search for: 1.4.29

Hi All, Is it possible to use Quintum AFT800 on Asterisk 1.4.29 as Trunk for Analog (like Digium Analog Card) ? And if it's possible, could any one please give me the reference how to configure it on Asterisk 1.4.29. Thanks Regards, Zoel Hairi -------------- next part -------------- An HTML attachment was scrubbed... URL:

The Asterisk Development Team has announced the release of Asterisk 1.4.29. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.29 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix to Monitor which previously assumed the file to write to

The Asterisk Development Team has announced the release of Asterisk 1.4.29. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.29 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix to Monitor which previously assumed the file to write to

This has come up before on the list and archives but I don't seem to find a solution for this. On just a few nodes we have this situation where we see the IP disappear from the CLI iax2 show peers list but the status shows OK: 3012/3012 (Unspecified) (D) 255.255.255.255 0 OK (89 ms) How can the status be OK a few milliseconds ago and have no IP ?? The strange thing is

Hi, I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now having a problem with Originate and chan_local. I'm using the following Manager API action to originate a call: Action: originate Priority: 1 Context: trunk Callerid: 100 Channel: Local/100 at callback/n Exten: 123456789 Variable: USERFIELD=127.0.0.1|USEREXT=123456789 WaitTime: 30 This is intended to first call

Asterisk 1.4.29 BLF-SUBSCRIBE go to internal IP (ngrep output): U 2010/03/02 11:34:06.013515 212.78.xxx.xxx:2048 -> 62.134.xxx.xxx:5060 SUBSCRIBE sip:12 at 62.134.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP 212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport..From: <sip:K922002626 at 62.134.xxx.xxx>;tag=vyx8c0trgx..To: <sip:12 at 62.134.xxx.xxx>;tag=as13e7cb7c..Call-ID:

The AstLinux Team would like to announce that the 0.7.0 version of AstLinux is available for download. There have been many significant updates in this release including updating to the latest Asterisk Release (1.4.29), moving to DAHDI (2.2.0.2) along with several other system updates. For a complete list of changes, read the changelog available on the download page:

Dear all; I want to stop RTCP from Asterisk-server to phone. But I want to use RTP. I looked rtp.conf/sip.conf, but I can't know about it. Please tell me how to stop RTCP only. Because , when I access under NAT, my gateway shutdown the port as gateway received RTCP from server. I use Asterisk 1.6.2.6 or 1.4.29 . Also SIP/RTP. thx.

Hi all, I have asterisk 1.4.26 (and I tried 1.4.29) connected PRI all 23 lines and for the most part everything works. Dialing out on 23 lines to phones works fine. I have to use the Local channel to call the intercom system (from call files). If I only call 1 intercom system at a time so it uses DAHDI/1 everything seems to work as I can call all 8 intercom systems and play a message. The

Hi, I have several 1.4.29 installations with Sangoma AFT101d cards. Normally we have been collecting the raw data and then graphing channel use for these customers with: asterisk -rx 'show channels' | cut -f1 -d' ' | grep Zap | sort -u | wc -l Then I recently noticed that there were some "zombie" calls in this list that were not actually active anymore. They go

Hi! We've experienced asterisk has gone without any message, it wasn't any segfault, anything in asterisk messages log that says about shutting down. Asterisk process has just diapered. Has anybody got similar problem? Asterisk is version 1.4.29-1 from digium repository.

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can make call outside and exten 2006 to 2010 can not make call outside. heres my dial plan. ? sip.conf ? [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend context=outside secret=1234 host=dynamic [2001] type=friend context=outside secret=1234 host=dynamic [2002] type=friend context=outside

Hi, I think I got a problem with the FXListBox widget. ( ruby 1.8.4 (2005-12-24) [i386-mswin32]) ) FXListBox.new(parent, nil, 0) do |libo4| libo4.appendItem("0 very weird") libo4.appendItem("1 very weird") libo4.appendItem("2 very weird") libo4.appendItem("3 very weird") libo4.appendItem("4 very

Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? best regards Thomas

Hi all, I'm having a strange issue, wanted to see if anyone had any suggestions. Due to the recent spike in VoIP related hacking attempts I decided to tighten security by writing iptables scripts to only allow traffic to my servers which is white-listed, since then I've had an issue under certain circumstances. I have two boxes (gateway) + (end-point), both running Asterisk 1.4.29 and

Hi all, i've 2 asterisk box with dahdi (server A ver. 1.4.29 and server B ver. 1.4.26) connected with IAX channel using gsm codec. - Calling from A to B the call has no problem: ring , answer a speak without problem. - Calling from B to A : B phone always listen ring also when A phone answer. After answer A phone don't listen anything. when A phone hangup the call disconnect. full logger

Package: logcheck-database Version: 1.2.23 Followup-For: Bug #254681 Please generalize "nobody" to "[_[:alnum:]-]+", as some cron jobs su to other users: Jul 11 06:51:16 tux su[10385]: + ??? root:hinfo Jul 11 06:57:25 tux su[29801]: + ??? root:www-data Thanks. -- System Information: Debian Release: testing/unstable APT prefers unstable APT policy: (500,

Hello, klibc-1.4 and klibc-1.4.29 don't export the creat function: $ klcc -static -s -Wall rtfs.c -o rtfs rtfs.c: In function 'move_ent': rtfs.c:318: warning: implicit declaration of function 'creat' rtfs.o: In function `move_ent': rtfs.c:318: undefined reference to `creat' $ grep creat\\b /usr/lib/klibc/include/ -r /usr/lib/klibc/include/zlib.h: descriptors are

Hi, I need some help debugging a sip situation. I started to have problems with sip trunks, using more than one trunk (and sometimes using only one) the sip module seems to freeze. My local extensions lost registration and also the trunks. The only way that I can restart the sip is removing the trunks. If I make sip reload or restart asterisk the sip module takes many many time before

Hi, I've configured a new Asterisk 1.6.1 box to replace an old Bristuffed 1.2 Asterisk. Since then, it happens that forwarded calls are not presented the way they used to be. It seems that now, some endpoints are displaying the original caller id (that's what I'm trying to achive), while some are displaying the redirecting number : so if A calls B, B forwards to C depending on where