similar to: [Fwd: Re: BroadVoice configuration changes for Outbound]

Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=<phonenumber> authuser=<phonenumber> secret=<registration password> Dan

Hi, Is anyone using mozPhone? If so any feedback you can provide? Thanks, Glenn

Are these attempts to scam SIP calls through my Asterisk server: [May 13 22:50:41] NOTICE[30888]: chan_sip.c:17295 handle_request_invite: Call from '' to extension '084312297134' rejected because extension not found. [May 14 13:36:35] NOTICE[30888]: chan_sip.c:17295 handle_request_invite: Call from '' to extension '0114312297136' rejected because extension not

Went from 1.6.0.6 to 1.6.0.8 and resulted in segmentation fault. Reverted to 1.6.0.6 and back to normal. ------------------ Linux asterisk.hulber.com 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24 EST 2009 x86_64 x86_64 x86_64 GNU/Linux Apr 3 11:49:56 asterisk kernel: asterisk[3780]: segfault at 00002ce1ac0537a8 rip 0000003e980715a8 rsp 00007fff5bf00c30 error 4 Apr 3 11:50:00 asterisk

Is the Broadvoice service up? I just signed up with them and started receiving calls in no time but could not make calls. And after a few minutes I cannot even place calls. register => [number]:[password]@sip.broadvoice.com [broadvoice] type=peer fromuser=[number] host=proxy.lax.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice dtmfmode=inband any help would be

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

Thanks Mark will try that out! -----Original Message----- From: MF Hulber [mailto:asterisk-admin@hulber.com] Sent: 22 March 2005 05:25 To: Reuben Grech Subject: [info] [Asterisk-Users] :: BIOS Motherboard Settings :: I have the same motherboard. I put the card in the 2nd slot from the bottom. In this slot, if you look at the manual, it will possibly be in conflict with some USB channels. I

I want to buy a new server to run Asterisk and after looking at prices for the Athlon XP 3000+ it costs the same as an Athlon 64 at the same speed rating. I was wondering if Zaptel/Asterisk will compile/work on an Athlon 64? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V.

I started to have a problem today that all my calls through voipjet result in just timing out after my assigned timeout period. I tried multiple of their servers with the same problem. Anyone else having a problem? I am running: Asterisk SVN-branch-1.2-r24381M built by root @ asterisk.hulber.com on a i686 running Linux on 2006-05-03 14:14:07 UTC I can connect with other IAX providers.

Here's a link to it: http://forums.digium.com/viewtopic.php?t=4363&highlight= Regards, Scott -----Original Message----- From: Scott Keagy Sent: Monday, December 04, 2006 5:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. A while back I posted a fully functional though somewhat elaborate

ast_waitfordigit that accepts milliseconds as input. Douglas Garstang wrote: > Admittedly I have not used the ExternalIVR app. Is it any good? > > I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, > it can do it, but boy it is UGLY. There's also the fact that you can't > call Backgound() in a macro, which forces you to use Read() which >

Van, It's a new version and there is no inventory in stores yet I know you'll be pleased once it finally arrives. Best, William -----Original Message----- From: Zanzamar Majere <Phoneman@wbtllc.com> Date: Thu, 10 Mar 2005 11:05:07 To:Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: Re: ***SOLVED*** [Asterisk-Users]

Hi guys, I'm trying to write hangup causes from asterisk into the CDR record. Using version 1.4.24.1 at the moment, but no joy so far. Has anyone implemented this? Neeraj Chand Support Analyst Fiji Islands Australia T: +6793342526 T: +61388924326 M:+6799344012 New Zealand www.ocis.com.au T: +649 980 7022 -----Original Message----- From: asterisk-users-bounces at

Hello, My question is about which Linux distribution to choose for Asterisk. (/me holds breath) OK, hopefully you're still reading, because whatever you were thinking now, you're thinking wrong! ;) First of all, I want to make clear that I have read EVERY message and reply that I could possibly find about this topic, so that includes the dozens of messages here on the Asterisk

Hallo Did you allow udp outgoing on 4569 as well.. i found udp bit different than tcp when comming to firewalls liaan ----- Original Message ----- From: "Bartosz Wegrzyn - asterisk" <junk@lexon.ws> To: <timebandit001@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, February 21, 2005 12:29

Currently, I have attempted to get musiconhold going up through a custom handler. I have guaranteed it works all the way up to asterisk, and when asterisk is running I see the music on hold process. The current issue is that when asterisk gets a sip invite and starts the session, I hear nothing. /usr/bin/ecasound -s:/home/dweber/xm001 -C -q is sending raw output to stdout Dan