similar to: Determine if Call is from a cellular phone

Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/92c7f946/attachment.htm

Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the XXXXXXXXXXXX-directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in

I am having major problems with the first run of Snom 360s that rolled out last month. I am working with the US vendor and they in turn are working with Snom but I wanted to see of anyone else was using these or having issues with them. Issues: Speakerphone/Hands Free volume spikes up and down during a call. You have to manually set the volume during every call. This makes it totally unusable.

Hi List, I am using asterisk 1.2.14 with real time and I am trying to send the email to more than one email address. In that field I put in user1@domain.com;user2@domain.com. When the call goes to VM I see in the CLI: uniqueid => 17 customer_id => 0 context => techmast mailbox => 14 password => 1234 fullname => Sales and Service email => user1@domain.com email =>

Hi List, I was thinking the other day that even with T.38 there are still some issues with faxing. I was thinking of a protocol that instead of just sending down the fax tones an ATA or "VOIP fax machine" would get the entire fax convert it into some sort of image and pass it down the line to the receiving end. I got the idea from RFC2833. Yes I know that fax machines send bit by bit and

Are there any phones whose display can show queue statistics, ie: calls waiting, etc, on the phone itself without too much trouble with Asterisk? Especially while the phone is in use (on a call)?

Hi List, I have a client that has an interesting request. He wants to have people call in to a conference room and all be able to talk however they should not hear each other. There should be admin access to for one user to call in and be able to listen in to everyone that is talking (they may want this admin to be able to talk to). Any ideas ? Thanks. Dovid -------------- next part

Hi List, I wanted to know if anyone knew of a way with asterisk to "switch the voice" of a caller from male to female or vice versa. Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070426/2d483875/attachment-0001.htm

Does anyone know if it is possible to change the packetization time in Asterisk ? I was told by a client of mine that adjusting this with using G729 can greatly lower the amount of bandwidth used. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070819/b0cc470f/attachment.htm

Hi, I am currently playing with DD-WRT and I like it. I am looking for something that is more "SIP Aware". Anyone know one those that are out there ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071126/eb28ce44/attachment.htm

FYI. http://www.telular.com/products/index.asp These look like the right solution for any one wanting a cellular FXO device for * that interface with the digital cellular networks (GSM ,CDMA etc) similar to the www.cellsocket.com or the mystery FCT's http://www.ericsson.com/products/products_az.shtml#F or those old Motorola Bag phone adapters -------------- next part -------------- An HTML

Um texto embutido e sem conjunto de caracteres especificado associado... Nome: n?o dispon?vel Url: https://stat.ethz.ch/pipermail/r-help/attachments/20061101/4218da11/attachment.pl

I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider ----> Asterisk -----> ATA and vice versa (ATA -----> Asterisk ----> SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP

Hi List, I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I keep getting an error from asterisk of -- Got SIP response 415 "Unsupported Media Type" back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone have a hint as to what it may be ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have a client that wants a phone system that will play sounds from a sleep machine. I tried using all different formats (GSM, WAV, WV49, MP3 etc.). Over SIP it was OK however with the PSTN it broke up from time to time. I assume this has to do with the fact that the PSTN is limited to 8khz. Is there something I am missing here or is this simply a limitation of the PSTN? Regards,

Does anyone know of any Cellular providers that will allow VOIP connections to thier cellular phones? I'd love to be able to call a cell phone without using the PSTN. I would think it would make sense for the provider, they still get to charge normal minutes, but don't have to burn a connection to the PSTN.

Maybe someone here has found a good solution to this problem. I voulenteer with a local Search And Rescue unit and I was speaking with the senior members about how they interface the command trailer PBX with the PSTN or cellular networks when they are on scene at a remote location. Turns out they don't. Thus that got me to thinking about how one would get Asterisk to interface with a

Hello all, I was wondering if anyone has had much experience with Realtime Asterisk. I like the ability to setup my extensions and voicemail boxes in MySQL, but I have a huge worry. What if MySQL crashes. I played with rtcachefriends, but can't seem to find a way to have asterisk store the extension information to ensure the phones will continue to work even if MySQL has a hiccup. Any

Hi, i have some spare (read: Boss get's a new one every few month ;)) Android Phones laying around. Does someone know a way of using them as a mobile gateway for asterisk? I could not find any SIP-Gateway in the Market, and i don't think it's possible to use the GSM Audio directly with something like chan_datacard... Regards, Jay -------------- next part -------------- An HTML

Hi all, I have an asterisk box set up with a bri card (using zaphfc). I have a bri cellular gateway connected to it beacuse I'd like to route all my cellular calls through that gateway. The probel I encounter is that when trying to dial a phone number, I've the message : unable to create a zap channel. My card is normally well configured because when connected to the NT, It works