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...ts, since for both friends and peers, we now have *one* object in memory that handles the limit for both incoming and outgoing calls. During the week, I've also added a few other patches by other contributors. Read the README.test-this-branch here: http://svn.digium.com/view/asterisk/team/oej/test-this-branch/ README.test-this-branch?view=markup ** PLEASE help the community, please test this branch. Check it out like this svn checkout http://svn.digium.com/svn/asterisk/team/oej/test-this- branch test-trunk Then cd into test-trunk and run "make" then "make install&q...

...< 0) ms = 0; return ms; gettimeofday might be failing!! Why? I'd add a call to perror() after "his should never happen" I'll bet it has to do with time zones But then maybe ms is beingset to zero... Good luck --- "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 21:25:04 +0100 > > Perry E. Metzger wrote: > > > "Olle E. Johansson...

...ngeLog for more information about these changes. * Resolve deadlocks related to device states in chan_sip (Closes issue #18310. Reported, patched by one47. Patched by jpeeler) * Fix channel redirect out of MeetMe() and other issues with channel softhangup (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. Patched by russellb) * Fix voicemail sequencing for file based storage. (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by jpeeler) * Guard against retransmitting BYEs indefinitely during attended transfers with chan_sip. (Re...

...ngeLog for more information about these changes. * Resolve deadlocks related to device states in chan_sip (Closes issue #18310. Reported, patched by one47. Patched by jpeeler) * Fix channel redirect out of MeetMe() and other issues with channel softhangup (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. Patched by russellb) * Fix voicemail sequencing for file based storage. (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by jpeeler) * Guard against retransmitting BYEs indefinitely during attended transfers with chan_sip. (Re...

...(Closes issue #18497. Reported by vois. Tested by vois, rossbeer, kowalma, Freddi_Fonet. Patched by dvossel) * Don't delay DTMF in core bridge while listening for DTMF features. (Closes issue #15642, #16625. Reported by jasonshugart, sharvanek. Tested by globalnetinc, jde. Patched by oej, twilson) * Fix chan_local crashs in local_fixup() Thanks OEJ for tracking down the issue and submitting the patch. (Closes issue #19053. Reported, patched by oej) * Don't offer video to directmedia callee unless caller offered it as well (Closes issue #19195. Reported, patched by one4...

...(Closes issue #18497. Reported by vois. Tested by vois, rossbeer, kowalma, Freddi_Fonet. Patched by dvossel) * Don't delay DTMF in core bridge while listening for DTMF features. (Closes issue #15642, #16625. Reported by jasonshugart, sharvanek. Tested by globalnetinc, jde. Patched by oej, twilson) * Fix chan_local crashs in local_fixup() Thanks OEJ for tracking down the issue and submitting the patch. (Closes issue #19053. Reported, patched by oej) * Don't offer video to directmedia callee unless caller offered it as well (Closes issue #19195. Reported, patched by one4...

...g posted on this from someone using Polycom phones so it is > happening on at least 3 totally different phones. > <http://bugs.digium.com/view.php?id=6047> > > Sorry if I am hijacking this thread. > > > -----Original Message----- > > From: Olle E Johansson [mailto:oej at edvina.net <http://lists.digium.com/mailman/listinfo/asterisk-users>] > > Sent: Thursday, March 23, 2006 12:15 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Re: Subscription state after > > reload (New subject...

---------------------------------------------------------------------- Message: 1 Date: Tue, 6 Mar 2007 20:02:07 +0100 From: Olle E Johansson <oej@edvina.net> Subject: [asterisk-users] Building a new voicemail system... Testers needed! To: Asterisk Non-Commercial Discussion Users Mailing List - <asterisk-users@lists.digium.com> Message-ID: <A8C949D0-6208-41FF-85BD-E8BDDA6BFCF5@edvina.net> Content-Type: text/plain; chars...

.../acl.h.patch - added the follow to /usr/src/asterisk/channels/Makefile chan_sip2.so: chan_sip2.o cd /usr/src/asterisk make make install I assume that problem is with what did or didn't add to the Makefile Thank for any help ----- Original Message ----- From: "Olle E. Johansson" <oej@edvina.net> To: "Glenn Dalgliesh" <asterisk@techhat.com> Sent: Tuesday, April 20, 2004 1:29 PM Subject: SIP re-invite > Could you please test this with my chan_sip2. I have a hunch it will work with > that channel. > > /Olle >

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>

Hello Previous context :- After Looking up sip and IAX2 that require configuration at router level which may cause some problems like connection break etc... so i left them ......... and start wondering if there is some thing that dont require configuration at router layer. The task to accomplish to make and recieve calls from outside local network using any protocol whose soft phones are

...nteresting challenges. So take one of these standard rack servers from HP and run a telco for a small city on one box. While you're at it, buy a spare one, hardware can fail ( ;-) ). But don't say that Asterisk does not scale well. Those times are gone. /Olle --- * Olle E Johansson - oej at edvina.net * Open Unified Communication - SIP & XMPP projects

This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider

...7999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchuan, grecco. Patched by rmudgett) NOTE: Be sure to read the ChangeLog for more information about these changes. * Fix channel redirect out of MeetMe() and other issues with channel softhangup (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. Patched by russellb) * Fix voicemail sequencing for file based storage. (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by jpeeler) * Guard against retransmitting BYEs indefinitely during attended transfers with chan_sip. (Re...

--- "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)...

Apologies if this is an old question; I've searched the list and the wiki but have not been able to find a definitive answer. I have an Aastra 480i phone registered with * 1.2.4; I want to generate UK ringback tones when the handset dials another internal extension. On my Zap channels, I have this in place by editing /etc/zaptel.conf; however I've had no luck with the Sip handset (I have

This might be a better question for the dev list, but I don't think they want to be bothered by my silly questions. Does anyone know when we can expect to see a jitter buffer for SIP channels? I know they've been working on a generic jitter buffer since around last summer, just wondering if there's been any progress.

*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. 20.09 skrev mustardman29: > > > Sorry f...

...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 address the other phone is at. Doug > -----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. 20.09 skrev mustardman29: > > > Sorry f...

...els - Totally rewritten SIP transfer code In testing right now is the new t38 passthrough code, which will be integrated after some more reviews and tests. Please help us test that code too. Thank you all for working hard on this until I return online! Regards, /Olle --- * Olle E. Johansson - oej@edvina.net * Asterisk Training http://edvina.net/training/ * Next training and dCAP in Stockholm, Sweden, June 2006! --- * Olle E Johansson - oej@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden