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Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number

...e channels we use (SIP, IAX2, Zap, ALSA). Can anyone help with this, or has anyone seen this? The mp3s play fine on any computer and haven't changed since they did work. Those wishing to hear for themselves, feel free to call extension 8800 at the number/addresses below. Thank you, Bryce Chidester Rhino Equipment Corp. bryce@rhinoequipment.com SIP: 305@rhinoequipment.zapto.org +1 (480) 940-1826 x305 IAX: guest@rhinoequipment.zapto.org/305 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/...

Hello All, I recently put in a zaptel 1fxo/1fxs card. I am experiencing heavy static. Even with the pots line disconnected, if I do a dial I still get static. This way I know it's not the line, but rather something on the card. I tried alternate pci slots. This card has a power connector, does anyone know what the power requirements are? The unit is in a small case with a 2.4ghz p-4 and

I was just browsing Digium's web site and noticed they are taking orders for the new IAXy. Has anyone purchased and tested one of these yet?? I have thought about buying one for testing, but want to make sure it isn't going to be a flop like the last one. Robert

I would like to setup a test number, that speaks back my phone number. How can I set this up? bye Ronald

I have asterisk with two zap channels which are analog ports off a T1. They each have a inward DID number If they are used for outgoing they show the T1 main number not the DID's number. Is there any way to send caller ID of the inward DID number not the main number Jeff

...faxes is that, a document feeder machine is a must, and although internet based fax sounds good, when your in a rush it just does not cut it. Regards, Greg -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Bryce Chidester Sent: Wednesday, August 03, 2005 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk TDM card connected to phone linesAND fax line On Wed, 2005-08-03 at 10:23 +0000, Andres Tello Abrego wrote: > Assign an extension to the fax at extension.c...

Hi ... probably one of those RTFM kind of questions (while I'd be happy to know where a good reference "FM" is :-) ) Has anyone an idea on how to disable the console sound driver. My problem is that a running asterisk is muting my speakers. Thank you in advance for your help Conrad

Hello! I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks. We are an outgoing call center with 30 internal analog phones hooked up to 2 Rhino CB24 channel banks. The banks are connected to a Rhino R4T1 card in a Dell 2950 server with 8 gigs of RAM. The 2 other ports on the R4T1 go to our 2 PRIs. In this configuration, we have trouble maintaining stability. It may be fine

Hi, What do I need to do to get asterisk to NOT pickup a Zap channel when it rings? The channel in question is used for outbound calls only, and all incoming calls are answered by an analog phone elsewhere in the building that does not run through asterisk... so.. either make it not answer.. or make it delay for like 90 seconds.. I've tried wait's.. but it still seems to pickup the

Hi All, I am looking for a GSM VoIP gateway for use with Asterisk. I have come across VoiceBlue by 2N but it's price is beyond my reach. Are there any other alternatives out there? I've scanned across the mail achieves for an answer to this without much success, if the question has already been answered kindly point me to the resource. Allan.

Is there a softphone for windows that supports IAX? I can't seem to find anything out there...maybe im looking in the wrong places... Jeromy Grimmett VoipEmpire.com jeromy@voipempire.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/e668136a/attachment.htm

Hi, I have to connect 30 phone lines to my asterisk server, can somebody help on how I have to do it ? I have a TDM405P and one TDM400P with 4 FXO ports. Do I have to use 8 TDM400P ? Or, is there another way to do it ? Thanks, Angel.

Hello list, Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten => s,1,Dial(SIP/74118@193.136.252.5,30,r) but this way all calls go to 74118@193.136.252.5 ..... Then I tried: exten => s,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r) but this way, the