similar to: LDAP authentication in Asterisk

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, 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.

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 =>

I am trying to register my Asterisk server to a SIP server which doesn't accept an Expires: field smaller than 1800 seconds and indicates it correctly with a Min-Expires: in an error response when Asterisk tries to use its default of 120 seconds. Is Asterisk supposed to honor this field and retry with the proposed minimum Expires: field? It looks like it doesn't, and I had to change the

Hi Guys, I started getting this error only from one of our ITSP's once we upgraded from 1.2.16 to 1.2.17. Can anyone shed light ? --- (12 headers 0 lines) --- Transmitting (NAT) to 209.212.93.53:5060: SIP/2.0 603 Declined (no dialog) Via: SIP/2.0/UDP XXX.XXX.XX.XX;branch=z9hG4bKf928.2b3b0de5.0;received=XXX.XXX.XX.XXX Via: SIP/2.0/UDP

Hello, In a couple of Asterisks, after type "sip show channels" we have a lot of these: IP_PEER dst_number something 00102/00103 unkn No (d) Rx: BYE IP_PEER dst_number2 something2 00102/00103 unkn No (d) Rx: BYE We are using ASterisk 1.2.x When I say "a lot" I mean more than 180, more than 230, etc. Is it normal? How we can remove it? Thank you very much, --

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

Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is

* NEW CHANNEL DRIVER FOR ASTERISK 1.6 AND VOXSWITCH 3 ADDS AUDIO AND VIDEO TO MICROBLOGGING! In a surprising move, Digium in partnership with Edvina today released a new channel driver for Asterisk, chan_tweet. The driver connects seamlessly to several microblogging platforms, including Twitter, Facebook, Laconi.ca/Identi.ca and GSM text/SMS. The main feature of this new module is to

In Sweden, where I live, it's snowing like crazy. The Stockholm area is covered in white stuff and there's really no reason to leave the computer and get out anywhere. More white stuff is coming down all the time. Boring. I am sure your weekend is no better - rain, snow or just another boring sunny day. Let's find something cool to do during this weekend! Join the cool crowd

Hello List I am very interested in developing a research project on security protocol for VoIP, under the GPL. For some time I have been reviewing ZRTP, I would like to know the opinion having regard to whether and under asterisk, but I see that this closed implementations according am Http://bugs.digium.com/view.php?id=10024 Are Zphone and ZRTP the future for the Voip Security? Opinions?

Does any one know if there's an mechanism (internal to asterisk or otherwise) to replicate dynamic SIP device registrations across a pool of asterisk servers? I'm in the process of creating a asterisk cluster using a SIP hardware load balancer and so far this is one of the challenges I'm facing. One thought I'm currently investigating is to use openSER to intercept and

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))

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

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

I made a similar question in a previous thread, but there was no answer, so I think I was not very clear making the question. What I need is some configuration that works like "promiscredir=yes" in sip.conf that enables me to do the same thing with transfer (REFER), letting me transfer a sip call to a non local sip address. Thanks in advance, Thiago Abra sua conta no Yahoo!

Hello Asterisk users around the world! Recently, I have been working with pretty large Asterisk installations. 300 servers running Asterisk and Kamailio (OpenSER). Replacing large Nortel systems with just a few tiny boxes and other interesting solutions. Testing has been a large part of these projects. How much can we put into one Asterisk box? Calls per euro invested matters. So far,

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

What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. Yuan Liu

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