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Is anyone else experiencing trouble with Teliax? I can only intermittently register to, and am not able to place any outgoing calls through my assigned gateway; voip-co3.teliax.com. -Rusty -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060120/aa8ab017/attachment.htm

Hi, I am trying to get Asterisk 1.2.6 to run on Solaris 8 (sparc). I was able to get it to compile, but when I try to start asterisk (./asterisk -cvg from /opt/asterisk/usr/sbin), I get the following error: (snip) Asterisk Dynamic Loader Starting: [res_musiconhold.so]Mar 28 09:23:48 WARNING[18299]: loader.c:325 __load_resource: ld.so.1: ./asterisk: fatal: relocation error: file

...st. Its taken off of the Asterisk.org site. Where is this used in Asterisk? Is it in the extensions.conf, is it used when logging into the console, where? How is it used? Is this referring top contexts in the dial plan? Message: 1 >Date: Thu, 13 Apr 2006 23:30:24 -0400 >From: "Rusty Dekema" <rdekema@gmail.com> >Subject: Re: [Asterisk-Users] What is Multi-layerAccess control >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> >Message-ID: > <68171c120604132030x177dad62o92b8d671850bdaa4@mail.gma...

...tion show up as serial/tty type devices (or some such) so that I can run regular Linux pppd on both of them? I see that there is a plugin for pppd in dahdi-tools, but I don't know how to configure the spans in system.conf, nor what device to point pppd to, in order to do this. Sincerely, Rusty Dekema

...com/DrDeke/9436874 Running dahdi_tool shows only one span; "DAHDI_DUMMY/1 (source: HRtimer)", in the "UNCONFIGURED" state. If anyone has any ideas for me to try, or would like to see any further debugging information, I would be very grateful if you let me know. Thanks, Rusty Dekema -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140308/097444fc/attachment.html>

I read the thread about what IP phone is best for business deployment with great interest. Our need is slightly different however. We are deploying VoiP as a value-add with our high speed internet service and are having trouble finding the right SIP analog terminal adapter. In order to support people's existing phones and wiring we need to use an ATA. 1) The first priority is we want

Hi, I have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but that in turns means my outgoing calls don't work. - Strange Anyhow I was getting an error: Process_sdp: No compatible codecs! And from the SIP

I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- Steven

Asterisk 1.2.3 has been released and contains a number of bug fixes. One of the fixes is for a critical bug introduced in version 1.2.2 that will cause an Asterisk server to stop processing calls correctly when the server's clock reaches January 25th, 2006 (today). It is vital to upgrade all 1.2.2 servers with this release as soon as possible. Thank you for your support for Asterisk!

Our users are experiencing some unacceptable delay when trying to have a conversation. The delay is so noticeable that they keep stepping on each others words and resort to calling the customers via cell phone. Here is the setup SDSL Connection (PPPoA) Speedtouch 610 SDSL Modem 3Com 2224PWR Plus Switch (phones on separate VLAN) 8 Cisco 796 Phones All connecting to a remote Asterisk Server. We