similar to: SIP registrations

On a new * asterisk install onto new install Gentoo 2003.4 upon startup of asterisk: WARNING[16384]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/res_perl.so: undefined symbol: PL_thr_key WARNING[16384]: loader.c:429 load_modules: Loading module res_perl.so failed! perl -v = v5.8.5 built for i386-linux-thread-multi <= I installed ithread support in perl Have not been able to

Using an example provided by "The Hitchhiker's Guide to Asterisk", I made the following addition to my extensions.conf file: [inbound-analog] exten => s,1,Wait(1) exten => s,2,SetVar(counter=0) exten => s,3,Answer() exten => s,4,Wait(1) exten => s,5,DigitTimeout(15) exten => s,6,ResponseTimeout(10)

I know this is probably way out there but............ Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an

I would like to use one of my Snom 200's 5 programmable keys to park calls. I am using image SIP 2.04g. I have tried a variety of combinations and have come to the conclusion that: 1) On the Key Mappings administration page, I must select the "Transfer" under the "Break Keys" option box to be able to successfully transfer calls using the "Transfer" button. 2)

I created an account at IAXTEL.com to route 1-700-XXX-XXXX calls through. IAXTEL.com gave me a number (example) of 700-555-6226. I have made the following changes to my: /etc/asterisk/extensions.conf: [iaxtel700] exten => _81700XXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1}) exten => _81800NXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})

Hello all: For fun, I am learning about Asterisk, and trying to get Asterisk working at my house. I installed Asterisk@Home. It seems to be functioning fine. I installed a couple of softphones, and have them registered with Asterisk. I actually work for a CLEC, and I have registered my Asterisk box with SER (which I don't begin to understand yet) at the office. In order to try to

Hello list, I try to pick up a ringing extension but nothing works. To be clear, I'm trying to pick up extension 10. [Jun 14 17:37:34] -- Executing [**10 at from-TESTCORP:4] Pickup("SIP/testcorp3-00000041", "10 at 123456") in new stack [Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No target channel found for 10. [Jun 14 17:37:34] --

Friends, Please send all offers for help *off list* to us at info@astricon.net. Do not disturb the list with offers of your services, please. I repeat: Only the Developer's Meeting will be considered for broadcast at this time. In order to enjoy the conference, you will simply have to be there. It's an IRL experience - meeting all the other Asterisk user's from around the globe,

Hi folks! I bought a X100P a while ago and know I've tried to get it working here at home again ... but I can't manage to get my X-Lite client working with Asterisk (CVS from a day ago) ... I've downloaded the latest version of X-Lite and I believe that I've set it up correctly ;-) But I cant get it to register with my Asterisk - I only get "Login timed out, contact your

Hi all, I have a small system of two hardware boxes (residential gateways) running Linux with Asterisk on them. Each RG has some FXS ports to which analog telephones can be connected. I already had a working system including an external SIP provider, where both RGs would register to that provider with a telephone number and they could call each other via that telephone number. Each RG had a line

I'm having trouble connecting a asterisk server to a SIP Express router. Inbound calls to my asterisk server works just fine, but when i try to make outbound calls I get the following error message: Nov 27 22:40:48 NOTICE[4687]: chan_sip2.c:7967 handle_response: Failed to WWW-authenticate on INVITE to '"username" <sip:username@mysipprovider>;tag=as5399a078' I'm

Hi, the documentation of sip.conf is telling me this: ;register => 1234:password@mysipprovider.com ; ; This will pass incoming calls to the 's' extension In reality it jumps to the extension 1234 in the context and not to s So it is much more complicate to write an proper dialplan. Is this an bug or is the documentation not up to date? best regards Thomas

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

Hello list, why is it that GoToIfTime thinks a date of **|*|29-*|jun *is not valid ?? [Jun 29 14:06:34] -- Executing [s at macro-vac:10] *GotoIfTime*("SIP/testcorp-00000036", "**|*|29-*|jun*?onvac") in new stack [Jun 29 14:06:34] WARNING[3076]: pbx.c:4127 get_range: Invalid end day '*', assuming none [Jun 29 14:06:34] -- Executing [s at macro-vac:11]

If you update your source from the CVS, you'll get a new SIP channel that supports a new syntax for SIP calls in extensions.conf If you define a SIP peer in sip conf, like [mysipprovider] ... You can now use dial(SIP/mysipprovider/extension) Where the part "mysipprovider" is related to the sip.conf section. Also, you can dial any SIP URL by

Hi, I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). These two boxes talk to eachother via sip, not iax. Since the upgrade, I get the error "Failed to authenticate on INVITE" trying to make calls to/from either box. Removing the secret from each box's sip config seems to work but is utterly braindead. Has anyone seen this? - Eric

Hi All Total noob on the list so all help appreciated.... I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones.... The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......

I installed SUSE LINUX Enterprise Server 9 that uses a Linux 2.6.5 kernel. When I installed the OS, I chose a minimal package install. I compiled the latest asterisk out of CVS. So far things have gone OK. I can establish phone calls (SIP, TDM, etc) and the quality is good. Only problem is that I can get no sound out of Asterisk itself: prompts, comedian mail prompts, etc. I can see messages on

Hi guys, I''m setting up asterisk to run with another pbx server. This pbx server support a feature that allows 2 extensions connect to the same FXS. No I put asterisk in the middle. Asterisk receives the call and dial to a SIP/peer. How the pbx installed support 2 extensions to one fxs... How can I figure out in asterisk which extension was dialed before the call came to asterisk?

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike