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...formation on the list that I can use a 100D T1 card to bridge the PBX to an asterisk system. We are looking to head this route as well as replacing are AUDIX VM system with Asterisk. Has anyone done this that wouldn't mind sharing some of the merlin specific setup? Regards, Sean Cook scook@kinex.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050804/6ac4fa17/attachment.htm

I am having a problem with the Vmail Web app and was hoping someone has some advice. I have been able to install vmail.cgi. I am able to bring up the vmail web page and login with out problems. However, I never see any vmails listed even when there are vmails present in the directory [root@linux-home INBOX]# ls -l total 40 -rwxrwxrwx 1 root root 255 Dec 2 14:45 msg0000.txt

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I am stuck in a situation where I have two te110p's and really need a te210P. Anyone interested in a trade? They are in working condition guaranteed. If this is the wrong forum for this, I am sorry... I will take it off list. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird -

What and When is the next version of Asterisk? /Obelix

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Anyone have a 1650 running successfully in production mode with 2-4 PRI's? I want to make sure I don't have a motherboard compatibility problem before I buy one of these. We are going to be using a Digium TE210P to start off with and probably moving to the TE411P down the road aways? thanks, Sean -----BEGIN PGP SIGNATURE----- Version:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Ok... I am having a serious brain fart this evening. IIRC, the next sip draft addresses shared lines and I thought I remembered something on the list about support for it in the near future. I also thought that chan_sccp supported it with in asterisk. Am I loosing it here? Is there an implementation for shared line support in asterisk? I know that

OK... maybe I got a little anxious and ran out and bought a Tyan GX28 with dual Opteron (dual core) processors. (It is a nice server ;) ) I did neglect to find out that you can not manually set the IRQ's on this motherboard. I am now stuck sharing an IRQ with the ethernet controller and no foreseeable end to my dilemma. I have a Digium TE210P and zttest consistently runs at 99.97% which

Does anyone know of any obscur reference for detecting an incoming fax. I currently have AMP running and everything else is working great. Installed the spandsp patches and software... using the default AMP extensions.conf, I start sending a fax, I hear it pick up and transfer to voicemail after 20s. Fax is set for system... Here is the detail from the extensions.conf [global] FAX_RX = system

[New Features] 1. Added focus and tab-order to all input fields 2. Dynamic generation of date/month/year listboxes a. It is no longer possible to schedule an invalid date. 3. Added 'Extend' and 'End Now' buttons to the monitor page. 4. Invite button on the monitor page. This greatly simplifies the process of adding callers to a conference. The ./lib/defines

Ok... I am reluctant to ask this question as I believe that it may be like asking what someones favorite linux distribution is... but I need to make an informed decision. We are getting ready to upgrade from a TE210P to a quad T1 card with echo cancellation. I am trying to decide between the Sangoma card and the Digium card. I need this to have great quality and I need it to work well. I would

...Regards, Girish > > > > __________________________________ > Do you Yahoo!? > The all-new My Yahoo! - What will yours do? > http://my.yahoo.com > > > ------------------------------ > > Message: 3 > Date: Wed, 01 Dec 2004 09:33:54 -0500 > From: Sean Cook <scook@kinex.net> > Subject: Re: [Asterisk-Users] broadvoice and gsm codec > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <1101911634.18099.2.camel@sean.benchmark-systems.com> > Content-Type: text/plain > &gt...

Anyone know if #include works in ael yet? extensions.ael: #include "inc/pbx/global.conf" context test_context { }; *CLI> ael reload May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 handle_root_token: Unknown root token '#include' May 30 13:56:45 WARNING[8516]: pbx.c:3758 ast_merge_contexts_and_delete: Requested contexts didn't get merged

I set up hints and presence monitoring on some Polycom phones connected to an asterisk server with the expectation that the phones that are "watching" other extensions would be notified when the other extension sis ringing, in addition to the other statuses (on the phone, statuses set by the user on the phone, not registered, etc). I can see when the line is in use, and when it is

Hi, I have already successfully created a loadable module to perform a transformation pass that is used by LLVM's opt program. However, now I would like to incorporate a specialized parser (built with flex and bison) within that very same loadable module. This parser's function is to parse a simple language that will be used to specify how and where to perform the transformations. What

Hi all, I am an LLVM newbie and I am having trouble understanding LLVM's makefile system for building projects. I would like to create a customized loadable module to be used as an optimization pass with LLVM's "opt" program. My problem is that I would like to incorporate a parser (built with flex and bison) within the same loadable module that performs the

I know that this phone will not do sip as it has limited memory, however has anyone been successful getting it to work with mgcp/h323? I mistakenly bought one of these thinking it was a 500 (description noted the 500 documentation). Anyway... just hoping I didn't blow $150 and that I can at least get it to work sean

I am not sure if this is really possible but I figured I would ask anyway. I have a customer who wants an asterisk system. Currently they have a BizFon system. The feature that he really wants is to be able to pick up any line and have all the stations show up on his phone. Is this possible in asterisk? If so can someone point me in the right direction? Sean

I have been hearing a lot about the new Gnet SIP phones. Is anyone using them? How do they perform? Sean

Is the Cisco 7960 capable of monitoring other extensions (hint status) with a SIP implementation? Seems like it could, just can't find any info on it... Sean

I am running unixODBC to connect to postgres for your realtime data for things like call forwarding, dnd and have noticed a significant delay when running the realtime application. Has anyone else encountered this? Even from the CLI if I do "realtime load cf_data exten 4501" it lags for almost 2 full seconds before I get the result set back. Queries directly to the database from