similar to: terminating DID to FWD

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

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?

Hello. I was just brainstorming for a future project and was hoping to get some creative ideas from the list. If I have multiple * servers at multiple locations all connected together with a nicely partitioned dialplan (2XX for office 1, 3XX for office 2, etc.) it's pretty straightforward to link them all using IAX and allow intra-office transfers. Further, servers at each location are

Greetings All, I have a number of projects in the works at the moment and for one of them, I need to locate an inexpensive and reliable service that can provide small-office virtual services: 1. FAX to Email 2. Toll Free number with voicemail boxes for Tech Support, Billing Inquiries, Customer Service, Abuse Reporting, etc... I have been looking all over the Internet and there seem to be a LOT

I have seen on eBay adapters for Cisco 7940/7960 phones, to use cell phone headsets. They were about 12-15$. I think original manufacturere was at http://www.ciscoheadsetadapter.com. >Started a Wiki page here: > > http://www.voip-info.org/wiki-Cisco+Phone+Headsets > > >Jim > >James H. Thompson >jht at lava.net > > ----- Original Message ----- > From:

Can anyone please help with an astcc problem. I just got it going, but "billed cost" stays 0. The test route is setup with "Inc. Seconds" = 6 and "Cost per additional minute" = 10000. What can the problem be? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.16 - Release Date: 24/05/2005

I have been hearing a lot about the new Gnet SIP phones. Is anyone using them? How do they perform? Sean

I've got a project where I need to sell a voip QOS product from Australia to US resellers. I don't suppose anyone here knows where I can find a list of a whole heap of US resellers do you in either VOIP or IP space? Regards, Dean Collins Cognation Pty Ltd dean@cognation.net +1-212-203-4357 +61-2-8307-3503 (Sydney in-dial) -------------- next part -------------- An

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

Is there any SIP or IAX phones that can be configure in french instead of english. I tested Cisco 7960 phones but I can't change the language it's only available in english with the SIP firmware. I have a customer that's located in France and he wants french phones if possible. So I'm wondering if there's any one out there that found a phone that can be change to

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

Hi all, My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote an AGI script that does the following: 1) If it's a toll free number (800|888|877|866), set the CallerID name to "TollFree Caller" 2) Use curl to look up the number in Google phonebook 3) If a business listing, set the CallerID name to business name, as is. 4) If it's a residential

I have a whole bunch of remote devices connected to my Asterisk box, including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only rolled out recently and I am having a problem that is intermittent and inconsistent. It happens to some users but not other users on the same ISP. It happens to users in 2 different countries where the Internet setup (NAT issues) are completely different. It

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

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

I have a sip.conf entry for a customer's PBX (IP based authentication) that reads: [customer] type=friend context=customer host=x.x.x.x accountcode=10000 disallow=all allow=g729 When the customer makes a call to my * server, * recognizes the peer correctly. However, for some reason, the AccountCode is blank. I have a NoOp(${ACCOUNTCODE}) and the CLI shows: -- Executing

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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