similar to: Proper Syntax

Hi all, I bought a couple phones for really cheap just for a simple solution. I'm trying to get a few 7910 to work with *. I'm just not sure how to get them to work. The 7910 just sits there "configuring IP" Here is a copy of my skinny.conf. the extensions.conf is default. I just want to bring the system up in default before a start making changes. Do I need to make

Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really great solution for remote users... even supports QoS. Too bad it doesn't also have VPN functionality built in. Here's a link to the product: http://www.linksys.com/products/product.asp?prid=652&scid=29 -Ron -------------- next part -------------- An HTML attachment was scrubbed... URL:

qualify= and mailbox= do not work with the realtime configuration engine. It doesn't matter if you specify them in the database, the thread that handles them will never look at the peers you have defined in the database, only the ones defined in iax.conf. --------------------------- Thank you. Will this be a permanent situation, or be resolved in future releases? ===== Jason Goecke

I have just installed * (RH9, P4 3.0GHZ, 1G RAM) in a small office, using three TDM400's with 4 FXO's each for incoming calls. Outgoing calls are (for the moment) routed via VoicePulse. Phone sets are Cisco 7940G's using SIP. I'm getting intermittent echo on outgoing calls, and my understanding, based on reviewing the wiki and several posts here, is this: >>>> The

I'm preparing to roll out Asterisk for the voicemail portion of my VOIP network. This week I downloaded a fresh version from CVS of Asterisk and installed the following MySQL 4.1.7 RPMs directly from Mysql.org For some reason after I enable MySQL for CDR and Voicemail in the cdr_mysql.conf and voicemail.conf I don't get any MySQL functionality at all. It almost seems as though MySQL

Good Day List, I have my asterisk box setup to be an ntp server, and my zultys 4X4 phone is able to get the time, however I must first select the TimeZone Offset and then it will use the time setting from my server. This is a hassle because every time the phone reboots the user must answer this question and as you can imagine End users do not know what to do and since the phone is not

Hi All, I am investigating the deployment of VoIP/* in Eastern European areas where there is no PSTN infrastructure. As you can understand DSL/Cable connections are a dream. The only option is satellite. Does anyone know of any satellite providers that have low enough/acceptable delays for VoIP? Thanks, Yiannis.

I'm having trouble with the Realtime setup. I've followed the instructions on voip-info using odbc but I get this message during asterisk boot: Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session

Before I start, I just want to say this is not necessarily a problem with ASTCC, but may be a problem the way I have set up ASTCC (and possibly the way others have set it up as well). The issue is that ASTCC tries to match the pattern *anywhere* in the called number, not necessarily only at the beginning. I have set up ASTCC Routes like this: 1800 Tollfree Trunk1 0 0 100 1416 Canada Trunk2 0 0

Hello, I am just curious how many people are hooked up with BroadVoice and have recently been experiencing a lot of dificulty. Joel

I've setup the voicemail that auths against the mysql db. Now, everything works ok, except voicemail() calls fail with Sep 26 18:09:34 WARNING[157070336]: app_voicemail.c:1517 leave_voicemail: No entry in voicemail config file for '' all my users are in 'sip' voicemail context, but adding context to it: voicemail(@sip) doesn't help.. while if I put a vmbox # to it, it

I had a hard time coming up with a subject. I have some SP IP500s that I'm using with *. When I access voicemail or get the busy announcement, I'm only hearing the last half of the first word or coming in on the second word. As example, when accessing voicemail, I hear ' have 1 new message.' It's like * is starting to play the announcement before opening the channel. This is

I've narrowed this down to a single kernel (kernel.ll), which does a fixed-size matrix-matrix multiply: # ~/llvm-32-final/bin/llc kernel.ll -o kernel32.s # ~/llvm-33-final/bin/llc kernel.ll -o kernel33.s # ~/llvm-32-final/bin/clang++ harness.cpp kernel32.s -o harness32 # ~/llvm-32-final/bin/clang++ harness.cpp kernel33.s -o harness33 # time ./harness32 real 0m0.584s user 0m0.581s sys 0m0.001s

Has anyone had successfully installed more then two digium wildcards in the same machine? I'm going for four. thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

Does asterisk support the intel 537/md3200 chipset? I don't want to start any flames here, I know all about using generic crap in asterisk,[*] which I really don't approve of other than for testing, but I have a customer demanding a generic chipset for his one backup analog line. He will not spend the money for a Digium card and says he will find another company if I can not provide a

I'm seeing a difference in how LLVM 3.3 and 3.2 emit unaligned vector loads on AVX. 3.3 is splitting up an unaligned vector load but in 3.2, it was emitted as a single instruction (details below). In a matrix-matrix inner-kernel, I see a ~25% decrease in performance, which seems to be due to this. Any ideas why this changed? Thanks! Zach LLVM Code: define <4 x double> @vstore(<4 x

Hi, Yes. On Sandybridge 256-bit loads/stores are double pumped. This means that they go in one after the other in two cycles. On Haswell the memory ports are wide enough to allow a 256bit memory operation in one cycle. So, on Sandybridge we split unaligned memory operations into two 128bit parts to allow them to execute in two separate ports. This is also what GCC and ICC do. It is very

Check my reply to your last post. Use SetGroup and Checkgroup before sending the call to your agents. Robert Jackson -----Original Message----- From: Henry Devito [mailto:hdevito@qwest.net] Sent: Thursday, September 30, 2004 10:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Queue Setup almost got it Ok I think I have the queue

Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk -> TE110P ->Kentrox (csu/dsu) -> Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami e&m=1-23 dchan=24

Hi everyone, I'm a newbie to Asterisk. Will Asterisk run on RH9, easily or does it have to run on FreeBSD? Will the drivers for the Digium cards work on RH9? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040919/7d981e5e/attachment.htm