similar to: PRI D Channel and Caller-ID issue......

Hello, Does anyone know how to conference a call on the SNOM 200 phone? Whenever I push the cnf/tran button it just hangs up on the active call. The manual says you have to push the cnf function key but it doesn't appear in the lcd on my phone. Thanks -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! Has anyone on the board successfully installed a Mediatrix 1102 or 1104 as a SIP peer on Asterisk? I'm trying to configure different user accounts on each FXS port, but I'm having authentication problems; Asterisk is saying the client is not authorized. Interestingly enough, I can dial a "9" and make a local call through the Mediatrix. Thanks! chris --------------

Let's say you were wanted to terminate calls onto your Asterisk system but your only available codec was G.729 and you had no control over the remote SIP proxy sending you the traffic. What would you do? Does anyone have an update on Asterisk supporting T.38 with SIP? Thanks! chris -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! Net2Phone is getting a common SIP status code, "404 Not Found," when trying to place a call to our Asterisk server. We're hoping someone on the list can shed some light on why this is happening. We can process a call from Asterisk to Net2Phone without any problems. Net2Phone sends the INVITE but immediately gets the "404 Not Found." The "To:" field

If anyone could help me with this, I'd appreciate it! I've got an Asterisk deployment where I'd like to use an existing external Octel voicemail system. I've been trying to define an extension that if the call isn't answered in a few rings, to dial our external voicemail number. That voicemail system works by seeing the CALLED number and routing the call to the

We just did an upgrade on our Asterisk to the CVS version and our Mediatrix 1102s stopped working correctly. Our Asterisk is connected to the PSTN with a PRI. Calls from the PSTN to the Mediatrix 1102 work fine. The issue is calling out to the PSTN from the 1102. Asterisk looks like it process the call just fine except there is no talk path. Get this, though: If you flash hook and then

UPDATE. The circuit has run clean since the 7th. It seems the telco found a problem after all but they can't tell me what they did... cause they think it was fixed as part of another case. Grrr. Well, I never doubted the T400P too much but I went ahead and bought a different card just in case. Haven't had to use it so far. One of the biggest reasons I did not think the Wildcard was at

Gentlemen We are attempting to use * in a simple switching application: +-----> office lines | V LEC <--PRI--> * <--PRI--> modem bank (56k dialup modems) The problem is that (even with no office lines active) the modems have difficulty establishing a connection, the connection is slow (way too slow for 56k modems) and the connections are

I am trying to block/hide outgoing caller id on a PRI E1. It seems that it should be fairly simple, but it is defeating me. https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID says: "to hide your caller id, use: Set(CALLERID(num-pres)=prohib)" That doesn't seem to do it. The calls are originated from AMI and I have tried a blank "CallerId:" line and

This seems like a piece members of this list would find interesting... === There is growing concern over the interaction of VoIP systems with the legacy PSTN, and the transmission of caller identity data--most notably, Caller ID on the PSTN. It is not always possible, or obvious how, to handle Caller ID data when moving to or from VoIP and the PSTN networks. There are even business models

ktorrent Lec

Is there a way in asterisk to find the information in more than just a PRIs callerid field? for example the charge digits (who is paying for the call) and the original dialed digits, before any digit translation or forwarding? G. Phil

We have been pretty much able to solve our echo problems, except for the primary mode in which we desire to operate our system. See system diagram at bottom. Prior to making adjustments to cancel echos (all echocancel=no): Call Type Result (Before) --------- --------------- CP <- LEC PRI * TDMoE * FXO -> AP

Pardon my ignorance, but I was hoping someone could clear up something for me. - For a few POTS lines, digium has a single port card for that, or a T1 card to a channel bank. - For 10 or more lines, digium has a T1 or E1 card for that too based on PRI channels - For 100's to 1000's of lines, I suspect a soft-switch is in order??? A traditional phone company will sell: - POTS lines for

>Isn't that a KDE-specific program? Yes, it's from kde. >Works with Gnome as well? All kde programs work in all desktop managers provided that you install the required libraries. The same is true for gnome programs. LEC ----- Reply message ----- From: "Sorin Srbu" <Sorin.Srbu at orgfarm.uu.se> To: "Centos" <centos at centos.org> Subject: [CentOS]

Good Day all, I have a Fractional PRI connected to my Asterisk Box via a T100P card. When I initiate a call out to phone number 123-8888 the call sounds great no echo what so ever. If the person at 123-8888 hangs up and calls me right back (same handset on both sides) same trunk line The call always has echo on it. The Asterisk sip extension hears them selves echoing. The remote party

With ktorrent? No idea. Didn't even think of checking that one out. //Sorin Sent from my tablet, please excuse the brevity. Jeff Allison <jeff.allison at allygray.2y.net> wrote: What's missing? On 28/12/2014 8:30 pm, "Alexandru Chiscan" <lec at easterng.ro> wrote: > ktorrent > > Lec > > _______________________________________________ > CentOS

Hi Most of our customers have generic names like "Hospital", so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as "Reading Hospital" so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you.

Oh, well, there's no coulmns I can find to show various speed, trackers used, remaining time, ot able to sort on name, speed etc. Basically it's the gui I don't like. It's fine otherwise and does its job excellent. -- /Sorin ________________________________________ From: centos-bounces at centos.org [centos-bounces at centos.org] on behalf of Jeff Allison [jeff.allison at

Does anyone know if there's a service out there to -- for a fee -- inject our DID into the LEC's CLI database so a called party gets our associated name? /rg