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I tried to connect my * to IAXtel, but i always get this errors. chan_iax2.c:5849 socket_read: Registration of 'mnetwork' rejected: Registration Refused On dial a iax number i get: chan_iax2.c:5526 socket_read: Call rejected by 69.73.19.178: No authority found chan_iax2.c:5528 socket_read: Immediately destroying 3, having received reject chan_iax2.c:2411 iax2_hangup: We're hanging

Hello, Great job on the Asterisk@home project. Looks great this new version is really nicer looking. But I have a few questions. 1) For the new web access http://localIP/maint how and where do I change the password. 2) Since I don't use the Amp section for setup the .conf files I use my own. How do I get the asterisk server running status up. I have it running and works but shows up as not

It should be a mpg123 problem, not a spandsp problem. Stop asterisk, make clean, make install and start asterisk again. Have fun. "Ariel Batista" <arielb27@hotmail.com> wrote in message news:<BAY22-DAV14862521E60E81DB568FDCDBB60@phx.gbl>... I have compiled Spandsp without any problems. I got no errors I have also done the patch without getting any error. I have tried pre4 and pre6 version with same problem. I got no errors with the ./configur...

I feel this is in error some place. If I call a sip device that is not registered or not connected at the time. Asterisk will send that call to voicemail to busy not unavailable. Is there a way to correct this? Ariel Batista Kasi International - Computer Networking Ph: 305-574-6721 Fx: 305-574-0212 -------------- next part -------------- An HTML attachment was scrubbed... URL:

...kernel, I would like to settle on an OS for my customers and don't want to have to readdress the situation in one year because the 2.4 kernel is no longer the supported /stable version. Does anybody believe this likely to happen?? Thanks -----Original Message----- From: Ariel Batista [mailto:arielb27@hotmail.com] Sent: 15 March 2005 15:00 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Kernel 2.4 or 2.6 for the latest asterisk ?? This question really has no one reply. The different Linux builds all have there reasons. If your used to Fedor...

.../listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > ------------------------------ Message: 3 Date: Thu, 19 May 2005 11:44:39 -0400 From: "Ariel Batista" <arielb27@hotmail.com> Subject: RE: [Asterisk-Users] SIP Phone Recommendations? To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <BAY104-DAV156696C5A973C8D2F9623CDB080@phx.gbl> Content-Type: text/plain; charset=&qu...

Hello I get this warning all the time when I am using iax2 for inbound calls or outbound. Aug 24 13:48:41 WARNING[-1105474640]: chan_iax2.c:4873 socket_read: Error: Resource temporarily unavailable I get the calls and the sound is fine. But the screen on the cli is full of these warnings and Error: What can I do to fix this. I get it when using calls to iaxtel, FWD, VoicePulse, Nufone and

I have compiled Spandsp without any problems. I got no errors I have also done the patch without getting any error. I have tried pre4 and pre6 version with same problem. I got no errors with the ./configure make make install. After I add the directory for /var/local/lib into the /etc/ld.so.conf I have ran ldconfig. I am running asterisk as user asterisk on a RH 9 Linux distro. This system has

I just have to make my view known about this. 1) I agree that one is needed but! 2) I feel that there should be a way to get a self study course which will lead to a way to take a test for the Certification. 3) Cost and who set this up is really something that I think should be done first from digium and not out side sources. 4) Which Linux Distro will this be based on is important to make this

Great setup for the Asterisk@home .06. I have a few questions about the console mode. If you go to the Ctrl Alt F9 area you see asterisk loaded but it's displaying some funny Greek letters. I did the following but it did not help. Eliminating some internationalization errors: In /etc/sysconfig/i18n, the first line reads something like LANG="en_US.UTF-8". Change it to read

I have looked at the wiki and the mailing list. But I need to find how do we setup the external IP address and the rtp ports for the Polycom IP-500 and IP-600. There web interface has a nat setting but can't find instructions on how to set this up. I would like to set this up via there ftp file setup instead of via there web setting. Also There QoS settings are set to 5 and 2 but there it

Well I just need to say I got my phone last week. Here is my quick review of the phone and hope that someone has a possible fix for it or I will be sending it back. First the phone is nice looking in my view and it's heavy so it feels like a real desk phone. But it has these stick, gummy or I really don't know how to describe the bottoms on the phone. There good size but when you press

I have a problem which I need to resolve. We are trying to put an asterisk between a Mitel PBX and the world. We are adding Voip service via Asterisk. Here is are config files for the settings but our problem is the following. We are able to send calls to the Mitel pbx and it's the T1 connections is green saying it's ok. The support department from Mitel said that they use e&M and

I have Spandsp working fine. Asterisk sees a fax on the zap port and redirects the call to the fax-in area. This works if I have a simple dialing rules that goes answers first and waits 10 secs then goes to the next item. If it hears a fax it goes to the right place. Here is a sample that works. [incoming] exten => 2019,1,Goto(test,s,1) [test] exten => s,1,answer exten => s,2,wait(5)