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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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Asterisk SIP Messaging

Asterisk SIP Messaging


For correct messaging support, there are some changes required to core. Anyways, if you require this functionality within it's limits, there's a way to do it (at least for SIP). First, you'll need Asterisk presence set-up and working from CVS HEAD. Messages are delivered according to presence hints (and not according to dialplan). If you want, you can enable queuing, which requires sqlite3 database.


After applying presence patch and configuring presence, visit this page: http://juraj.bednar.sk/work/software/asterisk/messaging/. sip_message_support.patch is a patch for chan_sip.c. On the top, choose if you want queuing support (queuing messages for delivery to devices when they become online): if you don't want queuing support, comment out the include, compile and that's all.

Otherwise, edit channels/Makefile and add -lsqlite3 to chan_sip.so CC line. Grab the chan_sip_queue* files, edit chan_sip_queue.h and set the path to the queue database (you will probably want to change that). Run chan_sip_queue_init.sh (from the channels directory, it needs chan_sip_queue.h in current directory to determine the path), which will initialize the queue (create a database and a table which will hold the queue). Compile, run and there you should go.


Tested with Xten's eyeBeam. There should probably be better delivery checking, cleaner installation, etc. Patches should be mailed to author and asterisk (at) bednar (dOt) sk.


There's also an interview http://www.sineapps.com/news.php?rssid=933 with Joshua Colp on Asterisk News page, where he mentions he is working on better messaging implementation for Google's Summer of Code. His work is availaible for download http://quark.file-radio.com/asterisk/asterisk-messaging-1.tgz, although I have no idea how it works besides what's mentioned in the interview. I hope he'll finish soon and post 1.2 asterisk will include comprehensive messaging support.


Here is a version of the patch for asterisk 1.4.0 as 1.4.0 does not support SIP/SIMPLE messaging without the patch. It has been moderately tested so use with care. http://www.sipalive.com/dev/asterisk/


Comment by oej:

Although this is a great SIP patch, it will not be included in Asterisk due to it's one-channel perspective. As Juraj says, this is a hack and proof-of-concept. There are work going on to create a multi-protocol IM and presense solution for Asterisk.

See also


Created by juraj, Last modification by Lukas Oberhuber on Wed 03 of Jan, 2007 [01:01 UTC]

Comments

Comments Filter

not worrking

by cingkele on Monday 18 of September, 2006 [04:04:10 UTC]
I try to patch but not work for me.

Sep 18 12:02:27 WARNING2722: chan_sip.c:7545 receive_message: Received message to "222"<sip:222@voip.ung.ac.id> from "Iwan"<sip:555@voip.ung.ac.id>;tag=91435f6e, dropped it...
 Content-Type:text/html
 Message: <FONT face=Arial size=2>test</FONT>

can you help me please

Jurai patch updated.

by Francesco P. Sileno on Tuesday 14 of March, 2006 [20:19:38 UTC]
While waiting for SIP messaging integration in Asterisk, I've updated Jurai's patch to get it work on Asterisk 1.2.5. I've tested it only in non-queueing mode.

I've notified Jurai, in the meanwhile you can get it here: http://www.rlyeh.it/tmp/sip_message_support_1.2.5.patch.gz (this is only the diff file, you must download the other files from Jurai's site).

If I've missed some other working alternatives or upgrades to Asterisk, please le me know! :)

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