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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

Is anyone else having an issue with GXP-2000s and transmit gain? All my other phones are fine on my TDM400P with txgain set at 0, but the GXP-2000 caps at about a third of the scale in ztmonitor. I'm getting people complaining they can't hear me on my GXP-2000s, whereas my Snom 320 and Polycom 301 are great, and my Budgetones are overmodulating. Is there any conceivable fix on the

Had I have been smart originally I would have done this to start. Some rudimentary documentation above and beyond Asttapi 0.10's poor documentation is available along with the download at http://www.kirkhamsystems.com/asttapi. Clint -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kerry Garrison Sent: Tuesday, May 16, 2006 2:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: Asttapi f...

So are there any plans for bridged line appearance support in Asterisk? The new Linksys SPA9000 supports it. A lot of other VoIP systems from Nortel, Sylantro etc. supposedly support it. Seems to me that Asterisk needs to get on the bandwagon or be relegated to call centers, specialized voicemail applications, and phone chat businesses. It's not needed for companies used to PBX's but

I have been struggling with this issue for about a year now. There were just too many IP phones to choose from at all sorts of price points and not enough information about any of them. Now I am looking at the situation again and if anything it has gotten worse. There are even more phones and all sorts of opinions. For every person that says phone x is great there is someone else complaining

I've been racking my brain for the last two days to try to figure out what I could possibly be doing wrong in my configuration for a SIP trunk that's setup through my local ISPs Metaswitch. I've setup a very simple SIP Peer, which I've played around with a lot in the past two days but still comes back to the following basic setup: [provider-fireball] type=friend

I've finished a patched version of asttapi that will work with asterisk 1.2. There were fundamental changes to the Asterisk Management interface between 1.0 and 1.2 that broke asttapi. I think my patched version will work on 1.0 and 1.2 branches, but I have no way of testing since I don't have a 1.0 install nor do I want one :). I'm looking for testers, if anyone's willing to