similar to: SIP IOS for cisco 7902G IP Phone

> 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

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 =>

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:

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've set up a dial plan so that outside callers hear a "Welcome" message which asks them to enter an extension or press * to dial by name. This works great. I also want to allow a remote employee to interrupt the message by pressing #, which will direct them to voicemail. The issue I am having is that the READ command uses # as a termination symbol. Is there any other way

I have found out that the reason why my call transfers are not working when using the IAX protocol is because Asterisk is performing a native bridge. If I force the user of one of the clients to use a different codec so that Asterisk is unable to do a native transfer then it works. How can I disable native bridge for IAX calls? I know for SIP you can put 'canreinvite=no' but this does

Hi Guys, After days of fiddling, I can't really get my SIP device to work communicate with Asterisk behind NAT. Sometimes the STUN server is flaky, sometimes the device isn't reachable if the connection is dropped and then put back on, sometimes it registers OK, sometimes it doesn't, etc. I've come to the same conclusion as the wiki: it's probably better to avoid this

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

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 am just curious how many people are hooked up with BroadVoice and have recently been experiencing a lot of dificulty. Joel

I finally got my incoming and outgoing to work on Broadvoice with a configuration file that is no where close to the one given by them. Here it Is (sip.conf). For others who have a working config could u please share so that I could compare. Thank You [broadvoice] type=friend username=[number] fromuser=[number] secret=[password] host=sip.broadvoice.com fromdomain=sip.broadvoice.com

> 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

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

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

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

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

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

Could someone please direct me (via personal email) to a provider with good prices on Polycom Soundpoint IP 500's with POE cables? I need 14 of them. Thanks, Adam ________________________________ Adam S. Robins Executive Vice President & CIO PHARMACENTRA, LLP 5901B Peachtree Dunwoody Road, Suite 380 Atlanta, GA 30328 Office: 770-395-0088 x34 Fax: 770-395-0989 Mobile:

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

Guys. What do I need to configure in order to let my Asterisk receive calls from sip phones, etc not registered with my server on my extension? For example, let people use their asterisks or sip phones to call blah111@server.com?