similar to: Asterisk ISDN: Problem Setting CallerID as DID Extension Numbers.

this happened to me on a cvs update, rebuilt a clean chan capi cm and all is well. Greg ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Voicomm User Sent: Monday, September 19, 2005 3:29 AM To: Armin Schindler Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk ISDN: Problem

I am brand new to Asterisk. My question is if I want to have multiple participants all listening, or listening and talking, do I need to have a separate telephone line for each, or can they all dial in using a single telephone number and a single line? Thanks, John Fistere -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have a 100 number indial range and every so often get fax calls on our voice numbers (our fax number isn't in the 100 number range). If you just hang up the sending fax will often try a few times before finally giving up. Our outgoing fax is connected to the PBX (not asterisk), and we can do a blind transfer to that which will print it out, but right now the fax is printing a misdialled

Hello, I am using chan_capi and would like * to behave differently depending on the MSN the caller dials. Is there a way to route incoming ISDN calls to different contexts based on the MSN dailled? I have tried something like msn=1234 incomingmsn=1234 context=msn1 msn=4567 incomingmsn=4567 context=msn2 in capi.conf but with no results. Thanks for any hints. -Walter -- Walter Doerr

The company I work for has gotten the go ahead to start dipping its foot into the shallow end of the asterisk pool. The client we will be setting up for currently has an NEC PBX of some sort with 8 analogue lines in. They use lines 1-4 as indial on a rotary group, lines 5-6 as indial for two 1800- numbers (1800 is Australian prefix for freecall), and lines 7-8 as outgoing. There are approx 15

Hey Guys I have an indial range of 61211118[01234]X being trunked sip to xxx.yyy.189.65 Now I want to break this down into 612111180x going to xxx.yyy.188.145 and 612111184x going to xxx.yyy.189.199 reminder being used for fax->email etc etc etc I have created the outbound routes and sip trunks I can see that all the sip trunks are up I can see the outbound routes are there and

the Telco, I can make calls in. What I am trying to get though is how to pass through the DID range. The E1 that I am connecting to the Telco with, used to connect direct to the NEC system and already has hunt group calling enabled for all 30 channels. Also, I was given a 100 number indial range (from 00 -> 99). If the E1 is connected to the NEC directly, I can call 5555 7320 and the NEC

This is an email to all New York based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many

Hello all, I have the following working (somewhat) setup: TELCO | | E1 (30 Chan -- TE210 SPAN 2) | | Asterisk box 1.6 with DAHDI drivers loaded Digium TE210p | | E1 (30 Chan -- TE210 SPAN 1) | | NEC PBX

Has anybody had a problem registering their IAXtel account? I just signed up for an account and followed the documentation on iaxtel.org and my registration is always rejected. When I type "iax show registry", I get the following output: Host Username Perceived Refresh State 12.37.165.130:5036 xxxxxxxx <Unregistered> 60 Rejected

Grandstream owners, I just noticed that there is a new firmware release, for those that are interested: http://www.grandstream.com/BETATEST/ Doug

Hi, I have a legacy Norstar system that I'm looking into integrating with my Asterisk setup. My first attempts have worked, which involves a Wildcard X100P FXO card in the * box connected to the Internal ATA (FXS port) on the Norstar system. Calling from SIP -> Norstar works fine, since the SIP caller initiated the call and generally will be sane enough to hangup the phone when

http://bugs.digium.com/view.php?id=4760 If you use the L() option on dial and say the latest CVS-HEAD in the past month you're potentially getting screwed out of a lot of money. We originally wrote the L() option for dial and it worked great till someone came along and hijacked the timer for something else thus causing the L option to fail/reset the timer to zero thus causing it to

I'm trying to get a basic Asterisk configuration together for ISDN incoming / outgoing calls. I have two Cisco 7905g phones working (at least talking to each other) and have purchased a NETjet-S PCI ISDN card for routing calls to / from ISDN. The state I've managed to get it to is:- -- Executing Ringing("SIP/PHONE2-d557", "") in new stack -- Executing

Software zur Erkennung von "Spam" auf dem Rechner priamus.teamware-gmbh.de hat die eingegangene E-mail als m?gliche "Spam"-Nachricht identifiziert. Die urspr?ngliche Nachricht wurde an diesen Bericht angeh?ngt, so dass Sie sie anschauen k?nnen (falls es doch eine legitime E-Mail ist) oder ?hnliche unerw?nschte Nachrichten in Zukunft markieren k?nnen. Bei Fragen zu diesem

I think it's time to 'ping' John Todd and Digium on this topic again. What happened Digium? Why did you say you were going to take this project on but have not come back to the community with an answer yet? Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial).

I am having problems getting incoming caller id to work on a Telstra Onramp 10. I have changed "/DEFAULT_CIDRINGS 2"/ Is there something i'm missing ? My Cisco 7960 just shows "asterisk" Thanks, Nathan [zapata.conf] context=incoming usecallingpres=yes relaxdtmf=no rxgain=0.0 txgain=0.0 busydetect=no pridialplan=local usecallerid=yes callerid=asreceived

community there is a real possibility this may come off so if you have an interest in this space and want to contribute to the discussion then this is your opportunity to do so. =20 I look forward to all opnions on this topic. =20 The slide deck for the agenda of this call is located here http://voipusersconference.org/2008-05-09-Slides=20 Cheers, Dean=20 ________________________________

Hello all. I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Ie, pick up NEC handset, dial