similar to: [patch] Binding rtp to specific interface

http://bugs.digium.com/bug_view_page.php?bug_id=0001429 * Support for other language syntaxes in saynumber Accidentally I opened this can of worms to see if we can add support for other language syntaxes for saying numbers. Seems like Swedish, english and norwegian follow the same syntax. I've integrated existing patches for french, danish and soon portuguese syntax. The steps we're

The recent addition of recursive mutexes to Asterisk is causing a lot of problems on FreeBSD servers. I need help from someone that knows mutexes on FreeBSD to make it work, otherwise the FreeBSD port of 1.0 will be useless. See bug report http://bugs.digium.com/bug_view_page.php?bug_id=0001411 for more details. Thank you for your help! /Olle

http://bugs.digium.com/bug_view_page.php?bug_id=0001485 After spending a lot of time saying numbers and dates, the Asterisk I18N project now targets voicemail. The voicemail prompts are very much based on english language syntax, which works for some languages and doesn't work for a lot of languages. Fran Boon, aka Flavour, have done an excellent job in merging patches and building a

Please forgive me if this is a silly question. I've been following this thread in the hope that I could put my * server and snom 200 into full-time service very soon. I need to find out how to have the lines configured so that it does not return a busy reply when only one call instances is engaged. Am I supposed to create multiple extensions on my asterisk dialplan to reflect the 5 call

Your log file almost looks like a bug in Asterisk doesn't it? Why call poll() with a zero timeout while passing only one FD? and then why do the read when there is no data? Read the man pages for all the system calls Take a look at the source chan_sip.c /* Wait for sched or io */ res = ast_sched_wait(sched); if ((res < 0) || (res > 1000))

Actually, I have tested this here with an Aastra 9133i and an Asterisk@Home server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own. > Thanks Olle, > > So am I to understand that you

Trouble getting chan_sip2 to compile below is what I have done -download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software cp /root/software/chan_sip2s.c /usr/src/asterisk/channels cd /usr/src/asterisk/ patch -p0 acl.c /root/software/acl.c.patch cd /usr/src/asterisk/include/asterisk patch -p0 acl.h /root/software/acl.h.patch - added the follow to /usr/src/asterisk/channels/Makefile

** Asterisk Bootcamp in Stockholm, Sweden The next Asterisk Training is the Edvina.net Asterisk Bootcamp - the class we have been giving for over a year under the brand name "Astricon Training". The same teacher, the same material and a new name. All students have a PC and will install a fully working Asterisk PBX. During the week, we will build a business PBX configuration as

Today is Swedens national day - since a few years a holiday too. We don't have a tradition on how to celebrate. Sweden has not been to war for a very long time, so there's no real spirit for the country here - it's been aroundfor such a long time, so what? :-) Guess we have to learn from abroad, to get a celebration feeling like July 4th in the US or May 17th in Norway (from

Friends in the Asterisk community, I've been talking for years about the new version of the SIP channel. I've been trying to get funding and get going. Well, the funding part remains to be handled, but I have other news - if you kan keep it to yourself. ...I've began coding. Finally. With a happy smile on my face I removed "pedantic=yes" the other day. After years of

Friends in the Asterisk.org community, There is a lot of cool stuff going on in Asterisk development, things that will change Asterisk and make it work better in your organisation, make it easier to sell in your area or give you more consulting oppurtunities - in short, functionality that will make a lot of sense for you users. However, developers can't really get anywhere without a

*lol* I think he was referring to a ASTERISK reboot, not a phone reboot. > -----Original Message----- > From: Olle E Johansson [mailto:oej@edvina.net] > Sent: Thursday, March 23, 2006 1:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Re: Subscription state after reload (New > subject) > > > > 23 mar 2006 kl.

How can a reload clear registrations? If I 'reload' without using realtime, I keep my sip peers (as well as astdb). I can still contact other phones... registration info is still there. If I `reload` with realtime, I lose my sip peers (but astdb remains). I can _STILL_ contact other phones.... registration info is still there and Asterisk must be referring to astdb to find the IP

When creating users in the sip.conf file, they do not appear when running the "sip show users" command from the CLI until i restart. A reload doesnt make them appear. As i said, I am new to the whole Asterisk thing, however have worked with IP/SIP PBX's for a few years - its most likely a user problem though! Check it out and let me know what you get. Cheers Chris PS - I would try

Time to reboot and re-start Asterisk, well, hrrm, monthly, news. It's been a hectic fall with a lot to do, both before and after Astricon. At this time, we're preparing for two Astricon shows in 2005. And no, we haven't made a decision on where to run the European Astricon, not yet. I am preparing to travel to the USA again this coming week. Today, I'm spending my time finding

--- "Olle E. Johansson" <oej@edvina.net> wrote: > From: "Olle E. Johansson" <oej@edvina.net> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Asterisk on FreeBSD > Date: Mon, 27 Oct 2003 08:24:22 +0100 > > Rich Adamson wrote: > > >>My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD > server.

Friends, To update you on recent changes in svn trunk, I can inform you that we now have ever smarter ways to handle media streams in Asterisk than we do in 1.2 for the IAX2 and SIP protocols. * IAX2: Splitting signalling and media apart Starting with the IAX2 protocol, we now have the ability to transfer media streams to go directly between IAX2 servers and keep the signalling path.

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files

http://bugs.digium.com/bug_view_page.php?bug_id=0001589 Has anyone else heard an audible blip, break or garble between answer and the native bridge attempt using sip? If I change the usleep(500000); to usleep(5000); in rtp.c the proble totally goes away... even the note above it says it needs to be fixed. bkw -------------- next part -------------- An HTML attachment was scrubbed... URL:

If you're using MGCP, we need your help. There's a patch in bugs.digium.com that needs testing by the community. Please spend some time testing and adding your comments to the bug tracker. The author writes: ------------------ I'm trying to make work Asterisk against a Cisco IAD2431 with chan_mgcp. Since chan_mgcp assumes the Line package is the default which is not the usual with