similar to: ASTCC single stage + no access number + auth usingsip username and password

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

> 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

I saw these adapters on eBay. 2.5mm stereo jack to modular RJ-9 jack. I think original site is http://www.ciscoheadsetadapter.com Mike >Nabeel, > I am very interested in what you came up with for a 2.5mm to RJ-10 > adapter. > >I played with every combination I could think of but the best I was able to >come up with had a echo of the far end voice back to the far end.

Hello. I have my * server set up and working perfectly. I wanted to allows calls to sip:nabeel@sip.myserver.net. In sip.conf, I have: [general] context=default Also, in extensions.conf, I have: [default] exten => myname,1,Goto(internal,nabeel,1) However, when I make a call using a "Direct Dial IP" account in X-Lite, I get the following error in *: Failed to authenticate

> 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

Ronald, Grandstream products have a one year warrantee. If you don't have any luck with Pulver, contact us and we can probably get your phones exchanged. Please don't assume that your experience with Grandstream is typical. We sell a lot of these phones and the overwhelming majority of the purchasers are very happy with their units. The quality has improved tremendously over the last

> 9105551212 => 1234,Gary Carr,email@domain.com,attach=yes Syntax is: Mailbox => password,Name,email,pageremail,options So, that should have been (added delete, it's a good idea if you're attaching). 9105551212 => 1234,Gary Carr,email@domain.com,,attach=yes|delete=yes -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990

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 =>

Thanks for that. I just got rid of packet 8 and went with 100% asterisk in my house. But I use the PAP2-NA and RT31P2 from Linksys for my FXS ports. But would like to have an extra FXS laying around just in case.. .o-------------------------------------------------------o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From:

It seems that when setting qualify = 200 or qualify = yes in the database for a sip friend/peer, RealTime does not update the registration status like it should. I also have several peers which have been offline and Asterisk still reports them as registered, even though the registration seconds are only 200. Asterisk Ver: CVS HEAD 12/1/2004 Layout of sip_buddies: mysql> describe

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. 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

Come to think of it since the DTA310 uses DNS to find the SIP server, you could setup a DNS cache and override the DNS entry for what packet8 uses (proxy2-eqix-sjo.packet8.net : 15062, over here) to point to the IP of your own SIP server? Kind of a hack but it should work as long as it's running on port 15062. I am very new to this so I don't know if there's a port standard for SIP

Hello. I have an * server set up on a public IP. I have SIP clients at three different locations, all behind NATs. I have all the SIP users set up this way: [user1] type=friend username=user1 secret=password1 callerid="User 1"<101> host=dynamic qualify=yes context=outgoing All three SIP clients are configured to use STUN (using stun.fwdnet.net:3478). Furthermore, I have

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

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 have got my first * server set up and serving users in three different locations over the Internet. This is currently a test setup so I am experimenting with the different features of *. When I set up asterisk, I only checked out the Stable source of Asterisk from CVS, and compiled it. I did not download nor compile libpri or zaptel. Now, I have internal calling and calling through my IAX

Hello. I've set up a X100P and got it working. Now, I need to set it up so that it checks if the line is being used before attempting to make a call on it. I tried: exten => _NXXNXXXXXX,1,ChanIsAvail(Zap/1) exten => _NXXNXXXXXX,2,Dial(Zap/1/${EXTEN}) exten => _NXXNXXXXXX,102,Dial(IAX2/voipjet/${EXTEN:1}) but that only goes to 102 if another device on the * server is using the Zap

Dear List members- I am trying to configure ASTCC (Asterisk calling card application) but having a hard time to configure it properly. My project deadline is approaching and couldn't figure out how to make ASTCC functional. Here are some details what I have done so far. 1) I have installed ASTCC successfully. 2) I can access astcc-admin.cgi script without any problem. 3) I have created