similar to: sip via tcp

Thank you very much for all feedback on Asterisk Sunday News! This is the last issue for June. This week I'll go on holiday and will be back with more news in early July. My kids are getting summer leave this week and we'll be visiting the south of England for a while. Another part of Europe that still use their own currency. If you think there's an European standard, you're

asterisk-users-bounces@lists.digium.com wrote: > Is there anyone else with the same problem? Yes, we've seen the same problem. We have found a work around, but I'm unable to to look into it today. -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

It looks like chan_sip2 may solve my problem with outboundproxy support. However, I am having problems getting the solution installed. From what I understand these are the tasks... Add chan_sip2 to the channels/Makefile * Rename the file downloaded to chan_sip2.c * make / make install * Change your modules.conf Add "noload=chan_sip.so" if you want to run chan_sip2 * Restart

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

I am going to change my Digest realm to match my DNS SVR record. I dug through the code in chan_sip.c and on line 2748 I found it hard coded <frown> : snprintf(tmp, sizeof(tmp), "Digest realm=\"asterisk\", nonce=\"%s\"", r\anddata); I'm going to change this to : snprintf(tmp, sizeof(tmp), "Digest realm=\"isdn.net\",

Hi, Does anyone have any detailed install instructions for setting up chan_sip2.. I patched acl.c but could not see an acl.h file to apply the patch.. I copied the chan_sip2.c file into the channels directory.. I am not sure what I need to do exaclty in the Makefile to get chan_sip2 to build.. Any help and anything to be careful of in chan_sip2 would be usefull.. Thanks, Later..

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

sorry if this has been asked before. is it possible to configure asterisk to use an outbound sip proxy? thanks __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

Can a call be kicked out of a queue if it reaches a specific timeout? I don't see an obvious way to do this in either queues.conf or extensions.conf any pointers or patches to do this? <smile>

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

Is it possible to configure * so that if a caller reaches voicemail for someone in Engineering, but doesn't want to leave a message they can press zero (0) and reach the Engineering Secretary or if they are calling someone in Accounting and reach voicemail, pressing '0' would reach the Accounting secretary, not the Engineering secretary? Don Pobanz

I have tried for over a month off and on to get iaxtel for inbound to work... and tonight after alot of troubleshooting we noticed this: iaxtel inbound will use the last entry in your iax.conf to auth against. So if [iaxtel] is at the top and say [voicepulse] at the bottom. An inbound call will try to auth against that [voicepulse] entry even with the [iaxtel] entry at the top of the file. Has

This is a followup to the posting earlier about Hunt Groups with Call Screening. I have implemented the following macro and for some reason the Macro does not exit and continue the context it was called from when the called party rejects the call - It always drops through to the NoOp at the end and connects the call. Below are two examples of the dial commands I am using to call the macro.

Hello, Right now I have app queue working with incominglimit=1, there is no call waiting signal, but after a while( like couple of hours) some phones randomly get stuck. The * thinks that they are in use and doesnt ring them, when they are infact not in use. sip show inuse, shows that they are inuse. typing reload on the console resets this and they are again available for working. anybody

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

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

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

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

A friend and I have recently added the ability to announce the callers position in the call queue every x seconds.. or even just inject an anouncement every x seconds. All setup in queues.conf and can be setup per queue. My next project is to add the ability to announce the callers estimated wait time. I want some feedback to see whats the best method to calculate that? What do you want just

I logged a bug I wanted to see if anyone else is having this problem or if it's just me. http://bugs.digium.com./bug_view_page.php?bug_id=0000486 I just downloaded the newest version from CVS(Tuesday@~7pm) and I am getting an error whenever I call the asterisk box. I cannot here any audio on the budgtone. This works fine with my pingtel phone and my sip 7960. Also if I call my Skinny 7960 it