similar to: Cellsocket Report Card (GSM/PCS to FXS gateway)

When I posted my last message about the cellsocket, I hadn't thought to try incoming calls. (I bought the cellsocket to use only for outgoing calls.) Unfortunately, I cannot get the cellsocket to work for inbound calls. CallerID does not work even though I have a GSM phone. The cellsocket answers the phone and then starts ringing any attached phones after answering. Most importantly,

I am posting this in case someone need help.. ========================================================= YOU THA MAN!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! No sure how I will repay you, but anything you need, just let me know!!!!!!!!!!!!!!!!!!!!!!!!!!!!! Thank you, thank you, thank you!!!!!!!! -- Executing GotoIf("SIP/2007-12c7", "0?4") in new stack

I need help from someone who has a working cellsocket, I have received couple email of people who wanted to help, but they just think they know how it supposed to work, but they don't have a working units, and they confused more...I need someone with a working solution to get my cellsocket going.. Thanks!!! Write offlits @ mawise (AT) mail.com -------------- next part --------------

Hi, I'm looking for a reliable, reasonably-priced, single-channel interface between * and US GSM. The VOIP GSM Gateways listed at http://www.voip-info.org/wiki-VOIP+GSM+Gateways (VoiceBlue, QUTEX) are multichannel systems, very expensive ($2500 or more). Next step down, there are various Fixed Cellular Terminal (FCT) or Fixed Wireless Terminal (FWT) devices. These typically have an FXS

Hi all, I've just connected a cellsocket to my * server so that I can keep call costs low for outgoing calls to mobile phones. I'm using an old fashioned 5110 cellsocket (because I have a few of these phones and I think that they are very reliable). For those that don't know the cellsockets they convert a mobile phone into a POTS line. The cellsocket has a funny dialtone but that

Hi, I try to use ISDN4Linux drivers with Asterisk. In modem.conf i put /dev/ttyIO. Everything is OK when i lauch asterisk but, when i call Asterisk nothing happen. Someone can help me ? Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030407/13e78d8e/attachment.htm

hi, I use IAX softphone with asterisk and I notice that a call between two IAX softphones end after 1 min. Then I can't hear anything but the call still in progress. I have this log in asterisk IAX debug: Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 21589 DCall: 00001 [192.168.1.22:4569] Tx-Frame Retry[000] --

What models? Jeff Noxon (jeff-asterisk at planetfall.com) wrote*: > >I just purchased a bunch of Nortel Meridian POTS phones that support >intercom on the 3rd pair. I intend to get it working with Asterisk. >The phones support MWI, have a 3-line display, callerID, call waiting >callerID, 2 lines...very nice. > >On Thu, Feb 27, 2003 at 01:07:19AM -1000, James H. Thompson

Hi, A question about transfert. How can I make transfert with the the person who call. X call Z and X transfert Z to Y. I only succeed to do X call Z and Z transfert to Y. If someone have a solution it will be very good =) regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I notice that i don't have callerID in my Voimail when someone drop me a message from H323 Client. Is there a tip to have this CallerID ? Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030804/af4203ca/attachment.htm

Hi, I use with asterisk gnugk a gatekeeper for h323 client. I don't understand why asterisk can't have the H323-ID (callerID). In the gatekeeper's monitor I have this H323-ID but not in asterisk. Does anyone know something about it, or how can I send a caller ID to asterisk ? Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi everybody, Is there SIP client which work with Asterisk and can be embedded in a HTML page ? Thanks Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031029/226a8b1b/attachment.htm

Hi, Is there IAX client in Applet JAVA which can be embeded in a web page ? Best regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031208/c388ef61/attachment.htm

hi, how can I add or remove this line "include => context" by the command CLI ? regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030827/979ddd76/attachment.htm

Hi, Chatsiri! > ---------- Forwarded message ---------- > From: Chatsiri Ratana <insiderboy at gmail.com> > Date: Wed, Jun 6, 2012 at 2:15 PM > Subject: [LLVMdev] Compile-rt throw error undeclared identifier 'O_CLOEXEC' > To: llvmdev at cs.uiuc.edu > > > Hello All, > > I build LLVM source code version 3.2 from SVN repository. After I > build

Many of you are familiar with how lousy Festival sounds. AT&T has a product, NaturalVoices, that sounds much better. There are male & female voice fonts for US/UK/Indian English, French, Spanish, and German. I am considering offering a linux-based text-to-speech engine based on the NaturalVoices runtime. An asterisk module would also be provided, making it easy to add natural sounding

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

Hi, I try to use conferences with Asterisk but i'm not succeed in doing this. I have set in meetme.conf : conf => 4000 and add an extension in extension.conf. But i have Asterisk WARNING Unable to open pseudo channel. Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, Someone know how to install conference (MeetMe) without a zaptel devices ? Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030414/f8a9140a/attachment.htm

Hi, I have a little question, I use asterisk with Netmeeting client. When I call netmeeting client with a phone. I don\'t have his ID in netmeeting window i have something like : ???;..dhz instead of 28. Someone know a way to display this ID ? Thanks you so much Rattana