similar to: Socket 478 Motherboard for use with TDM400P

I'm considering upgrading, and am trying to choose a modern motherboard that nevertheless fully works with Centos5/RHEL5. Though I'm partial to AMD processors, and would like to use one in my new configuration, that's not a requirement, if the most appropriate board happens to be intel-compatible. However, given the way ASUS is dumping LInux and crawling more firmly into bed with

Howdy, How can you tell if RTP traffic has been reinvited/is bypassing an * server? Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com

Has anyone had problems getting their preffered codecs on the Teliax web interface taking effect? I have two accounts, two separate yet similarly configured * servers. On one account the settings took right away - on another server I am getting no result. In fact, no matter what I change the settings to, only the old codecs are usable (otherwise * says it can't negotiate a codec). Teliax

Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24 hours or so. Since upgrading to 1.2.3, though, the whole system has locked up twice. Once on Thursday, and then about a half hour ago. The server would reply to a ping, but no ssh login, no local console login - just locked up. This ain't good for

I have one use on our PBX who has been experiencing seemingly random disconnects. The user is on the same LAN as everyone else, using the same type of phone (79XX loaded with SIP firmware) as everyone else. He had some disconnects a few weeks ago, I suspected the phone, so I swapped his with mine. I have since not had issues with his old phone, however, he has had issues using mine. So, the

Dearest List, I guess I missed this point: Is it true that if you change the echo canceler in zconfig.h, and then recompile/install your zap modules, that for this to be taken into effect by * you must then recompile/install *? I would have figured that the zap echo cancellation method was independent of *, and I don't recall seeing any docs mentioning either way. Sincerely, Brent A.

Is it possible to dial more than one extension with a different CID to each extension? I'm thinking macros might be needed, but I don't have a good handle on macros. Is it possible? Any hints? BTW - this would be used for showing an internal extension to one phone and a PSTN accessible number to another phone. Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road

Howdy: Does echo only occur on analogue PSTN lines, or can it also occur on PRI and BRI lines? If so, for the same reasons? This is a part of our consideration to transition to BRI. Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com

I am trying to use a the SIP tapi from www.enum.at <http://www.enum.at/> . This works fine from all kinds of applications which support TAPI, like outlook and Dialer Pro. However when making tapi controlled calls, the signaling to and from PSTN seems to fail. I have used the digium hardware ISDN PRI boards, but also a SIP gateway. Both result in a audio message from asterisk

Hi, I have got my setup almost how I would like it now, but I have just two last remaining issues that I cant seem to find answers too so i'd be grateful if someone could help? 1) Since upgrading my Cisco 7960 SIP phone to P0S3-08-2-00 the phone now displays the IP address of my asterisk server alongside the caller ID of the incoming call. For example "0123456789@192.168.0.1",

I am thinking of turning on the NAT option in our Cisco phones (and the corresponding sip.conf modification) to allow the phones to be taken outside the LAN. Can anyone think of any reason not to just always turn on the NAT enabled option? I can't think of a reason not to always operate these phones with this enabled, since it would likely allow them to be taken outside our LAN and used.

Hola, What would a valid regexp in Asterisk be to identify a NANP number, i.e., NXXNXXXXXX? Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:brent.torrenga at torrenga.com web:www.torrenga.com

I think A@Home allows a user to search a directory by either first OR last name, right? I don't know for sure since I don't run A@Home. I would like to offer that functionality in my system - and I'd have done it by now if there was a prompt where Allison asks "press 1 to search by first name, press 2 to search by last name". But I don't think that prompt exists. Can

Out internet connection was out this morning. It seems that the SIP extensions on our LAN were affected. Behavior like: Call comes in over POTS to a TDM400P, there is a delay then before the Cisco 79[46]0's start to ring. If we were lucky enough to get a call through, then we could not transfer the call, or place the call on hold, or park the call. Outbound calls seemed to have a delay

We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This should get rid of static on the line (at least any static generated by our half of the circuit), right? I am a total virgin to ISDN. I understand that we need two BRI circuits to provide four voice channels, and that the hardware to speak to the BRI circuits can be passive or active, with the active type being much more

Does anyone know how to rewind the music on hold? Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060202/ab26b18f/attachment.htm

I'm used to seeing the same versioning (maybe I've been gone too long) Is zaptel 1.2.15 the right one for asterisk 1.2.16 ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070314/e57842ef/attachment.htm

Previously, when I wanted to forward to incoming callerid when I forwarded a call to another number I had to set the callerid on the outgoing call to be that of the incoming number. So today I do this: exten => s,n,Set(CALLERID(name)=${CALLERIDNAME}) because I want the outgoing callerid that I forward to not be the normal callerid of the local extension but I want to forward the incoming

Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. Benoit Panizzon -- I m p r o W a r e A G - System Services ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz

Jean-Louis curty to Asterisk More options 4:38pm (7 minutes ago) Hi everybody, I'd like to know if anybody tried to write a xml doc to monitor the number of calls in Q, when working with an ACD it's convenient to see how many calls are waiting so the agent can speed up the conversation when it gets too busy :-) I was wondering if it was poss to display this info on a display of a