of that class could result in a crash due to dividing by zero. This set of changes fixes up some places to prevent this from happening. (closes issue ASTERISK-10500) Reported by: jcomellas Patches: res_musiconhold_division_by_zero.patch uploaded by jcomellas (license 282) Additional changes added by me. ........ r89042 | oej | 2007-11-06 14:53:37 -0400 (Tue, 06 Nov 2007) | 2 lines Bug fixes to tdd support in zaptel. ........ r89045 | tilghman | 2007-11-06 15:09:06 -0400 (Tue, 06 Nov 2007) | 2 lines We went to the trouble of creating a method of tracking failed trylocks, then never turned it on (oops). ........ r89046 | qwell | 2007-11-06 15:09:30 -0400 (Tue, 06 Nov 2007) | 4 lines Correctly set the total number of channels from a zaptel transcoder board. SPD-49, patch by Matthew Nicholson. ........ r89053 | russell | 2007-11-06 16:18:49 -0400 (Tue, 06 Nov 2007) | 3 lines Fix init_classes() so that classes that actually do have files loaded aren't treated as empty, and immediately destroyed ... ........ r89079 | tilghman | 2007-11-07 00:07:49 -0400 (Wed, 07 Nov 2007) | 5 lines Suppress AEL warnings on load. Reported by: eliel Patch by: eliel Closes issue ASTERISK-10703 ........ r89085 | mmichelson | 2007-11-07 11:56:49 -0400 (Wed, 07 Nov 2007) | 5 lines Fixing a segfault in the manager "core show channels concise" command. (closes issue ASTERISK-10708, reported by arnd and patched by ys) ........ r89088 | murf | 2007-11-07 17:40:28 -0400 (Wed, 07 Nov 2007) | 1 line In response to 10578, I just ran 1.4 thru valgrind; some of the config leakage I've already fixed, but it doesn't hurt to double check. I found and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major, tho. ........ r89090 | mmichelson | 2007-11-07 18:40:35 -0400 (Wed, 07 Nov 2007) | 6 lines This patch makes it possible for SIP phones to dial extensions defined with '#' characters in extensions.conf AND maintain their escaped characters when forming URI's (closes issue ASTERISK-10265, reported by cahen, patched by me, code review by file) ........ r89093 | tilghman | 2007-11-07 19:39:37 -0400 (Wed, 07 Nov 2007) | 7 lines The member refcount must be incremented, to avoid using it after deallocation. A huge thanks go to lvl- for patiently providing the necessary valgrind output that was necessary to finding this problem of memory corruption. Reported by: lvl- Patch by: tilghman Closes issue ASTERISK-10699 ........ r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4 lines If callerid is configured in sip.conf use that for checking the presence of an extension in the dialplan. (closes issue ASTERISK-10710) Reported by: spditner ........ r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8 lines Add support for allowing one outgoing transaction. This means if a response comes back out of order chan_sip will still handle it. I dream of a chan_sip with real transaction support. (closes issue ASTERISK-10498) Reported by: flefoll (closes issue ASTERISK-10472) Reported by: ramonpeek (closes issue ASTERISK-9288) Reported by: atca_pres ........ r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6 lines Improve the devicestate logic for multiple devices. If any are available then the extension is considered available. (closes issue ASTERISK-9843) Reported by: nic_bellamy Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299) ........ r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4 lines Do not add a sip: to the beginning of the To URI unless needed. (closes issue ASTERISK-10331) Reported by: goestelecom ........ r89103 | tilghman | 2007-11-08 00:55:19 -0400 (Thu, 08 Nov 2007) | 2 lines Typo ........ r89105 | kpfleming | 2007-11-08 01:26:47 -0400 (Thu, 08 Nov 2007) | 2 lines fix a glaring bug in the new SRV record handling that would cause incorrect weight sorting ........ r89111 | mmichelson | 2007-11-08 12:47:23 -0400 (Thu, 08 Nov 2007) | 5 lines I made this same adjustment in trunk to fix a bug, and it makes sense to do it in 1.4 as well. If an imapfolder is specified in voicemail.conf, don't ever explicitly connect to INBOX since it may not exist. ........ r89115 | qwell | 2007-11-08 14:45:15 -0400 (Thu, 08 Nov 2007) | 4 lines Avoid warnings on load when using sample configuration files. Issue 11195, patch by eliel. ........ r89119 | mmichelson | 2007-11-08 17:00:08 -0400 (Thu, 08 Nov 2007) | 7 lines Rework of the commit I made yesterday to use the already built-in ast_uri_decode function as opposed to my home-rolled one. Also added comments. Thanks to oej for pointing me in the right direction ........ r89125 | qwell | 2007-11-08 19:52:35 -0400 (Thu, 08 Nov 2007) | 4 lines Properly say the seconds here.. Issue 11203, fix described by vma. ........ ------------------------------------------------------------------------ | ||