similar to: Request to schedule in the past?!?!

Hello. I am having an issue where sometimes the cheapest provider for certain international destinations is not always reliable in completing calls. However, there is not problem once the call is made (i.e. no lag or echo or anything). The way I have it set up right now (for example) for Dar es Salaam, Tanzania is: exten => _925522XX.,1,Dial(IAX2/livevoip/011${EXTEN:1}) exten =>

> Many thanks Julian. Are you looking for the pinout for a single plug 2.5mm (cellphone) headset or a dual plug 3.5mm (computer) headset? -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeel<at>jafferali.net

This is somewhat related to my other query on the list regarding NAT traversal. I have heard many times that IAX is "NAT-transperant". I am unsure how it accomplishes this. I do know that SIP works like this: your SIP device send a request to the SIP server (usually on port 5060) with whatever command. The SIP server respends to your device's "apparent" IP and port (this

Hello. I picked up a couple of IAXy's for testing. Unfortunately, I read the negative comments only after I bought 'em :( Regardless, I provisioned one unit using my local Linux computer. Now, I'm trying to set it up to provision using the remote * server whenever it tries to register, but it seems I need to know the "service identifier" for the specific device. I can't

I just downloaded the new astcc and it includes now a new field in the list of the cards: Brand Great! How can I use it in the dialplan? bye Ronald

I am trying to get a SPA-3000 to work behind NAT - for the sake of the exercice. The SPA is on the local network at the address 192.168.0.125 behind a NATted linux router. The machine I am trying to work with is a friend's (let's call it lolo.dyndns.org) and I've installed Asterisk 1.0.3 on it. I can see the SPA register but when I try to make an outbound call I get the message:

Hi all I was looking for the SIP IOS of the Cisco IP Phone but i can´t find it in the cisco web page. I need to now the name os de file or a specific category link where i can download it. If you can send me the file is beter ;-) Thanks in advance Regards Wert --------------------------------- Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. Learn more.

Hello. I am looking for a simple Asterisk billing solution. I expect about 50-100 users (a mix of IAX and SIP) through 3-5 outgoing providers (all IAX). I need something that can handle monthly fees and per call charges (depending on destination, obviously), and should provide a web interface for customers and administrators. Something that can tie in to one of the existing management GUIs

I want to block following types of numbers in my extensions.conf like the premium number in Taiwan: exten => _90204X.,1,Congestion Since I have a DID in USA, I need to block these numbers in USA, as well all emergency numbers, but still let open free (???) service numbers. Can you help me to compile such a list? bye Ronald

Anyone in the Toronto area interested in getting together to share notes and swap war stories? -- Jim Van Meggelen jim@vanmeggelen.ca -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 06/01/2005

Hello All, I am looking for a sip phone that is capable of automatic nat. The Cisco ata186 for example works fine for natting with iconnecthere, but as for asterisk, both my 7960 and polycom ip600 require you to set the nat ip on the tftp. Does anyone know a good phone (or ata) that can do this automatically? For example, I want to give a phone to my brother, who is going to europe. His ICH

Before I start, I just want to say this is not necessarily a problem with ASTCC, but may be a problem the way I have set up ASTCC (and possibly the way others have set it up as well). The issue is that ASTCC tries to match the pattern *anywhere* in the called number, not necessarily only at the beginning. I have set up ASTCC Routes like this: 1800 Tollfree Trunk1 0 0 100 1416 Canada Trunk2 0 0

Hello. I have set up ASTCC properly, calling it like this: DeadAGI(${ACCOUNTCODE},${EXTEN}) It seems to be working correctly, but I have two questions: - Although the cards' credit seems to be maintained correctly, I cannot see the call details in astcc-admin. When I try to view information on the card, it's just blank. Any ideas? - When does the 2nd, 3rd and 4th trunk get used? I have

Would anyone mind emailing me, or maybe posting somewhere their music on hold .so file? thx -ben

Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Thanks, Ben Blakely -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/a7e575cc/attachment.htm

I'm putting together an Asterisk server for a local charity to use as an announcement system. I've been thinking about how to write the dialplan to allow different options for different groups' announcements, as well as mailboxes for the various groups and the charity's administrators. Of course, this would also need to include an option for the heads of the different groups to

Just upgraded to the current stable ver. when I start asterisk with -vvvvvcg I get the following error [pbx_loopback.so]Dec 23 19:25:33 WARNING[1633]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined symbol: pbx_substitute_variables_varshead Dec 23 19:25:33 WARNING[1633]: loader.c:440 load_modules: Loading module pbx_loopback.so failed! Asterisk

Sure would like to hear experiences using various FXO to VoIP gateways with *. It seems that any thread that has anything to do with problematic FXO interfaces goes on forever with speculation about everything under the sun. Unless there is someone out there with the engineering experience to build a better one it is a waste of time, let Digium deal with it. If the TDM400P can ever be made 99.99%

> http://www.mixdown.ca/~andrew/dump/threaded_email.png is what > a mailing list looks like to most people, and you can see why > replying to a message, erasing its contents and starting an > entirely new email about a different topic is frowned upon > (yours is the highlighted message). I know this is OT, but can you recommend an email program for Windows that does something like

I'm using P0S3-08-2-00.. I noticed the callerID started showing up with the number, then @<proxy-addr>... So the callerID on the phone looks like: 2145551212@10.10.10.10 which of course is logged in the missed calls exactly like that, and completely foobars the dialing string if you try to dial a missed call by simply hitting the dial button. Can anyone else verify this problem? 7.5