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Same here. I called them yesterday plus email and still no reply. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of brett-asterisk@worldcall.net Sent: Friday, January 21, 2005 10:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Webmin Module for Asterisk (and thirdlane) Henry Devito wrote: > > www.thirdlane.com <http://www.thirdlane.com> has already written a > close ds...

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > brett-asterisk@worldcall.net > Sent: Tuesday, November 30, 2004 2:53 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] National (US) callerid name > resolution for yourasterisk box > > Hi All, > I've been in contact with a CNAM (caller name) directory pr...

Hey All, I'm just daydreaming here.. but what's the status of SIP-T in Asterisk? I haven't been able to find a whole lot of info on SIP-T but seems like just an extension of SIP. Right? Now if I had a PSTN Gateway (that is a SS7 gateway) that supported SIP-T, could I signal * with SIP-T from it and have asterisk utilize MGCP to sieze a particular DS0 on a remote DS1? Hmm.. What am

Dec 1 12:08:43 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! what does this

Hi All, I've been in contact with a CNAM (caller name) directory provider. Currently they offer 2 products. One is CNAM via SS7 and the other is Directory Assistance data via http/xml. I am very interested in getting the CNAM data via http/xml (or DNS TXT). I suggested this to the sales rep and he told me that I'm the second person in less than a month who has asked for this. What

Hi Muhammad From: "Muhammad Talha" <talha@worldcall.net.pk> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Wednesday, December 22, 2004 2:07 PM Subject: Re: [Asterisk-Users] Zaphfc/BRI Configuration help > Thanks for lan for your reply can you share your extention.co...

Can anyone point me in a good direction for configuring SIP through a PIX using 1:1 NAT. I have read anything I could get my hands on and tried them all with very little success. I can get it to work through the cheap little cable modem routers, but not this PIX. I -can- make a direct SIP call using the IP address of the * server (ie.exten@ipaddr), but when I do that * still doesn't

Hi All, I'm trying to hook up asterisk (CVS-HEAD-02/09/05-13:44:11 ) to a ADIT 600 via MGCP. Got it working really nice but now have a pretty bad problem: 1. When I perform a flash on the telephone, I usually get a second dialtone, but when I dial, dialtone doesn't break. If I flash back and forth a few times, it will eventually give me no dialtone.. here if I dial, it successfully

I am attempting to update my Asterisk installation from 1.0 to the latest stable version. When I use CVS checkout, I am receiving the following messages on chan_sip.c: RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v retrieving revision 1.510.2.25 retrieving revision 1.510.2.27 Merging differences between 1.510.2.25 and 1.510.2.27 into chan_sip.c M asterisk/channels/chan_sip.c Then, when

Dear all i am using Fedora Core 2 . i have Planet BRI TA with HFC chipset ( hisax ) i can easyly connect to internet using BRI but this card is still not recognized by asterisk i am using i4l driver . some people suggest i should try bristuff from junghanns.net any ideas ? Thanks and Regards Talha

Sigh.. This shouldn't be so hard. Ok guys, I'm trying to figure out how to support end user features for my users. Perhaps some of them are typical verticle service features like *69, *72, *66, etc, you get the picture. Here's my deal. Sure implementing them one by one is easy enough. But building the logic on the incoming side (PSTN calling my SIP customer for example) is a real

Hi All, I'm trying to record a phone call. I'm using the Monitor command with the "m" flag for a SIP to SIP call. I'm running: Asterisk CVS-HEAD-12/17/04-16:55:26 One side of the call is significantly quieter than the other. Am I doing something wrong?? Thanks, Brett