similar to: PauseMonitor() Hanging Up Call

Hi I'm using asterisk 1.8 on CentOS 5 I'm initiating call recordings with MixMonitor and trying to pause them with the features.conf. Whenever I try to pause the recording the call dies. Is PauseMonitor incompatible with MixMonitor? Here are some key log excerpts features reload == Parsing '/etc/asterisk/features.conf': == Found == Registered Feature

I have got call recording working on our 1.4.30 asterisk box together with a recording pause ability and being able to play different audio to each party at the start and end of the pause. This all works perfectly but one wish is to have the audio files have a beep or something in them so when you listen later you can tell where the audio was paused. So I changed things around so that instead

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon =>

After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault. [root at localhost asterisk-11.1.2]# asterisk -vvvvvvc Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components

I just installed tapi and some app called identapop pro. I havent tested incoming calls yet but so far, I cant get calls out using outlooks. I configured TAPI for asterisk inside outlooks and I set TAPI to these configs: TAPI connects using the manager to asterisk without problems. As channels I configure this: User channel: SIP/myphone and the phone actually rings when I tell outlook to dial

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, we've just released Asterisk-Java 0.3-m2 at http://asterisk-java.org. The Asterisk-Java package consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server. Asterisk-Java supports both interfaces that Asterisk provides for this scenario: The FastAGI protocol and the Manager

Hi all, I have to questions about queues. Member is a phone like SIP/myphone and only one member in the queue. At first, DIALSTATUS doesn't return any status. How to now if a call in queue has been answered or if caller just hangup? Second, how to deal with timeout, I have strange behaviors. If I put timeout=60 in queue.conf and I call the queue passing also 60 as timeout value,

Hello When a call comes in, I'd like to fork a Python script that broadcasts a message so that users see the CID name + number pop up on their computer screen, and simultaneously ring their phones. The following script doesn't work as planned: It waits until the script ends before moving on to the next step, which is Dial(): =========== exten =>

The call quality is really pretty good. I think better than Vonage over an FXO bridge. If you are looking for a home provider with direct SIP support and local phone numbers this is a good choice. If anyone has questions or comments about my configuration please pass them along. I have noticed that if you don't put fromuser=phone# then the extension caller id passes through. Also the

Alright I know how to do basic IVR in *. But what I'm working on trying to do now is a survey. I've found very little things out there on google or the archives for how to do surveys with the * ivr. Here is more or less what I'm trying to accomplish 1. Call comes in Plays Greeting 2. Starts Survey 3. Ask Q1, Record the answer (voice responses) repeat this

I am new to Asterisk and currently work mainly with Cisco Callmanager. With Callmanager I can setup partitions and call search spaces to determine where a given phone can and can't dial. Does Asterisk offer this type of functionality, and if so how? Blake Parker -------------- next part -------------- An HTML attachment was scrubbed... URL:

<P>Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this posting.</P> <P><A href="http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html">http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html</A></P> <P>I've tried copying the config in this listing with no success. </P>

Hi Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? thx regards Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090807/ef9ba45e/attachment.htm

I'm trying to integrate my Asterisk box with my call manager 8 server. I can call from the call manager to a phone on asterisk, but I can't call from a phone on asterisk to call manager. Any help would be greatly appreciated. sip.conf [2000] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes [2001] type=friend secret= dtmfmode=rfc2833

Dear, i using this scenario. jitsi---> asterisk---->EPABX------> Local Telephone when i am calling from jitsi to no 88 its giving this message and getting busy tone. == Using SIP RTP CoS mark 5 -- Executing [88 at myphones:1] Dial("SIP/sandeep-00000004", "DAHDI/g0/88,20,rt") in new stack -- Called g0/88 [Nov 15 09:53:54] WARNING[3169]: chan_dahdi.c:7536

I'm in the process of testing a TLS/SRTP install. My experience is improving with each new challenge, but this one is a great test of my 2 month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the extensions and sip.conf files. *CLI>

I have tidied up the script and added some help text, feel free to download and maybe improve. http://www.masonc.com/phoneconf Usage: Usage: ./phoneconf [config|help] phonemacaddress extention username context ./phoneconf help will print syntax info ./phoneconf 0004f201aa11 500 MyPhone defaulot will configure the phone with that mac with one extension (500), and add the phone to sip.conf.

Hi I'm trying to implement a Callback mechanism whereby I generate a Call file and connect an arbitrary extension with my cellphone (via a SIP Channel). If I create a .Call file that connects the channel "SIP/12345678@Provider.net" with a local extension/context I get some weird issues with DTMF tones. I've set dtmf=2833 and the codec in use is G711a. For example - I create

Scott Robert Ladd has realized an analysis of GNU C and C++ optimizations, using a genetic algorithm to discover the most effective optimization flags for different algorithms. Check it out: http://www.coyotegulch.com/acovea/index.html AMD Optimized Windows XviD codec: http://net314.myphone.gr/xvid_amd.html Happy coding ! <p>>>Forward Agency In progress we (always) trust.