similar to: Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem

Hi All, I'm having an issue where Asterisk continuously sends out a GAZILLION "SIP NOTIFY" messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf resolves this and everything works fine. I'd like to confirm the layout of the

Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register => 908366554:396444 at carrier.jazzey.com register => 908366554:396444 at sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444

Hello, I'm new to asterisk and this list. The ISO download appears to have 1.6 with the FreePBX GUI but I am looking to use the Asterisk GUI. The only option for the Asterisk GUI is to use 1.4. Is it as simple as installing 1.6 only then using the yum repository to install the Asterisk GUI? If so, what packages are needed? Thanks!

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > brett-asterisk@worldcall.net > Sent: Tuesday, November 30, 2004 2:53 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] National (US) callerid name > resolution for yourasterisk box >

Hi everyone, This is my first post to the list, although I am a long term user of Asterisk. I have recently found a problem that I just can't seem to solve. I have a client that has an Ubuntu x64 based Asterisk server with and ISDN Dahdi interface and about 25 SIP handsets. Everything was working fine in Asterisk 1.4 and now after migrating the config to Asterisk 1.6.2.5 I have one single

I have a rather simple setup running under Asterisk 1.4. I'd like to move it to a new install of 1.6. Before I bring it online are there any gotchas I should look for? A Gotcha README would be better but searching with Google and the forums, for me, gets hits that deal with hardware issues -- cards etc. Nothing about depreciated/changed commands. TIA, Rod --

Dear All, Today I upgraded asterisk 1.6 to 1.8. As the result of this when peers trying to register to asterisk the system shows: NOTICE[24698]: chan_sip.c:23417 handle_request_register: Registration from '"50" <sip:50 at 192.168.1.109> <sip:50 at 192.168.1.109>' failed for ' 192.168.1.80:5062' - Wrong password even though on 1.6 everything was OK here is

Hi i have a Te205P connected to a PRI E1, can i force the outgoing callerid to change for each context? for example: [outgoing_context_one] ;force callerid to 12345 exten => _XXXXXXXXXXX,1,Dial(Zap/${EXTEN}) [outgoing_context_two] ;force callerid to 22222 exten => _XXXXXXXXXXX,1,Dial(Zap/${EXTEN}) Can i do that? thanks to all -- /*************/ nik600

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Hi All, I've been in contact with a CNAM (caller name) directory provider. Currently they offer 2 products. One is CNAM via SS7 and the other is Directory Assistance data via http/xml. I am very interested in getting the CNAM data via http/xml (or DNS TXT). I suggested this to the sales rep and he told me that I'm the second person in less than a month who has asked for this. What

Im trying to figure out the base way to check the callerID being sent to my Asterisk box and use it if it is a valid NANP number, but replace it with a static NANP number if it is not. (Why? I have a few carriers that require this, and a few international users - if it happens to take one of the carriers that require it, I want it to set a static number that is valid). I'm playing

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

I downloaded the latest 1.8 (27922) but got some segmentation faults. The first one was when it loaded cdr_odb, and so I changed menuselect not to compile that one, but the second one was when it tried to load chan_agent and so I stopped there to see if anyone else was seeing this. The agents.conf is all commented out except for [general] . Anyone know what is happening? Thanks. P.S. I deleted

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body text="#000000" bgcolor="#ffffff"> <font size="+1">Does anyone have links to the most recent audiocodes

How can I let asterisk immediately dials a trunk when off hook?

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm

Hi, Is it possible to record say 30 seconds of audio and then have LumenVox convert to text ? or any available tool open source for speech to text . Regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100914/b56c3d9c/attachment.htm

Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing "sip set debug peer PROVIDER": Sending to 123.123.123.123 : 5060 (no NAT) ^^^^ That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks