similar to: Parallel dialing / running dialplan process in background

Asterisk will try calling both at once. As soon as one is answered it cancels the call to the other. What you can do is for extension 101 to put it in it's own context and then call the agi from the h extension. So something like this: [from-internal] exten = 514316XXXX,1,Answer() same => n,Playback(hello) same => n,Dial(LOCAL/100 at extensions&LOCAL/101 at extensions) [extensions]

Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3

I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls

ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and archive w/o good results. Thks in advance for any help, Dave sip.conf -------- [general] port =

Hi, Can a call to multiple devices be cancelled in all of them at same time? With next dialplan, exten => 100,1,Dial(SIP/101&SIP/102) when a call rings on 101 and 102 and one of them rejects the call "with 486 Busy here", is it possible to reject the call in the other device at same time? I read application dial options but I can't find any that can help me to achieve this

Hi All, I am a new bee to *. I just installed Asterisk@home on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message "All routes are busy". In

Hi All, I have registered "abhijit" for SIP in asterisk Server. I am able to register my softphone (SJPhone) to the server using the name "abhijit". But whenever I try to make any calls I am gettinh the following error message:- *CLI> -- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120 May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper:

I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and fxoks configurations without avail. This is a single asterisk@home system with a single T1 card. Robbed Bit T1 ami, d4. ------------------inbound call

I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink T1 ---- Asterisk. I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk extensions over the T1. I do not get DID nor CID on the Asterisk, so I want to use PRI between the PBXs. I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are different cards) I see this as my least

I have a little confusion about the general settings (other than the register values) in the SIP General area. I understand that for examle in a SIP context like [FWD] or [BROADVOICE] the entries in those areas are ths settings that take effect in any communication woth FWD and/or BROADVOICE. However, I'm confused as to the purpose of the "general" settings -- to what or which

Hi, Here is the third version of memory hotplug support for Xen guests patch. This one cleanly applies to git://git.kernel.org/pub/scm/linux/kernel/git/jeremy/xen.git repository, xen/memory-hotplug head. On Fri, Aug 06, 2010 at 04:03:18PM +0400, Vasiliy G Tolstov wrote: [...] > Testing on sles 11 sp1 and opensuse 11.3. On results - send e-mail.. Thx. On Fri, Aug 06, 2010 at 12:34:08PM

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

Hi, We are using VOIP-SIP gateway to route outbound PSTN calls. Recently, I am getting == No one is available to answer at this time message, after making 5 SIP attempts (Retransmitting #5 (no NAT):), and the calls are going out through alternate Zap-trunk. I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls. Strange thing is that this is happening randomly,

I use Jack for getting callee sound. Dial with option M(): JACK_HOOK(manipulate,i(rec_737219:input),o(rec_737219:output),c(rec_737219))=on This works fine, but I need to connect the sound channel to Jack *before* the actual answer. As you can see in the AMI log, between "Ringing" to JACK_HOOK there is a 6 second break. I don't want that. I need a way to launch Dialplan function

I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place calls inbound, everything works fine. If I place calls outbound, originating from the Asterisk box, everything works fine (I have done this with the use of the .call files). If I setup an extension with the findme-followme feature and have it try to hair-pin a call back out the same trunk to the Avaya, I get a

I''ve been working on this on the odd occasion, I think it mostly works, or it did last I tried which was a while back. I''m sure it is too hacky in places. My plan was to clean it up on the next test day. I''m mostly just sending this for Wei''s benefit since he is independently looking at adding Debian HVM guest tests for OVMF purposes. Ian. commit

Hi all, I often record radio shows for posterity, and sometimes I have friends who would like to listen to them live over the net. When I am recording for my own purposes, I use a command of the form: "brec | oggenc", (options omitted for clarity) and I send the output to a file, call it radio.ogg. Locally, I use Debian stable. If I want to listen to the show as I record it, I can

I've been having a problem where ices-kh (the jack'ified version) disconnects from its jack input source unexpectedly. This happens mainly while other jack clients are being started/stopped, or connected/disconnected, but also at other times (e.g. switching between different X sessions). I'm planning to do a bit more work on tuning up the jack setup to see if I can get rid of the

Hello, I need to capture calee's audio in real-time in order to capture operator messages (I've written sound recognition software that works with Jack: http://github.com/Motiejus/SoundPatty/). Jack does the following: Incoming call audio -> audio in to jack, audio out from jack -> current Asterisk application Outgoing call audio <- current Asterisk application However, I need