similar to: CallerID Name with IAX Providers

Hi folks, I think my little agi script is ready for the big one-oh-oh. Available at http://muware.com/asterisk is cid_rewrite-1.0.0. This agi-script does the following: - Standardize incoming caller-id numbers to adhere to US dialing code; NANPA numbers are reformatted to 1+10, international numbers become 011<country-code><number> (this is customizable with a little PHP knowledge).

Can Meetme be made to work with G.729? (I gather not) If a call comes in (internally or externally), the call comes in as a G.729 call, which then re-negotiates to a G.711u call when if gets transferred to a MeetMe room. Is there a way to set up asterisk that will allow me to have internal phones renegotiate to G.711, with the external lines instead transcoding within asterisk. (runtime is more

For those folks who are still using it -- I updated the cid_rewrite script. I noticed that two of the providers were "iffy" and one had changed format a little while ago. It's working again. http://muware.com/asterisk has the latest (1.2.0) Enjoy, -- JM

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

Is it possible in any way to use an Outlook contacts database as the source for the internal Asterisk database that is used for callerid lookups? Thanks!

Anybody know a good IAX provider for Canadian DIDs? Mohit. -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] "There are 10 types of people. Those who understand binary, and those who don't."

I would like to use * as a standalone voicemail system. As such I need it to be able to outdial a certain extension for MWI-ON and another extension for MWI-OFF Is there anyway to get * to automatically dial an extension when a voicemail is left and another extension when the mailbox is cleared? Thanks -------------- next part -------------- An HTML attachment was

I am shopping around for an IAX provider that provides both outbound minutes and an inbound DID for a flat fee. I have been looking at TELIAX. http://teliax.com/ Does anyone have any experience with them? What codecs do they support? Inband or 2833 dtmf? Voice quality and reliability? Any comments and/or other recommendations would be appreciated. Thanks

Can i register a cisco 7902G with asterisk????? what i have to do??? thanks in advance. R -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041130/12bf3c06/attachment.htm

I have been having problems getting PrivacyManager to work correctly. Right now I am running the 1/21/05 CVS but I have been unable to get this to work on asterisk-stable either. You can see from the debug below that the inbound call is arriving via IAX2 and both the CALLING NUMBER and CALLING NAME are both set as "Unavailable". However, PrivacyManager executes and determines that

I'm using P0S3-08-2-00.. I noticed the callerID started showing up with the number, then @<proxy-addr>... So the callerID on the phone looks like: 2145551212@10.10.10.10 which of course is logged in the missed calls exactly like that, and completely foobars the dialing string if you try to dial a missed call by simply hitting the dial button. Can anyone else verify this problem? 7.5

I am having a problem with the Vmail Web app and was hoping someone has some advice. I have been able to install vmail.cgi. I am able to bring up the vmail web page and login with out problems. However, I never see any vmails listed even when there are vmails present in the directory [root@linux-home INBOX]# ls -l total 40 -rwxrwxrwx 1 root root 255 Dec 2 14:45 msg0000.txt

Hey Everyone, I am using NuFone for 866 inbound service and I am trying to figure out the callerid part of it. Any call into my * system just shows "Toll Free Call" and will not give me the calling party's caller ID info. Is this just something I have to live with using NuFOne, or did I miss some type of config in * that will grab the callerID other than the inbound 866 number...?

I recently changed to all IAX providers for my DIDs, and none of them offer incoming caller-names. Back in the days when I did have incoming caller names, I found the names provided by the various phone-companies fairly useless -- "WIRELESS CALLER", "YOURCITYHERE" etc. Last Friday I finally set off to roll my own, in order to meet the following requirements: - uniformely

> 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

Hi folks, I am working on my student internship to USA, california and would like to attend a R Advanced Programming course while I am in USA. Where can I find such a course. I am new to this list. Trevor O'brien >From Ireland. [[alternative HTML version deleted]]

Is there is an open source ACD component for Asterisk? Likewise, is there an open source CTI component that will work with Asterisk? Regards, Jim O'Brien -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040504/e6ee98da/attachment.htm

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

Sorry if this is a resend, but it didn't appear to go the first time. > Sorry if this is not the correct place to post this..... > > > I have downloaded the cid_rewrite scripts that are located at: > http://www.muware.com/asterisk/ to my AAH v1.1 system. > > I apologize for my ignorance, but it says that I need to modify the > agi_config.php, but doesn't

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 =>