similar to: Segmentation fault due to SIP registration N UMBER 2

I assume that from your previous post that you are using iconnect Is your register line in the format: Register => 18005551212:1234@213.137.73.178/18005551212 I've had good luck using the IP address vs. the fully qualified hostname. Remember that the register line goes in the [general] section of sip.conf. Also, are you using the latest CVS release of *? -----Original Message-----

Use oh323. Download the openh323 and pwlib tarballs from openh323.org Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY! good luck Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 sean.langley@gdcanada.com > -----Original Message----- > From: Senad Jordanovic [mailto:senad@cwcom.net] > Sent: Friday, September 12,

Hi all, I'm using Wine for a MotionCapturing Application (Qualisys). Although it's a quite recent application, it works well (I tried only data processing so far), but when I try to save a dataset (~180MB), it takes about 20 Minutes (!!) on my Computer. When I use it under VirtualBox (it does not run nicely there because of OpenGL), saving takes about 20 seconds. When I look at

Hi OutCALL 1.40 is released. It is available in two flavours: - Without extension authentication - With extension authentication Changelog: OutCALL 1.40 (2007-06-29): - Multi-language support (French-Canada is included in the setup, while the English PO file is distributed with OutCALL setup which can be translated and added into OutCALL in run-time) Please use http://www.poedit.net/ for

Hi, Matt: What?s your server specifications that did you use? Best Regards, Cleviton. -----Mensagem original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]Em nome de Matt Florell Enviada em: sexta-feira, 23 de junho de 2006 11:38 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] Meetme max users

Hello Everyone, About a month ago, someone put a question to the list about which ADSI spec to purchase from Telcordia. I looked in the archives, and it appears that this question was never answered, so I'll put it to the list in a slightly different manner: Do I need to purchase the Telcordia specs in order to learn how to write my own ADSI scripts? If so, which one? I found the Black

Hi Matt, Any decent quad, quad-core Opteron system should be able to handle it with ease! :) -- Regards, Hilton Travis Phone: +61 (0)7 3344 3889 (Brisbane, Australia) Phone: +61 (0)419 792 394 Manager, Quark IT http://www.quarkit.com.au Quark AudioVisual http://www.quarkav.net http://www.threatcode.com/

I'm wondering if anyone else has gotten something similer to this? I had FWD working fine on the asterisk box, then all of a sudden it just stopped working. I get the following errors (just keeps looping) *CLI> DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '6dc8436c7c568eea75fffdc75478ed54@142.55.31.179' of Request 102: Found

It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Matt Florell Sent: Monday, March 13, 2006 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hi! I am new to Asterisk too, I got the similar problem and I would like to know how to get * to work behind NAT. When I have the SIP Debug turn on, I got the error 479 from FWD when * try to register with FWD, it looks like * is using the local IP (192.168.x.x) in the Contact field. I have put the nat=yes in the [FWD.Pulver.com] content, but it does not seems to make Asterisk aware the

Make sure you are using G.711a, G.711u or GSM codecs.. I have not been able to get iLBC to work and someone the other days couuld not get SPX working.. You will need to enable/disable the codecs in X-Lite.. If you also want to control the codecs that * uses then put the following in the general section of your sip.conf disallow=all allow=alaw allow=ulaw allow=gsm Hope that helps.. > Hi,

hi, I'm getting registration errors I can't debug... [Mar 23 11:07:20] NOTICE[2952]: chan_iax2.c:7344 socket_process: Registration of 'host2' rejected: 'Registration Refused' from: '10.10.10.82' I was getting a 'Cause Code: 29' INV,POKE,...,REJ but I can't duplicate that level of debugging again in the CLI> on host15 10.10.10.15

Hi, I have enabled ztdummy in order to have * compile it. I can modprobe ztdummy with no problems. The sole reason, i need ztdummy is to heve musiconhold and meetme working. However when I start *, it says this and does not start. ---------------------------------------------------------------------------- ---------------------- == Parsing '/etc/asterisk/zapata.conf': Found

Sherwood, Your intentions are noble and your desire to build this, fullfills an immediate need for business. If your intention is just to build a GUI for Asterisk, read no further. If your desire is to build something more purposeful, your best bet would be to see the existing commercial GUI/HostedPBX offerings like Pbxware and Switchware from bicomsystems.com ( http://www.bicomsystems.com)

Hello all! I've got CGYWIN running on an NT4 server, which in turn allows me to run Rsync as a daemon on this system. And might I say, it works most excellently, except for one flaw. It doesn't work on my raid drive(s). I have a 56 gig NTFS RAID5 drive running off an adaptec raid-port controller card. I have two other drives in this system, a 4 gig SCSI boot drive, and 60 gig IDE

Hi all: I have problems to setup my zaptel E100P hardware. When I start * after receive the "Asterisk Ready" I see this: *CLI> Jun 22 20:37:55 NOTICE[1133718080]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 1 Jun 22 20:37:55 NOTICE[1133718080]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 2 Up to channel 31. anfter this: Jun 22 20:37:55

Bicom Systems releases outCALL, an Asterisk open source Outlook integration LONDON, UK (11th April 2007) - Bicom Systems announced today it has released outCALL, an open source desktop application allowing integration Microsoft Outlook. OutCALL allows users an easy way for placing and receiving phone calls integrated with users Outlook contacts. "The open source PBX market needed

I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced

This is a global message to all to announce our callcenter / dialer / predictive dialer / vicidial program is now open. Codecs: G711, GSM, G729, G723 Protocols: SIP Duration Rate : 30/6 (6/6 with monthly minutes over 100,000) Channels : 100 to start with , more on demand. We are predictive dialer friendly , your account will not be shut off. Contact us to do a test run. Mike

On Fri, 2006-10-13 at 10:50 -0600, Joseph wrote: On Fri, 2006-10-13 at 07:27 +0200, Remco Barendse wrote: > > On Thu, 12 Oct 2006, Eric "ManxPower" Wieling wrote: > > > > > Matt Florell wrote: > > > > If you downgrade, let us know if it fixes things for you. > > > > > > > > It's strange that there were so many changes in the