search for: tellink

...729 Calling from (SIP)UA through asterisk towards h323 devices or the other way round, I get only one-way audio. Called party can only talk, caller can only listen. Calling SIP to SIP is ok. All devices are on official IP addresses. (no NAT) Regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail shaoul@tellink.com

...04 It dials out successfully, but never times out. I have a basic Zapata config : group = 1 context = RedRockWeb language = en signalling = fxs_ks usecallerid = yes hidecallerid = no echocancel = yes echocancelwhenbridged = no immediate = no channel => 1-2 Suggestions? Thanks! Tom tweeks@tellink-corp.com www.tellink-corp.com 303-697-2357 303-697-3103 fax -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031112/e24cebfc/attachment.htm

...oldered it in the case.... I was able to split off 12v and 5v from the ups battery and run for days... Sounds like weird science - but it works!! -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Shaoul Jacobson - TELLINK Sent: Thursday, May 19, 2005 8:26 AM To: Wilson Pickett; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] OT: carrying a router, firewall, switch, server, some phones with me on flight to Europe Hi, > There is of course one obvious issue, that of powering y...

...VoIP only polycom build (video-) conferencing devices. One Cisco model (7930 I thnk) is a polycom in disguise. The code is not 'cisco-like' (at least the version I had. Both brands make very good quality equipments. Good sound, good support, ... Regards, Shaoul Jacobson VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail shaoul@tellink.com -----Original Message----- From: Sergey Kuznetsov [mailto:asterisk_biz@deeptown.org] Sent: jeudi 10 f?vrier 2005 4:15 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Please share the experience on VoIP phones he...

We have one Swissvoice IP10S running SIP firmware. Recently, I've been getting these messages: Mar 1 13:59:44 NOTICE[20933]: frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Any clues off the bat? I'm still researching other stuff.. Thanks, Matthew

...most under /usr/src/asterisk/include/asterisk. Only the 'channel_pvt.h' is missing. I updated via cvs rather recently. I did today a cvs again. but the 'missing file' did not came. And I could not find it in another directory. Regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail shaoul@tellink.com

...008) some 'patch' files are included. I am a newbie to linux and asterisk. I do not want to blow my config. Please give me a feed-back if those files helped you and how. Also if you have a work-around (like an old file to use) Thanks & regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail shaoul@tellink.com -----Original Message----- From: Dmitry Melekhov [mailto:dm@belkam.com] Sent: lundi 14 mars 2005 11:52 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 1.0.5 and h323 compiling problem Hello! Looks like h323 co...

Hi list! Sorry for this slightly off-topic message but does anybody know if the standard for ISDN BRI is the same in Spain as it is in the rest of Europe (or the Netherlands). Will a standard HFC-S card work?

Hello: Has anyone on this list had to configure hairpinning on a Cisco gateway running IOS 12.2 or 12.3 and using a PRI for connectivity to the PSTN? If so could you tell me how it is done? I'm told this is the source of my call transfer problems and yet I cannot find clear instructions for how the configuration is done. Thanks,Steve -- ISC Network Engineering The University of

Dear Fellow *-ers, First, you guys are fantastic. Keep fighting the good fight. Second, it sounds like comments in the code are coming, which sounds welcome by all, even those of us who couldn't code their way out of a papersack, but who need to read the source. Last, I might be traveling to Europe (from US) & want to tow along hardware & haven't done this before & was

I have the following configuration and am obviously missing something small that is causing * not to work as expected. I have the following defined in sip.conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco

Hi..... I am new to this software, and I want to implement a client (SIP or IAX) with PHP or at least to pass the main functions (connection,call, transfer, hangup, call id etc) to a CRM. Does anyone know if I could achive a project like that with AGI ? Any example using AGI with PHP ? Do I have all the functionality with AGI ? What about call id ? What is depend on ? (As I know * does not

Are there implementations of, e.g. Neville's algorithm, for interpolating polynomials through some data points? Nevilles' is an improvement on Lagrange interpolation. And how about interpolating rational functions? I could not find anything at rseek.org or at crantastic.org. thanks -- View this message in context:

I have a simple system of linear equations to solve for X, aX=b: > a [,1] [,2] [,3] [,4] [1,] 1 2 1 1 [2,] 3 0 0 4 [3,] 1 -4 -2 -2 [4,] 0 0 0 0 > b [,1] [1,] 0 [2,] 2 [3,] 2 [4,] 0 (This is ex Ch1, 2.2 of Artin, Algebra). So, 3 eqs in 4 unknowns. One can easily use row-reductions to find a homogeneous solution(b=0) of: X_1