similar to: Rewriting numbers while processing dial plan?

Dear all, I have ended up in a situation where CentOS 5.0 does not work for me - is it feasible to downgrade from 5.0 to 4.6 while the servers are up, or would the most sensible option be to just reinstall from scratch? Thanks in advance + best regards Jan

Hi gurus, In need of a little help here. I?m trying to do the Asterisk media release by using canreinvite=yes. But I found weird behaviour when comes to BYE. Below are my current setup: Client A is registered to Opensips Client B is registered to Asterisk A ? Opensips ? Asterisk ? B On hangup below are the SIP flow which I?ve notice from the Asterisk server itself: 1. Opensips forward the BYE

Hi! I am having difficultly in having users of various SIP devices obtain the correct behaviour when they call a busy number ie. only hearing the Congestion/Busy tone. I assume this might be because the SIP device itself generates the 'ring' tone? With my current setup in the dialplan extract (below) the user of the SIP device hears one 'ring' and then the busy tone if a number

Ciao a tutti ho appena iniziato ad utilizzare R per ora per attuare un'analisi geostatistica di dati. Volevo sapere come poter creare un oggetto gstat partendo da un file testo(che ho gia importato con read.table)e che contiene 3 colonne: x,y,value. Mi servirebbe far questo per costruire un variogramma. So che la domanda molto probabilmente per voi sara' banale....vi ringrazio

Dear all, I tried to make a call with extensions.conf. exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN}) exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME}) exten=> _00[1-9].,102,Hangup But the 2 and 102 will not be executed. So I can get the correct answered time via 2. Is any idea about it? Is it the problem of my ZAP channel's configuration? My zapata.conf is as below:

Hi, How can I have A*k convert a call from +441793xxxxxx to Dial 00441793xxxxxx instead? With the "_+." Below I can "catch" the call, but EXTEN doesn't get set as expected.. and then I need to figure out how to pass the call onto the outgoing-pstn context. Not sure if a Goto would work here... [outgoing-pstn-international] exten => _+.,1,Set(EXTEN=00${EXTEN:+1}) exten

Hi! I'm really hope you can help me solve a little mystery, the mystery is probably just my misunderstanding ! sorry... I've got an iaxy talking to my * box which connects to two providers. I'm running the stable release of the pbx. The only thing is that when dialling from the iaxy the ringing tone isn't heard while calling someone - you just hear silence then, they either

Hi. I am using R 2.5.1 on a Windows XP machine. Here is an example of a piece of code I was running in older versions of R on the same machine. I am looking for underscores and replacing them with periods. This result is from R 2.4.1: >gsub ( "\\_+","\.","AAA_I") [1] "AAA.I" > Here is what I get in R 2.5.1: >gsub (

I've got two patterns I want to match on making an outgoing call... (one day - to do Least Cost Routing for Cell/Mobile calls) Firstly - I prefer '0' rather than '9' to get an outside line... Either its a call to a mobile No... (072 -or- 082 -or- 083 -or- 084) or its just another number to dial... I added the following... the playback just advises me which 'route' is

Hi all, if i make a call between one SIP soft-phone to an other soft phone over asterisk, i get a Segmentation fault after take up. The extension is : exten => _00.,1,Dial,OH323/${EXTEN}@<myip>|60|r This means, if a SIP client comes with 00* then dial to <myip> over H323. If the H323 client takes up, a Segmentation fault occures. But, if the extension is exten =>

hi everybody, I try to record a call with *1 - one touch record feature in g723 format. exten => _00[1-9].,1,Set(TOUCH_MONITOR_FORMAT=g723) exten => _00[1-9].,n,Dial(SIP/${EXTEN}@ext-sip-account,,wW) I have chosen g723 format because in my CLI> show translation there is no translation between g723 format and wav (default for *1 feature). After pressing *1 sequence I have two

Hi, I'd like to setup oneway connection - so asteriskB can place calls on asteriskA and be safely authenticated with rsa keys. I just don't get any response on asteriskA. I've generated pair of keys: name.key, name.pub and put them on both servers - is it right to only have name.key on asteriskA and name.pub on asteriskB ? I get everybody is busy ... on asteriskB, and none

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

Hi, I start to be a little confused so I am asking to the list. I want to make with * a gateway from PSTN to H323, and to send all incomings call to a predefined IP, which will treat the h323 calls. let's assume that all my incoming numbers starts with 00 here is my extensions [incoming] exten => s,1,Answer exten => _00.,1,Answer exten =>

Hi everyone: The following code dials our prefix, sends a beep, and sends a DTMF "c" tone, then dials the phone number. I need to send the DTMF only if the phone is answered. [voip] exten=>i,1,NoCDR() exten=>i,2,Hangup() exten=>s,1,Wait(2) exten=>s,2,Background(beep||) exten=>s,3,DigitTimeout(6) exten=>s,4,ResponseTimeout(10) exten=>s,5,SendDTMF(c)

Hey. How would I set up my dialplan if a user wants to call its voicemail from an external phone? I'm thinking of getting the user to enter its mailbox number. Something like this: 1. User calls the dedicated voicemail number. 2. Phone prompts for mailbox number. 3. Voicemail(${mailboxnr}@context) Thanks.

Hello All, I am trying to connect to a third party's SIP based Hosted Billing service that supports G729 as I have found it difficult to get hold of prepaid billing solution for Asterisk. I was able to get an IAX2 connection working fine with another provider but as I am unable to find a billing service that supports IAX/IAX2, I need to connect to a hosted billing service using SIP and

I have a simple cdr configured with the default tables, here is a row of a good cdr report calldate | clid | src | dst | dcontext | channel | ect ..... ect .... 2008-08-29 10:16:49 | "C. BOUTON" <40> | 40 | XXXXXXXXXXX | phonesystems | SIP/40-08776938 | ect ..... ect .... I have replaced the number by

Hi all, I'm using VOIPJET to make international calls with an IAX2 connection between my local asterisk server and their server(s). At times I seem to have a problem if 5 or more international calls are made at once - I'm on a 1024kbps download and 256kbps upload DSL line (only the asterisk server uses this DSL line). Today I switched the codec from ulaw to ilbc in an attempt to lower