similar to: Queues and max time in queue timeout?

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?

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

If you add "t" to you out-going trunk "Dial" lines: exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@vocal||t) exten => _NXXXXXX,2,Congestion so that you can still use park to park a call or transfer the phones, You have a problem of not being able to use "#" on external IVR systems. Is there any solution to this problem? Other then training hundreds of users not to

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

I am having trouble getting park to work with SIP, I have these config files: /etc/asterisk/parking.conf [general] parkext => 8540 parkpos => 8541-8555 context => parkedcalls parkingtime => 45 /etc/asterisk/extensions.conf include => parkedcalls include => default [default] exten => 3874,1,Dial(SIP/3874|20|tT) Do I need something else somewhere? Is anyone using park and

I've just installed asterisk as a new phone system for our office but am having difficulty with the queues. Specifically I need a way to redirect our sales queue to voicemail when no one is logged in to the queue. I see I can use the joinonempty=no setting, however this setting doesn't work if you use the agent functionality (at least not with AgentCallbackLogin). I could, of

Hi there, I have 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

When using the callback feature on agents I notice that when the queue calls one of the agents and the agent picks up the call they hear nothing until pressing the # to accept the call. Only then does my announcement play back to the agent after which the call is immediately connected. Is there a way to have the announcement played to the agent before they press # to accept. I have ackcall=yes

Hi, A potential customer would like to be able to do this: If a call comes in for an employee who is on the phone, allow the front-desk to push the caller in a queue directly to the employee. Now, this is easily done by using queues, but I am curious: What is the performance impact on a system if _every_ employee (phone) has their own queue. How scalable is that in comparison to

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

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

I would like to setup a hunt group, not a group ring, using sip phones. Anyone done this with sip devices? Comments suggestions? I have not had much luck with the outgoinglimit=1, incominglimit=1 stuff that I would need to get busy extinctions to work right, which is why I'm asking on the list.

I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen when calling from Firefly. It is also fine on both clients with 1.133 of channel.c. I am using the ALAW codec. Mark at Digium can't reproduce the problem,

Is there a way to clear the Queue stats? That is with out restarting *? I'd like to reset them daily and don't see a way to do that. Unless the only way is maybe a cron restart asterisk like every weekday @ 04:00?

Before I go off and post a feature request on the bug tracker, I want to make sure I've not misgoogled or miswikkid and not found an existing capability. What I'm looking for is the ability to determine whether or not a queue has any queue handlers (active agents), and if it does not, bypass sending the caller to the queue and pass them on to a message or IVR system. Is there a

Has anyone does this with *? Ie, ask for the caller's name and provide that to the callee before bridging? For calls to an extension, it should be doable via the dialplan. For calls to queues, some changes would be required to app_queue.c to allow an addional file to be played after the announce message. -JimC

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 18:22:34 NOTICE[426000]: chan_zap.c:4811

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

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

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 *