similar to: Properly setup SRV?

Per my conversation below with digium, are there any legal alternatives to digium's G729? It is out of date, and doesn't support VAD nor silence detection. Digium has stated that they have no plans to update it anytime soon. VAD/Silence is a big deal with major carriers and we are having to fight a battle to get them to make special arrangements to turn off VAD/Silence in their

Even though they make the cards and advertise that they support data modes, digium won't support data mode on the $500 card they sold to me, so I must turn to the list. Has anyone configured a T100P to use a T1 strictly as bandwidth? Is there a HOWTO somewhere? Wiki has nothing I could find. I've got plently of public IPs I can assign to it but don't know how. Thanks, Matthew -----

Anyone used the combination above? We are and it sounds like crap. The audio drops out in regular intervals which suggests that someone's g729 isn't doing its job correctly. I'm blameing XTen cause when I make a ulaw call that gets converted to 729 using digium's 729, calls sound fine. Anyone else? Similar experiences? -Matthew --

All of the stuff I've googled for and read on wiki all relate to "Outlook". Has anyone been successful in getting "Outlook Express" to do click to dial? -Matthew -- ------------------------------------------------------------------------ Matthew Boehm, IT Director Cypress Telecommunications mboehm@cytelcom.com 3838 N. Sam Houston

I've read the Wiki on using asterisk's built-in transfer options (#8 and #6). They work fine but how does one cancle an attended transfer? Example: I have person on phone, I hit #6 to being att-transfer. I enter Sally's extension. I let it ring for a few seconds. Sally never picks up but her voicemail does. How do I hangup her voicemail and resume the previous call? The example on the

This is frustrating. Asterisk has crashed now twice today and neither crash has produced a core file. My ulimit is unlimited. I'm using safe_asterisk so asterisk is restarting immediatly, but how the hell am I suposed to find out wtf happened with no core file? Debug log doesn't say anything either. AGRHHHHHHHH -Matthew --

Given the following SRV records: _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060 sometimes.sip-happens.com. _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com. Why is asterisk (1.4.17) not honouring the priority and not failing over to using other records when a connection fails? For a given call to tollfree.sip-happens.com ares.sip-happens.com was chosen

I don't know...now I have a _X. in my CDR. -----Original Message----- From: Matthew Boehm [mailto:mboehm@cytelcom.com] Sent: Saturday, March 12, 2005 8:05 PM To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> Subject: Re: [Asterisk-Dev] SetVarCDR You must have some fux0red config 'cause using _X. works fine here. I haven't had an 's' in my CDRs for

Hi, SRV records have been working fine until my hoster decided to upgrade BIND. working wrong syntax: _sip._udp SRV 10 10 5050 mydyndns. correct syntax: _sip._udp IN SRV 10 10 5050 mydyndns. That kicked of one of my domains completly caused by a syntax error that did no harm to the previous version After inserting the missing "IN" the zonefile loaded but now

Why can't people learn to turn off this crap for lists? I don't care if you are on vacation and I'm sure that everyone else on this list doesn't care either. -Matthew ----- Original Message ----- From: "Dijkstra, Roelof" <Roelof.Dijkstra@nl.compuware.com> To: "Matthew Boehm" <mboehm@cytelcom.com> Sent: Thursday, January 06, 2005 9:24 AM Subject:

I suspect if someone recompiled the module that handles the transcoding it could take advantage of say a AMD64 bit cpu. Although I bet cache has a big impact on performance with this. Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -----Original Message----- From: Matthew Boehm [mailto:mboehm@cytelcom.com] Sent: Monday, January 10, 2005 4:20 PM To: Asterisk

I'm trying to determine if this is a zaptel issue or a sangoma issue. When I start our server, it gets to the point where it says "starting wanrouter" (thats the sangoma drivers) and it just hangs there. Nothing happens. Can't even use console. I unplugged the 4 PRI lines and rebooted. This time machine comes up. I logged in. While logged in, I plug the 4 spans back in. I try a

I posted this to the -dev list and have recieved no reponse so I'm gonna try the -users list. I'm trying to bring ast-ax-snmp up to date with current CVS. It's a small patch against chan_zap and pbx, but also has a loadable module component. That module is compiled against some NET-SNMPD headers/libraries. But net-snmpd also has module support and they have a function called

We have asterisk running on a quad processor dell. The kernel has been compiled with SMP. However, asterisk seems to only use 1 processor. 3 of the 4 always stay at 100% idle. Is it pointless to have a multi-proc machine? I was going to buy a new dual 3.6Ghz Xeon server but if nothing will take advantage of the other proc... Perhaps my conception of multi-proc/threaded is warped. If asterisk is

Check with your telco. We had the same problem on 1 of our PRI's, every day at 5:00 sharp, red alarm, with all calls cut off for 30 seconds exactly. Turns out the equipment at the CO was going into a test loop at that time because of a forgotten setting by a tech. Man, what a finger pointing exercise that was. -----Original Message----- From: Matthew Boehm [mailto:mboehm@cytelcom.com] Sent:

Let's say I have a third-party device acting as a sip<-->pstn gateway, a cluster of three asterisk servers, and a teensy bit of dns knowledge. Let's now say those asterisk servers are a1.company.com at 192.168.0.1, a2.company.com at 192.168.0.2, and a3.company.com at 192.168.0.3. 1. If I setup round-robin dns like so: asterisk.company.com. IN A 192.168.0.1 asterisk.company.com. IN

This for the archives in case it may help someone: I was able to upgrade two Cisco 7940's from firmware P0030301MFG2 to SIP 7.1 as follows: 1. Installed 7.1 images from the Cisco zip file to the TFTP server. 2. Specified "image_version: P0S3-07-1-00" in SIP<MAC>.cnf and SIPDefault.cnf 3. For the older of the two phones, renamed P003-07-1-00.bin to P0S3-07-.bin, making it

I have introduced some error in my dns resolution and I would like some help fixing it as I cannot seem to detect what I have done wrong. Briefly the setup is this: name servers: DNS01 - 216.185.71.33 DNS02 - 209.47.176.33 DNS03 - 216.185.71.34 DNS04 - 209.47.176.34 - offline DNS01 is a master DNS02-04 are slaves of 216.185.71.33 All are listed as authoritative for the zone test.com The

i think your email at fisher is stripping the images On Nov 30, 2004, at 1:00 PM, asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject

Can asterisk use SRV record from DNS? In example - I've got: _sip._tcp.gda.pl. SRV 0 0 5060 welbot.task.gda.pl. _sip._udp.gda.pl. SRV 1 0 5060 welbot.task.gda.pl. And it works with sipd (from Columbia) and vocal. I see that with asterisk it does not want to work... :-/ -- pozdr. Pawe? Go?aszewski --------------------------------- worth to see: