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...LSA). 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/attachments/20050519/13cf4f29/attachmen...

...u may consider it. http://www.mhspot.com/mhspot/sippyskype.htm - -- James Finstrom Rhino Equipment Corp. All Rhino products are made in America, Come with a Money Back gurantee and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -----BEGIN PGP SIGNATUR...

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...t; Disconnected from Asterisk server <<< dead... Thoughts? James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-877-RHINO-T1 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -------------- next part...

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

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

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

Just saw these new Asterisk compatible cards at Rhino's homepage. Anyone has experience with these babes? http://www.rhinoequipment.com/index.html http://www.rhinoequipment.com/t1card.html The 4-port FXO looks interesting. Looks like a very clean design. Leo

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

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

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.

Greetings list, I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel

I have the original version of the IAXY. I had it laying around collecting dust, now Im actually putting it to use. When I call my voicemail extension (8500), Before I get the voice prompts from the voicemail app, I hear tones that sound like the caller id tones that are heard when montoring a phone call. While watching my Asterisk CLI, I see this error at the sound of each tone: Jul 21 23:06:03

...where s automatically dials the fax machine on 4. You can still use 1, 2, and 3 for outbound if you group them and dial with one of zaptel's grouping options. Other idea being to make sure the fax machine picks up first, but this issue's been discussed on the list before. -- -Bryce bryce@rhinoequipment.com NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@...

I'm attempting to set up call recording with Asterisk. Using automon => *1 ; One Touch Record in features.conf does not appear to be working. I'm using Polycom 501's but when someone dials *1 while in a call, nothing happens. I'm wondering if the phone or Asterisk is even detecting the DTMF. I suspect that is the problem but don't know how to verify or

So I decided, for the formal asterisk rollout, to change over to less commercially-infringing MOH than the prior admin had thrown on the server. (plus: it was blown out and nasty sounding over the phones. Ew.) I changed the files in /var/lib/asterisk/mohmp3 to something else (can't dig up the link, but it was from the voip-info wiki). My musiconhold.conf looks like this: ; ; Music on

I have tried to get MRTG to graph my Asterisk box but have run into a problem. When I run the perl script provided at: http://karlsbakk.net/asterisk/ I get the following error: [root@tsr asterisk]# ./asterisk-mrtg -h myasteriskip.mydomain.com<http://myasteriskip.mydomain.com>-v -1 SIP -2 IAX2 -u 109 -p xxxx Asterisk Call Manager/1.0 Action: Login Username: 109 Secret: xxxx Response: