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Hi all, I have an external application commanding asterisk by AMI and AsyncAGI. I also have a dialplan like this: ; AsyncAGI extensions exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN}); exten => _8.,n,AGI(agi:async); exten => _8.,n,Hangup(); ; Meetme extensions exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT}); exten =>

Hi, I'm bringing this discussion here from http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ about how to manage stopping a playback on a extension previously launched with AsyncAGI and redirecting the call to another exension. If I make the Redirect without a playback, the Redirect works: http://docs.google.com/Doc?id=ahfnfrcrh3rr_30f7...

...configured in high impedance mode (I don't know if Digium or other vendors boards expose that functionality to users, but you may want to test and find out if works). More detailed instructions can be found at Sangoma's site or my blog: http://wiki.sangoma.com/sangoma-tap-system http://www.moythreads.com/wordpress/2009/09/26/sangoma-tapping-solution-for-asterisk/ The patches are already out there in the bug tracker along with some SVN branches. https://issues.asterisk.org/view.php?id=15970 https://issues.asterisk.org/view.php?id=15971 I'd love to get feedback in the bug tracker in order...

Hi, On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest. Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly 1-2 concurrent calls. No other activity on server. Top shows asterisk on top. Its quad xeon server with 4 gb ram. Any suggestion where should I start and how should I go about with my investigation. Thank you and have a great weekend. Sans

Hello everybody, i'm from Mexico, at the time i?m working on a production server with asterisk 1.2.25 + spandsp-0.0.4 + libmfcr2-0.0.3+libsupertone-0.0.2+libunicall-0.0.3 and zaptel-1.2.22. I installed this version of astunicall that i downloaded from http://www.moythreads.com/astunicall/ Everything works fine, i'm able to make outgoing calls and recive incoming calls with all ANI and DNIS digits, except for International incoming call. My phone provider(Telmex) gives me 10 digits of ANI and 4 digits of DNIS, that i?ve configured on my unicall.conf. My main iss...

Hi, I'm running Asterisk 1.4.2 on a Debian GNU/Linux and a Digium TE210P card. I have one E1 with MFCR2 Signaling. I compiled asterisk + libunicall, and I can make calls over E1, but can't receive. Here the CLI when I make a call: -- Executing [006191642208@ps5:1] Dial("SIP/23-081cbc40", "Unicall/g1/91642208|50") in new stack -- Called g1/91642208 [May 14

Hi List, I'm trying to connect Asterisk with a Nortel Meridian using an E1 with MRFR2 signaling. I've connected both cards, and compiled all the required software... The problem is every call (outgoing or incoming) got dropped, complaining about some "T1 timed out" Only for testing purposes I'm using an application called testcall included on the lib-unicall package, and

---------- Forwarded message ---------- From: Brian Hutchinson <b.hutchman at gmail.com> Date: Fri, 11 Jan 2008 12:22:09 -0500 Subject: Trying to build MFC/R2 Hi Moy, I downloaded your astunical-1.4.16 and have been studying it. I have a machine running the latest download of asterisk-1.4-17 and was trying to figure out how to build MFC/R2 support into it. I merged your unicall specific

...I also can place calls to the telco.. but I can't receive any calls from the pbx. When receive a call from the pbx I get this: cause 32771 - T3 timed out If I connect the pbx directly to the telco there is no problem, the calls are stablished correctly. Im using the package at: http://www.moythreads.com/astunicall/downloads/ http://www.moythreads.com/astunicall/files/astunicall-1.2.25-0.1.tar.gz that contains: asterisk-1.2.25 spandsp-0.0.4 unicall-0.0.5pre1 libmfcr2-0.0.3 libsupertone-0.0.2 libunicall-0.0.3 zaptel-1.2.22 My zaptel.conf is this: loadzone=mx defaultzone=mx spa...

Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The

...and hold without press any digits, after some seconds you got the congestion signal. Just for consideration I receive call without any problems, provided that performed the first setup. I have use http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 to do my configuration, sources are: http://www.moythreads.com/astunicall/files/astunicall-1.4.9-0.1.tar.gz zaptel-1.4.4-6 asterisk-1.4.9 libsupertone-0.0.2-1 spandsp-0.0.4-1 libunicall-0.0.3-1 libmfcr2-0.0.3-1 The only difference is I have use the sources to make a SRPM -> RPM files on CentOS 5. Here is my config files: zaptel.conf loadzone...

...after some seconds you got the congestion sign= al. > > Just for consideration I receive call without any problems, provided that > performed the first setup. > > I have use http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 to do my > configuration, sources are: > http://www.moythreads.com/astunicall/files/astunicall-1.4.9-0.1.tar.gz > > zaptel-1.4.4-6 > asterisk-1.4.9 > libsupertone-0.0.2-1 > spandsp-0.0.4-1 > libunicall-0.0.3-1 > libmfcr2-0.0.3-1 > > The only difference is I have use the sources to make a SRPM -> RPM files= on > CentOS 5. &gt...