similar to: Dial DTMF after bridging call

Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip >< sip TNT pri >< pri asterisk The TNT is running 11.0.6 and the asterisk servers are running 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to asterisk but not the other way. The call from asterisk to pri to tnt is good, the TNT is passing SIP invite to the

I'm trying to get inbound calls from a TNT working but get 407 errors from the TNT. This is what I have in sip.conf: [maxtnt] type=friend host=x.x.x.x dtmfmode=rfc2833 callerid="MaxTNT" <maxtnt> context=demo qualify=yes disallow=all allow=g729 allow=ulaw insecure=very This is what the TNT is spitting out: Jul 24 14:55:12 tnt1 1/17: Releasing

I'm dialing DTMF's in a SIP channel using the options: [sip.conf] dmtfmode=info [extensions.conf] exten => _XXXXXXX,1,Dial(SIP/gateway,,D(${EXTEN})) (this is a custom SIP gateway, which receives the DTMF's sent from softphones through Asterisk, and based on them, build the destination PSTN number). My problem is that Dial send the DTMF's to the SIP/gateway user at a rate

Hello, I'm currently testing Asterisk over a T1 cross connect to a MaxTNT chassis that we have. It is working fine switching the calls through, but there is about a 10 second delay from the time Asterisk initiates the call until the TNT accepts it. It appears to be a ANI issue, I've changed several settings and formatting options on the T1 between the two, as well as turning on/off the

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian

Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip<->h323 gateway to tie SIP & H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past

I''m using will_paginate in a Post class method; [code]def self.list(page) paginate :per_page => 10, :page => page, :include => :user end[/code] I can happily use the method on my posts controller index action; [code] class PostsController < ApplicationController def index @posts = Post.list(params[:page]) respond_to do |format|

It appears that no matter what I place into the <encode></encode> section of my ices2.xml, the stream is re-encoded. If I remove the <encode></encode> section alltogether my ogg files are streamed as is but the status page no longer shows the correct information. So what is the proper way to configure a stream that just 'passes through' the ogg files at whatever

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I've been unable to get app_transfer to work, could someone explain how it is supposed to work? Currently I have two Asterisk boxes. A call comes in via zaptel to ast1. ast1 dials ast2 using iax2 and gets instructed to transfer the call to a different extension. iax2 debug shows that a transfer cmd is sent to ast1, but nothing happens

I'm looking for someone familiar with setting up some of the more advanced features of the Lucent TNT, preferably someone with knowledge of Trunk Groups and choosing outgoing PRI channels based on call type and perhaps NPA-NXX We currently have 8 PRI's. 7 of them are for our dialup pool, the 8th is for our voip. We currently run the dialup PRI's to a seperate TNT We want to

I've been using Asterisk for a few weeks now. I have a (1) BT100 phone and a Sipura-2000 for all my analog phones. All has worked rather flawlessly, until today. I was on the BT100 phone today. During my phone conversation, the BT100 disconnected and went into a "click" mode. 2 "clicks" per second I think. Asterisk was fine, I picked up one of the analog phones,

I am using a grandstream phone with g729 and alaw odecs and in both modes I cannot seem to pass dtmf's, neither inband nor out of band, neither wthrough a lcoal server nor through a natted connection. Am I missing something ?

Hello, I will expose my problem here. Please tell me if it is not the right place as I am really new to that list. I use Asterisk as a SIP proxy. I have two users connected to it, Let's call them 1234 and 5678 In my dialplan I have two lines: exten => 1234,1,Dial(SIP/1234) exten => 5678,1,Dial(SIP/5678) The SIP phones (X-lite) are configured to send DTMF's using RFC 2833

Hello, Before throwing in the towel with my Sipura 3000 has anyone had much success with that adapter connected to a door phone? In our setup a doorphone is connected to the SPA's fxs port. When a visitor rings, asterisk calls a group of Polycoms and the person who answers has to enter *1 to trigger the door opening. However it seems the SPA doesn't relay the DTMF's to the

Hi Chris, On Sat, Jun 07, 2008 at 02:59:03PM -0700, Chris Lattner wrote: > On Jun 2, 2008, at 1:03 PM, Matthijs Kooijman wrote: > >> Can you give some background about what kinds of things you're > >> thinking > >> about for this? > > For example, when I have a function returning {i32, i32} and I want > > to add > > another i32 to that. If

I need to execute account number, device number after dialing main number; what is the best solution? Is it possible to pause during dialing. Dial 1-800-number press 1 for English wait 5sec enter device number wait 5sec enter device ID What are my best options? -- #Joseph

I've narrowed my scope problems with predict.coxph further. Here is a condensed example: fcall3 <- as.formula("time ~ age") dfun3 <- function(dcall) { fit <- lm(dcall, data=lung, model=FALSE) model.frame(fit) } dfun3(fcall3) The final call fails: it can't find 'dcall'. The relevant code in model.frame.lm is: env <- environment(formula$terms)

Howdy, I need some pointers (ideas or help) to build a solution to retrieve recordings out of an IVR. The program needs to do some predictive dialing functions I only need it to: 1) Be able on it's own to make a call (to the same number inside this script). 2) Detect that the call has been answered and that it has finished. I need some sort of AGI but it needs to run on it's own, i.e.

Dear Devs, (I hope this question is not that much OT for this list..) My question is about fast OpenGL context switching, i.e. when there are several processes using the same nvidia card, each one with their own OpenGL context. In my specific case, I am trying to dump 720p video simultaneously to multiple windows using OpenGL textures. So, to begin with, I have a process that spans child