similar to: SIP URI

So, I'm trying to get the Asterisk vxml (from i6net) working. Having no luck with it. My dial plan has: exten => _X.,1,Answer() exten => _X.,n,Wait(1) exten => _X.,n,Vxml(file:///tmp/menu.vxml) The /tmp/menu.vxml file has: <?xml version="1.0"?> <vxml version="1.0"> <form> <block><audio

Hello: I would like to use Asterisk as a redirect/proxy sip server to route SIP calls on a sip header/parameter basis. I've tried some things successfully: - SIP registration from clients. - On-the-fly compression for wan VoIP transfers: SIP G.711 --> GSM IAX --> (wan) --> GSM IAX --> SIP G.711 - Sending custom parameters in URI: exten => 1,1,Setvar,VXML_URL=var1=value1

I was browsing through the web config of a Sipura SPA-841 (Firmware 2.0.13) and noticed a setting marked 'paging' under supplementary services on the Phone settings page on the advanced admin login. Anyone know how it might be used? Could it be like the Snom - exten => 10,1,SetVar(VXML_URL=intercom=true) exten => 10,2,Dial(SIP/testuser) Craig

Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong

> > When dialling a number, my 7960 always says "External call". Is there > any chance to tell it the difference or make it display the dialled > number (especially nice when using speeddials)? > CallInfoMessage issue, it should be fixed by Jan in the CVS, I will check it out > Is there any way with chan_sccp to have asterisk taking influence on the >

I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten => 777,1,Answer exten => 777,2,Wait,2 exten => 777,3,DeadAGI,a2billing.php exten => 777,4,Wait,2 exten => 777,5,Hangup I am using 777 as the calling card

Asterisk 1.4 We are experiencing an issue on transfers where no progress tones are heard by the caller: 1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF). 1595 answers 2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears MoH. 3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595 hears no ringing When xfer is pressed and the

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=471 Summary: UDP stream DNAT problem Product: netfilter/iptables Version: linux-2.6.x Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: NAT AssignedTo: laforge@netfilter.org ReportedBy:

I recently upgraded to asterisk 11 from 1.8. I had VXML working via AGI in 1.8 - from extensions.conf: [VXML] exten => s,1,Answer exten => s,n,Set(ENCODED=${URIENCODE(${ARG1})}) exten => s,n,AGI(agi://localhost/url=${ENCODED}) exten => s,n,Hangup Using asterisk 11 on the same host with the same config in extensions.conf: -- Executing [s at VXML:1]

Hi again! Also, given my setup below, how do I send caller id to my cell? SIP/83 at callcentric is my cell, when I get incoming call when someone dials into Asterisk - I just see public calcentric?s DID number. I want to send a number of who CALLED IN into the Asterisk and possibly add couple numbers upfront or something like this to signal me that this call comes through the PBX and not

Hello All! I asked week a so ago about how to call multiple phones alltogether (home, office, cell) Dial app looks simple, this is kind of what I have now: --------------------- [globals] IVAN_HOME_OFFICE=SIP/BF8 IVAN_OFFICE=SIP/CFC IVAN_CELL=SIP/83 at callcentric [internal] exten => 101,1,Dial(${IVAN_HOME_OFFICE}&${IVAN_OFFICE}&${IVAN_CELL},60) same => n,VoiceMail(101 at

I have had problems with Samba since the Debian upgraded samba to 3.0.23c, the problem I had was that smbd refused to start. After reinstalling samba I found the problem lay in the smb.conf the new install would start, but if I used my old smb.conf (from my previous version of samba) it would not start smbd. So the point is now, samba is running I have added my shares to smb.conf, when I try

Hi, I have dovecot 2.0.20 running (its an old version, I know, it came from the stable archive at OpenCSW) with Solaris SMF integration working fine. It enables and disables okay. However, I cantt connect to it, it is allowing the connection, but spewing on permissions: Nov 24 17:34:20 proliant-1 dovecot: [ID 583609 mail.info] master: Dovecot v2.0.20 starting up Nov 24 17:34:27 proliant-1

Hello, I've seen that Nuance SIP audio provider supports additional information (parameters and extra headers) in SIP URIs, using the format: sip:user:password@host:port;uri-param1;uri-param2?header1&header2 For example, sip:1234@myserver.com;extra_header=Uui?Uui=Hello Does Asterisk support this format? Is there a way to retrieve the value of these additional headers, and then decide

As some of you know I've been trying to facilitate an involvement with www.tellme.com <http://www.tellme.com/> speech recognition tools and Asterisk. See www.studio.tellme.com <http://www.studio.tellme.com/> There have been a number of people who are already integrating the two and utilizing Tellme as an ASP to deliver speech recognition to their asterisk applications.

Sometimes I find it useful to put C or Fortran sources under subdirs of "mypackage/src", especially to distinguish between my own files and e.g. lapack or blas sources. To get this working I would need a Makefile in my package which contains the appropriate dependencies to these sources in subdirs. But I don't want to add a complete Makefile, I want only a Makevars file containing

Hello, I have some (possibly very simple) questions about subscriptions: I store my old mails in /home/oldmails/ in one Maildir where I have only read-only permissions (I cannot delete, create or modify any files). Under thunderbird, I can successfully log in and read my new mails, but I cannot subscribe (?) within thunderbird to oldmails (it is grey not black as other folders). What did I do

I have the seleniumrc_fu plugin installed and setup according to the instructions, and can run the selenium:test task if there are no test cases. My first test case simply does an open_home_page to verify that things are working at all. Unfortuniately there seems to be an issue with even this simple a test case. The following suggests that there is an issue with the getNewBrowserSession

The first release of Voiceglue is now available. Voiceglue provides a VXML interpreter using Asterisk telephony and the OpenVXI VXML parsing suite. It is released under the GPL, and thus compatible with Asterisk and OpenVXI licensing. The first release is available at the project website: http://www.voiceglue.org There is also a mailing list for those interested in continued evolution of

As a newbie to * there is a lot I do not yet understand. Before I jump onboard I could use some help in evaluating whether * is the way to go on applications I'm about to put together and spec. Here are a number of questions that will help me in my evaluations. I hope this belongs in this group. The first application is a outdial app. with the option of the called party opting in to be