similar to: Chan_Cellphone

Hi, someone can help me, I am ot able to find the way how to user Ajax.updaterto test if the request give some positive or negative result. I am able only to return the result inside a div. An example is appreciated. _______________________________________________ Rails-spinoffs mailing list Rails-spinoffs-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org

I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060617/8f4449fa/attachment.htm

Hi, I installed H323, however when I make a call from SIP Phone -> Asterisk H323 -> Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I installed this program but I am not able to configure, it does not want to work. Someone can help me?

I ran into a problem today while trying to compile chan_cellphone version 17 on a CentOS 4.4 machine. Apparently the bluez and autoconf versions were to old and as I tried to install the latest version, I found that the new bluez-lib would install and allow the chan_cellphone to compile, but bluez-utils required an update to D-sub which in turn required python 2.4 or better. That apparently in not

Hi, someone know a good webphone, possibily a free one Thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060627/0e83bc29/attachment.htm

There is a way to force the order of the codecs in the sip.conf since the allow seams to let know only the accepted codec. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071022/619b8f2b/attachment.htm

hi everybody, I'm currently planning a small-sized web-applicaiton allowing users to call-in via phone. the phonecalls should be recorded and processed further by some custom scripts - sounds like asterisk is a perfect match for this app. however, during prototyping I have no ISDN-connection whatsoever available, so I was asking myself if it's possible to connect a cellphone via

Hi, I saw one site (bubbleshare) that it is able to caputer the click on the log in link, however, I cannot understand how they can do that Someone can explaint it to me? Thank you _______________________________________________ Rails-spinoffs mailing list Rails-spinoffs-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org http://lists.rubyonrails.org/mailman/listinfo/rails-spinoffs

Hi, I'm using a script in order to send out my faxes with the application txfax, therefore, I do not know how to see if the faxes are sent. Any idea?

I have this simple context I am register to an external provider and when I am not home I would like to transfer the phone outside The problem that the call goes in loop I cannot understand why. Can you figure out my error? Thank you sip.conf register => user:pass@provider/400 [inside] exten => _4X.,1,dial(SIP/ext_400_124/5551234444,5,tT) exten => _4X.,2,hangup -- Executing

Changing it to only apply to the MinGW-Makefiles generator is maybe the safest method. Though you would have to check if MINGW is only set in the Mingw-Makefiles generator and not In the MSYS-Makefiles or Cygwin generators (should be in the docs). -- Johannes S. Mueller-Roemer, MSc Wiss. Mitarbeiter - Interactive Engineering Technologies (IET) Fraunhofer-Institut für Graphische Datenverarbeitung

Hi, this is my first post to this list, so please excuse if submitting a patch without previous discussion is considered bad form or anything similar. I encountered a bug in the CMake build while using MinGW (non-MSYS, non-CYGWIN) where the LTO_export fails with a "The syntax of the command is incorrect" error. This error was previously fixed for Windows in general using

I'm running chan_cellphone version 13 on the latest svn trunk (as root). I believe I have chan_cellphone set up correctly (bt addr and port retrieved from the "cell search" CLI command). When I load the chan_cellphone module, my Motorola V3m asks if I want to allow "Asterisk PBX", I say yes and enter the 0000 for the pin, then my phone tells me the pin is invalid. Here

This bug report http://bugs.digium.com/print_bug_page.php?bug_id=12038 apparently has been fixed. I dont see anything on the page saying what released version of asterisk this is in. How can I tell that? jerry

I have a Sangoma b600de analog card using dahdi 2.1.0.4 and I get the following results (same dialplan, config etc): Asterisk 1.6.0.1 => works fine Asterisk 1.6.0.9 => can't dial out unless I dial in once or apply patch >>>==> http://bugs.digium.com/print_bug_page.php?bug_id=14577 Asterisk 1.6.1.0 => can't dial out, regardless of patch or inbound call first.

I just took my first cent server into production and now saslauthd keep crashing after brute force attack. I found a bug report so this has already been reported but not fixed. http://bugs.centos.org/print_bug_page.php?bug_id=2860 I assume this has to be a large problem for many people and am surprised it hasn't been fixed yet. Has anyone found a work around for this bug? Is there a

Hello, This morning I noticed the following output from "yum update": Installing: drbd82 x86_64 8.2.5-1.el5.centos extras 209 k replacing drbd.x86_64 8.0.11-1.el5.centos As far as I'm aware DRBD works fine for me. Is there a way I can find out about the new release and weather I should upgrade? I can't figure out the CentOS issue tracking system

JABBER: gtalk_account OUTGOING: <?xml version='1.0'?><stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to=' gmail.com' version='1.0'> localhost*CLI> jabber show tes JABBER: gtalk_account INCOMING: <?xml version="1.0" encoding="UTF-8"?><stream:stream from="gmail.com"

28th February I am working with Asterisk 1.2.15. I have configured sip.conf for two soft phones (I am using Xlite).I have installed the Bluez stack and so far, i manage to make a phone call from a soft phone to a GSM network. However, i have an audio problem. The soft phone can be heart by the GSM costumer but the voice in Xlite is not transmitted to the GSM. In asterisk all i got is the