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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 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

Is it possible to terminate (or forward) lets say 800 DID number to FWD number. -- #Joseph

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

Guys. I know there are wifi sip phones out there but I have a question, are any of these phones "anti explosive"? By that I mean, there are certain regulations about phones or cel phones that are not recommended to operate in environments like gas stations due to sparks and the chance of ingiting gas fumes. Are there any wifi sip phones out here that have complaince with regulations to

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

Guys. Anybody using sipuras 2002 knows if there is a way to make the phones connected to it to autodial an extension when the phone is picked up? For example, if the phone is on a police booth (building entrance) and you want the guys to just pick up the phone and make the phone auto dial the receptionist extension without the guys having to dial anything (ala batphone). Is this possible with

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

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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

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

Hello. I am using a snom 320 running 5.3.6 with Asterisk 1.2.4. In the sip.conf entry, I have mailbox=1234@default and vmexten=*98. The light on the snom 320 turns on when I have voicemail and the retrieve button dials the correct extensions. However, the light turns off immediately after making the call to voicemail, even if I do not check the voicemail. Any idea on how to get this to behave

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 =>