similar to: Asterisk 11.3.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 11.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 11.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Hello all, I have an Asterisk PBX with the "Queue Log Analyzer" installed [http://www.micpc.com/qloganalyzer]. On the main menu, there's an option of "CALLS COMPLETED [ALL]" where I can see the completed calls that entered any of the queues and my question is: There's a column that states either "COMPLETECALLER" or "COMPLETEAGENT" and I want

I went to run my queue_log parser so that I could send out a monthly report to one of my customers, and I noticed that every valid call complete action (COMPLETEAGENT, COMPLETECALLER) is followed by an ABANDON: Here is a complete-caller: 1123325015|1123325011.2|mainq|NONE|ENTERQUEUE||00110102102 1123325020|1123325011.2|mainq|Agent/21|CONNECT|5

Hei! Here's my problem. I have an Asterisk with SS7 and SIP trunks. Asterisk version is 1.6. I'm setting up a custom CDR fields and I was wondering is there a way to know who initiated a hangup? Asterisk must be aware of that info somehow, cause in queue_log, that info is present (completecaller, completeagent) Is there a way to get that info on the regular SS7 to SIP (and vica versa)

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see

Howdy! I'm still trying to make it onto the 1.4 releases. Almost ready to make the switch, but here's one last thing that doesn't seem to work: Server A calls Server B over IAX, i.e "Dial(IAX2/SrvB/${EXTEN},60)". Both machines are set up in iax.conf to use RSA certificate-based authentication. The public keys have been exchanged. This setup works just fine as long as Server

Does anyone know why i would get this RINGNOANSWER events in queue_log when clearly the agent is busy and call-waiting is disabled. 1309550595|1309550570.399965|2253|Local/05 at from-internal/n|CONNECT|2|1309550593.399966|0 1309550632|1309550533.399961|2253|Local/11 at from-internal/n|COMPLETECALLER|1|74|1 1309550663|1309550640.399969|2253|NONE|ENTERQUEUE||zzzzzzzzzz

I am experiencing several entries in the queue_log with a duration of NULL at the COMPLETEAGENT or COMPLETECALLER event. Any idea how this can happen? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan

Hello, I'm facing a rights issue on with Asterisk 11.3.0 running on CentOS release 5.8. Asterisk process is running with asterisk since it is define in asterisk.conf as following: runuser = asterisk rungroup = asterisk You can see asterisk proccess here: ps aux |egrep 'python|asterisk' root 11581 0.0 0.1 65940 600 ? S Apr17 0:00 /bin/sh /usr/sbin/safe_asterisk

Hi, Before trying to script res-memcached installation (see res_memcached<https://github.com/drivefast/asterisk-res_memcached>), I banged into this on a fresh 11.3.0 setup: 1. When run for the first time bootstrap.sh displays a non-blocking error. # sh -x bootstrap.sh + uname -sr + MY_AC_VER= + MY_AM_VER= + AUTOCONF_VERSION=2.60 + AUTOMAKE_VERSION=1.9 + export AUTOCONF_VERSION + export

Has anyone observed a problem where using Local channels with AddQueueMember results in missing TRANSFER events? Right now I'm using straight SIP channels when I call AddQueueMember(). I'm contemplating moving to Local channels because the non-state-based wrapuptime blows when you have a channel in multiple queues (they can hang up and get a call immediately so long as it's from a

Under 1.4.5 and 1.4.6, I've seen a few instances where the channel name in queue log entries is replaced by a snippet of a manager event: --START-- 1183582823|1183582823.104763|queuename|SIP/XXXX|REMOVEMEMBER| 1183582828|1183582793.104744|queuename| Context: macro-dialout Extension: s Priority: 3 Application: GotoIf AppData: 0?blockclid Uniqueid: 1183582822.104759

Hi, For some reason (outbound call tracking) I've got a few different outbound call process (using a macro for queuemetrics logging, or direct call) i wanted to factorise the routing process so i came up with something like the following. All in one it's working like expected, however every "ael reload" command trigger a lot of warning like that "application call

Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext => 100 parkpos => 1-5 context => parkedcalls parkingtime => 100 transferdigittimeout => 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer => *2 blindxfer => # disconnect

Hi. I have a RHEL server that has some errors in dmesg , what do they mean, how do I fix them ? mtrr: type mismatch for f9000000,800000 old: write-back new: write-combining mtrr: type mismatch for f9000000,1000000 old: write-back new: write-combining mtrr: type mismatch for f9fe0000,10000 old: write-back new: write-combining mtrr: type mismatch for f9fc0000,20000 old: write-back new:

I'm trying to compile chan_mobile for asterisk 1.4 I've installed 1.4 from SVN and downloaded addons from SVN also. I make ./configure, make menuconfig, select only chan_mobile, and make. Then I obtain the following errors. (I've also tryed applying the patches I found at http://www.chan-mobile.org/?page_id=5 but with no better results. make[1]: Entering directory

Hi Folks, I'm trying to use my mobile as a trunk via bluetooth - calls done in a softphone go thru GSM network and calls destinated to my mobile are answered at the softphone. I have asterisk configured to do so but I'm facing an issue - Audio is audible but it?s not intelligible. I feel like the audio is breaking. Below is the asterisk log. I also get lots of ?hci_scodata_packet: hci0

Hi all, I receive this error while compiling chan_mobile: gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c chan_mobile.c: In function `mbl_load_config': chan_mobile.c:1745: erreur: trop d'arguments pour la fonction ? ast_config_load ? make[1]: *** [chan_mobile.o] Erreur 1 make[1]: Leaving directory `/usr/src/asterisk-addons' Does anyone know what's the problem? --