similar to: Monitor and SIP transfers (SIP REFER)

Hi there, I searched in mailinglist and in web, but no answer to my problem... Only this post with no answers: http://lists.digium.com/pipermail/asterisk-users/2004-March/038994.html I'm using CVS Asterisk (05/17/04) with chan_capi 0.3.1. (multiple controller support). In my Asterisk-box there are 2 Fritzcards (module for second card compiled with changes on sourcecode found in the web).

Anyone here running asterisk on openvz, if so what are your experiences? Right now we are trying to tune out the resources for the difference VEs, but not with a whole lot of luck. Just wondering if someone watching could shed some like on what has worked for them, and how many exts/simultaneous calls etc are happening. Thanks Miles -------------- next part -------------- An HTML attachment was

Hello list, Using Asterisk 1.2.29 I use the Monitor() application. In extensions.conf I have set MONITOR_EXEC to my script (for mixing files together and convert to mp3) and I set TOUCH_MONITOR on every new channel which has to be recorded. But sometimes I'm missing the recording files. I had a look to the Asterisk-Log and saw those lines: Feb 10 15:18:57 NOTICE[16772] res_monitor.c: monitor

Hello, I have a little problem with signalling. An E100p is connected to an Alcatel PBX, wich has an E1 to the outside. Located in Germany. zapata.conf: switchtype=euroisdn pridialplan=local prilocaldialplan=local overlapdial=yes signalling=pri_cpe .... With asterisk 1.0.2 I can call from a SIP phone to a phone connected to the Alcatel and the SIP number is correctly displayed at the caller.

Hello list, I have a problem with called ZAP channels making an attended-transfer or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk is wrong. At the moment there is a bristuffed Asterisk 1.2.18 running with bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit: [default] exten => 0123456789,1,Macro(dialpstn,${EXTEN}) [macro-dialpstn] exten =>

Hello, I have a problem with MoH at attended transfers. - Mobile A dials into Asterisk - Asterisk dials another Mobile B - Mobile B presses *1 for attended transfer and for example 20 to dial extension 20 - Asterisk sends "Remote hold" message to Mobile A, so the carrier of Mobile A starts playing it's own music-on-hold - Mobile B hang up, so Mobile A should be connected to

I have a customer that needs to record all calls coming in and out. The problem I am having is when a call comes in to the operator and it is transferred to another extension. The first mixmonitor begins recording when the operator picks up but the recording stops when the call is transferred. I need to have a single recording for the complete call no matter how many times it is transferred.

Hello, I want to enable call forwarding at my telco. In Germany you can press *21*destination# and all calls will be redirected to the destination without interaction with any equipment on my side. How to dial this with Asterisk and Zap-Channels? It can not be send as "called number", it has to be send as "keypad facility". Anyone here with some hints? The application

Hello List, i need to be able to record the call transferred from iax extension to sip extension when i call the sip extension from the IAX extension i can record the call without any issue but when i receive a call from customer in IAX and i transfer this call to SIP client the conversation between customer and IAX client is recorded but the conversation between customer and sip extension is

Hi, Just wanted to let you know that we have just made our GPL toolkit "Gemeinschaft" available to the public. (Finally.) Mostly German for now - about half of the strings in the language strings file have been translated to English. I'm a software developer, not a marketing guy, so ... svn co https://svn.amooma.de/gemeinschaft/trunk gemeinschaft-trunk German readers: see

Hello, I thought to post this here before my manager starts his own coding project to give us a workaround. My situation I'm running into is as follows: 1. A call comes into our Asterisk system, it's trunked from one office to another via IAX. 2. Call enters a queue and is picked up by one of the agents. 3. That agent has to transfer the call, could be for a number of reasons the client

Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

cdr_mysql doesn't set the userfield when it's set inside a macro called from a feature (1.4.25, addons 1.4.8). I have a feature code: autorecord => *1,self,Macro,apprecord The apprecord macro looks like: [macro-apprecord] exten => s,1,Playback(beep) exten =>

Hello Folks; I appreciate all of the help so far - thanks. Another question: I am using MixMonitor() to record calls and I would like to include the called number/extension in the filename: In my dialplan, I am able to save the file with the caller id in the filename. However, what I am a little unsure about is the incoming number/called number/extension - passing that information on to part

Greetings, I would like to display the R graphics window on one monitor of my computer while simultaneously working with the R console and multiple script windows in another monitor. Is this possible? I know I can use Tinn R or something similar but I would prefer not to use these secondary programs if possible. Any advice would be appreciated. Thanks, -Gunnar Carnwath -- View this message

Hi Folks, i implemented the required callback funktions to open an ogg file from memory. Now ov_pcm_total() returns always 0, strange thing about is that ov_pcm_total does not call any of the callback funktions. I also like to mention that ov_info (which uses the read and seek callback funktions) delivers correct values. Thats why i doubt that there's an error in (the rather simple) callback

Hello, I cannot get JGR installed. Here is what I have tried so far http://wiki.ubuntuusers.de/R 1. sudo -s # root werden R CMD javareconf root at gunnar-laptop:~# R CMD javareconf Java interpreter : /usr/bin/java Java version : 1.5.0_18 Java home path : /usr/lib/jvm/java-1.5.0-sun-1.5.0.18/jre Java compiler : /usr/bin/javac Java headers gen.: /usr/bin/javah Java archive

Hallo Zusammen, hatte gestern die Schnauze voll von Vista und habe mir daraufhin Ubuntu installiert. Bin also mit Ubuntu ?berhaupt nicht vertraut. Nun habe ich ohne Erfolg versucht, R zu installieren. Soweit ich der Anleitung folgen konnte habe ich es so wie auf http://wiki.ubuntuusers.de/R gemacht. Im Terminal kann ich R problemlos starten. Allerdings bei der Installation von

If you execute cat /dev/urandom at the xm console of a guest domain, it will spew garbage forever. Attempts to run xm destroy on it simply hang. _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users