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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 being used...

...ve serious problems with MGCP and Swissvoice ip10s, and it appears that recent CVS also introduced trouble for other MGCP users. Please check and add comments in the bugtracker so that we can get a clearer picture - thanks! Also comment if things are working fine for you. http://bugs.digium.com/bug_view_page.php?bug_id=0001542 http://bugs.digium.com/bug_view_page.php?bug_id=0000881 and other MGCP related bugs/fixed. Cheers, Philipp

I just ordered the Asterisk Developers Lite kit. My environment will be the RH9 Linux server and a Windows workstation with Samba. I also of course have analog lines and DSL. I am interested in SIP development. I already downloaded the Asterisk software. What else should I download. Is there a doc that basically tells you the steps to install Asterisk and get it up and running? I would like a

Can someone take a look, tell me if this is a bug, a possible resources issue, or my own damn fault? http://bugs.digium.com/bug_view_page.php?bug_id=0001381 Thanks, Derek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040407/f8f4d79b/attachment.htm

...ec. Mark at Digium can't reproduce the problem, although I don't know a) whether it is specific to GS phones, and b) whether Mark has a GS phone to try. Please could others with a similar setup try it? The app command is MeetMe with the M flag. The bug report is at http://bugs.digium.com/bug_view_page.php?bug_id=0002312 Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org

Any body used patch, http://bugs.digium.com/bug_view_page.php?bug_id=0001719 to get the callerid for BT Line. I applied the patch successfully but could not get it to work. Any help. Here are the logs: -- Starting simple switch on 'Zap/1-1' Jun 17 18:22:31 NOTICE[426000]: chan_zap.c:4811 ss_thread: Got event 2 (Ring/Answered)... Jun 17 18:22...

Someone submitted a bug about wav49 voicemail problems with the Windows Media Player here http://bugs.digium.com/bug_view_page.php?bug_id=0000254 bkw918 changed the status of the bug to resolved because he could not reproduce the error with his version of Windows Media Player. I am having the same problem as the original bug poster. I am using WMP 9.00.00.3075 running on Windows XP and using Asterisk CVS-01/13/04-00:08...

I am trying to find the app ChanSpy for Asterisk v1.0.7. I have tried looking on VOIP-info.org's ChanSpy page (http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ChanSpy)and also referred to the link regarding bug 3836 (http://bugs.digium.com/bug_view_page.php?bug_id=0003836). I downloaded the attachments and tried to use the patch and compile the source. However, it seems that these files are for a different version of Asterisk. Searching Google provides no relevant material. If anyone has any information as to where I can find ChanSpy for Ast...

Hello everyone. I am having problems getting speex support. It seems * is not loading speex. When i did a make in the codecs sub dir, the following error pops up when making speex: codec_speex.c:34:19: speex.h: No such file or directory is this file missing in the cvs as i just removed the whole * dir and did a new checkout and still seem to get this error, or do i need to get/install

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 fr...

...ls will receive all "Events" that happen on the Asterisk box - Not all Asterisk commands will be accepted with the "Action: Command" action - Disconnecting the connection between a remote connected terminal and the Asterisk box will often cause a deadlock (http://bugs.digium.com/bug_view_page.php?bug_id=0000861) - Any form of freeze on the connected terminal will cause a buffer overflow and will also deadlock Asterisk - "Status" Actions can yield upto a hundred lines of output depending on how busy your Asterisk machine is. - "Ping" and "Show uptime" may n...

a call on which caller ID is unavailable, and a call that's supposed to be private? As a side note, I have a phone on which I have caller ID blocked, but the Asterisk server still ends up getting caller ID from that line anyway. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET *

...atch being offered, we await William Waites' patch and disclaimer which will at least be more sufficient than the current externip= method. Those with an interest in the discussion of how Asterisk should handle being put behind a NAT should direct their attention to: http://bugs.digium.com/bug_view_page.php?bug_id=0000104 JT

So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=0000889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think thi...

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

After browsing through bugs.digium.com I saw no mention of any work to get chan_sip or chan_sip2 to listen on tcp, as well as udp. Just curious is any-e-one working on such a patch at the moment?

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 -------------- nex...

Hi Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the best/only way to get callerid working in the UK with a tdm400p? I thought I'd seen a patch that'd gone into cvs, but maybe I was just imagining things ;) Should this patch work against current cvs? Of the 3 files 2 are .patch and one is .diff - what's the...

...es added in the following link for call transfer? The authors seem to feel they are finished but it doesn't appear to have been integrated into what everyone can download. It is referred to as a patch but I don't understand how it could be applied. Here is the link: http://bugs.digium.com/bug_view_page.php?bug_id=0002010 I guess I just don't understand how to apply patches.... Thanks in advance, Roger Easlick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040906/3b03d233/attachment.htm

http://bugs.digium.com/bug_view_page.php?bug_id=0002945 Test it.. I couldn't sleep tonight... thought I would see if I could find and fix it... Also did this gem too for ya... http://bugs.digium.com/bug_view_page.php?bug_id=0002948 bkw