similar to: Sharing a dialplan

Hi, I have two sites and I'd like to connect them with a IAX trunk and share the dialplan. Extensions cannot be clearly separated. Do I need to use 'switch' statement or DUNDI/e.164? Using 'switch', does any user can call any extension on both sites? Thanks Mimmus

Greetings all, I'd like to start implementing a private DUNDi peering group between one of our asterisk servers hosted at a datacentre and the various asterisk boxes sitting at clients' premises. On most of the clients' boxes the dialplan will have an [in-pstn] section containing the various numbers that should be recognised by that box. Where they're from a VoIP provider they

Hello, I would like to known how to use DUNDi with a Lua dialplan ? In extensions.conf, we should do like these: |[lookupdundi] switch => DUNDi/priv [internal] include => dundiextens include => lookupdundi exten => _XXXX,2,NoOp(calling ${EXTEN}) exten => _XXXX,n,Dial(SIP/${EXTEN}) exten => _XXXX,n,Hangup()| priority 1 is either defined in dundiextens (local registered

Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani

Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus

Hi, I'm actually using a slightly old version of AAH with Asterisk 1.2.1, because at first install it was perfect for my moderate knowledge of Asterisk. It is working well but I gradually introduced many changes to dialplan during normal use and now I'm feeling like in a straitjacket! Moreover I'd like to have the chance to upgrade Asterisk regularly. I have not the experience to

Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR

HI, I tried to configure Asterisk 1.8 on one of my test-hosts. I've installed from centos-asterisk.repo (http://packages.asterisk.org/centos/$releasever/tested/$basearch/): Nov 26 15:34:56 Installed: asterisk-sounds-core-en-gsm-1.4.20-1_centos5.noarch Nov 26 15:34:59 Installed: asterisk18-core-1.8.0-1_centos5.i386 Nov 26 15:35:02 Installed: asterisk18-voicemail-1.8.0-1_centos5.i386 Nov 26

Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial("Zap/2-1", "IAX2/215|20|TtwW") in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) --

And I'll resend this one too. Silly scalix. --Rob -----Original Message----- From: Rob Thomas Sent: Thursday, 22 June 2006 12:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] zapata.conf: recent changes? Looks like you've stopped compiling libpri. All those options that are being ignored, are being ignored because they're

It seems that I am having a heck of a time explaining my attempts at distributing ACD Queues to the list. Here's one little problem, that's only a piece of the puzzle. dundi.conf: 180q => global_dundi_q_pbx1,100,IAX,dundi1:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => global_dundi_q_pbx2,200,IAX,dundi2:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q =>

At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees, each with their own extension and DID, and at headquarters, we have about 70 people, again

All, I made some progress, but it seems the further I go with clustering the harder things get. Hmmm, I guess if it were easy, it would be documented...... Anyhow, I have 1 * server as the DUNDi peering master with a ttl=1. The only function of this server is to lookup where other sip peers are registered and forward that info on to the requesting * server. I have 4 * servers accepting

Hi All, I've been banging my head on a small dundi problem... I have two * servers setup, both have almost identical dundi.conf files: root@tsjonge:/opt/asterisk/etc# cat dundi.conf [general] department=thuis organization=pipsworld locality=Amsterdam stateprov=NH country=NL email=remco@pipsworld.nl phone=+31207508308 ;bindaddr=0.0.0.0 ;port=4520 entity=00:02:b3:49:69:5e ttl=16

Hi All, I have a really strange issue occuring where if I run "show dialplan" or "dialplan show" or "dialplan show parkedcalls", then asterisk dumps core. It only appears to happen with contexts that are created within res_features. I am able to display all my other dialplans, but, every time I try to just do a normal "dialplan show" asterisk core dumps

Hi List. We have a small issue with making parked calls work with the new Asterisk 1.4. I have an impression that this used to work with 1.2, so its either I'm doing something wrong, or a regression. I hope its not the latter and you can tell me what I'm doing wrong. The setup is an Asterisk with sip users in mysql realtime and dialplan in mysql static (mostly - some stuff is built-in).

Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani

Who said I was a C programmer? -----Original Message----- From: Terry Wilson [mailto:otherwiseguy@gmail.com] Sent: Wed 6/14/2006 6:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] DUNDi Not Able to Handle Complex FailoverSituations pbx/pbx_dundi.c in dundifunc_read(). shouldn't be too hard to have it set some variables (i.e.

> I'm having an issue deploying softphones into my DUNDi/regcontext > setup. My current design is that all SIP users get registered into a > sipregistration context in the sip.conf. I then have a dialplan > function that includes that and does the dial: > > include => sipregistration > exten => _XXXX,2,Answer() > exten => _XXXX,3,Wait(1) > exten =>

At the moment I'm out of the office, but when I return I'll be certain to do that. Note that my solution is different from what you are working on with regexten, though I suspect some of the challenges that I've faced and overcome are not. I'm actually using UltraMonkey for load-balancing and failover of the Asterisk boxes, and my dialplan is set up so that it need not be changed