similar to: Toronto?

Just on the off chance that Canadian Asterisk users might be interested in a place to discuss topics specific to the "great white north" (sources, services, telcos, etc.), I created the asterisk-canada mailing list: http://lists.syonex.com/mailman/listinfo/asterisk-canada or asterisk-canada-subscribe@lists.syonex.com Cheers! John

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 =>

Folks, In light of all the troubles people report when running more than one TDM400 card in a system, I wouldn't mind hearing what your solution of choice would be when having to connect 5 or more analog telco circuits to an Asterisk. I'll try and compile the answers together and get them into the Wiki, as I figure this could be useful knowledge for the community. TIA, Jim. -- Jim

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

Hello, Ever since I started using Asterisk I always get this error: Jan 10 15:39:26 NOTICE[4501]: res_musiconhold.c:463 monmp3thread: Request to schedule in the past?!?! I have a dedicated system system that really runs only Asterisk: - Pentium III 500Mhz - 128MB of RAM - 10GB of Disk Space - SuSE v9.2 - MySQL - Apache (only for use with Asterisk) - NTP client for clock synch There is no X

Hi Kristian, Anywhere I can read about this Soekris/AstLinux project? ... Regards, Hans -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Kristian Kielhofner Sent: Thursday, February 24, 2005 6:02 AM To: jim@vanmeggelen.ca; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

Folks, I'm curious to know how the volume of Asterisk-Users rates as far mailing lists go. This list sees over 200 messages per day, which has GOT to put it in the top 5%, doesn't it? I'd love to know if anyone has knowledge of any organization that might maintain such stats. Regards, Jim. -- Jim Van Meggelen jim@vanmeggelen.ca -- No virus found in this outgoing message.

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

I would like to ask what people think the best way would be to build a low-power consumption, passively-cooled system. For example,one could use a fanless-Eden (mini-ITX/EPIA) system, but the loss of FPU power would limit performance. The obvious choice for FPU work are the Intels and AMDs, nut they're all power-hungry radiators. Is there something that can offer the quiet, power-savings of

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

I would like to start a discussion centred around the various ways one might serve up configuration files from an Asterisk server (I know, it's better to use a secondary server for all this, but let's talk about a smaller system). The types of things being served would include: - Logo image for sets that support that - XML directory files - XML or raw text configuration files -

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%

HELP! Ok, so I have the following SIP.CONF: [general] context=default port=5060 bindaddr=10.1.1.200 externip = XX.XXX.XX.XX localnet=10.0.0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register => ##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com

The thing I dislike the most about the 79xx phones is that in DHCP mode, they expect the DHCP server to tell them their TFTP server address. They won't let you set it manually. So if I don't have DHCP server that gives TFTP server info, which is most of the DHCP servers at out there, then the phone won't be able to download any updates made to the SIP000*.cnf file. Using dhcpd on

Hello. I am going to be putting together my first * system using FXO/FXS interfaces. All the systems I have set up thus far have been pure VoIP setups. The system I need to set up should have 3 FXO interfaces and 1 FXS interface, as well as several SIP phones. I have noticed people complaining about Digium's TDM cards - are these isolated incidents or are these cards unreliable? I intend to

We have a strange issue here - we have the following setup: Asterisk CVS-HEAD-12/15/04-07:42:16 40 SIP Cisco 7940 phones, linking to PSTN via EuroiSDN 30 channels. Often, when someone tries to dial any internal extension or external number, they get the "Reorder" message. If they try again, they get another "Reorder" message. If they try a third time, the call gets

I believe you can assign processors in vmware, and xen as well. So you could probably do something funky like that to try to reduce load. The only thing that probably becomes difficult is trying to manage physical hardware between virtual machines. John -----Original Message----- From: Jim Van Meggelen [mailto:jim.vanmeggelen@coretel.ca] Sent: Thursday, March 02, 2006 7:47 PM To:

> 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 folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'