similar to: Using Local Channels with Originate

I have Meetme conferencing compiled for Debian as per http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the outgoing dir, and it intitiates a call to a local extension as a channel, but the call seems to block on the Meetme() command. That extension completes the outgoing Dial(SIP) command to my phone, announcing that leg is the only member of the conference, and just waits. If I

Probably just me being dumb, but I am trying to update my asterisk to the latest (1.2.5) When I go to my /usr/src/asterisk I type: make upgrade make install This seems to be doing it's thing, but when I type show version from the console (after a restart) I still get: Asterisk SVN-branch-1.2-r7231 built by root @ notdeadyet-imac.local on a Power Macintosh running Darwin on 2006-03-04

Anybody knows if it's possible to disable zap echo cancellor from dialplan only for certain outbound calls ?? I share the same phone lines for voice calls and faxes. Iaxmodem works fine for me only turning off the echo cancellor, but I need it for voice calls. Any ideas ? maxx

Hello to everybody, I have a problem with the disposition filed that asterisk write in mysql table. What I notice is that for every outbound calls (for example to a mobile phone) I see in disposition field the string "ANSWERED" when I reject the call and also when I really answer the call, while in the variable DIALSTAUS I have the correct status of the call (BUSY, CHANUNAVAIL,

My problem is simple and I've issued it about 3 weeks ago. I want the UAs to authenticate with a number to the SIP server. Is this possible? For example, I configured an AT-RG613TX (Allied Telesyn Residential Gateway). In its configuration it is not possible for me to skip specifying a number (ex. 102) along with the username. I've looked into the source code (SIP implementation) of

Ok, so multiple people have said that hinting is possible with chan_sccp on the 7940/7960's and such, has anyone got this working? How do you go about getting this to work? I'd use the wiki, but it's link to the mailing list topic on that doesn't work anymore :( -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

This mail announces the release of Cortado 0.2.2 'Really Tested Verily Exceptionally'. This is Cortado, a multimedia framework for Java written by Fluendo. It contains: - JST, a port of the GStreamer 0.10 design to Java - jcraft, a copy of the JCraft JOgg/Jorbis code - jheora, an implementation of Theora in Java - codecs (currently only containing the Smoke codec, a variant on Jpeg) -

I am originating a call to a Local channel using an Originate Action: Action: Originate Channel: Local/dial at outdial Context: outdial Exten: answer Priority: 1 Timeout: 45000 ActionID: some_id In my dialplan, I have this: [outdial] exten => dial,1,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten =>

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri---- * ------ Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are

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

Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI> -- data = @89930:0107901723168212 -- capi

I am trying to get updateconfig working. I found an example of updating configuration files here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Upd ateConfig When I tried it the conf file was updated but the new entry was not added. action:updateconfig reload:no srcfilename:manager.conf dstfilename:manager.conf Action-000000:append Cat-000000:newuser

I'm having a real hair-raising problem here and I thought maybe someone could help. At least I hope so. My workstation was running 3.0.2a, upgraded to 3.0.5. After upgrading to 3.0.5, I can no longer mount shares on my 2003 server. This started happening on an upgrade to 3.0.4 as well, I might add. Permissions-wise: I own the directory mounts on the local Linux workstation,

is anybody else having problems with the new java 1.6.0.22 ? i am tried both cortado-ovt-debug and cortado-ovtk-debug - neither seem to work anymore once i installed the new java. once i go back to the old java 21 everything works fine. my jar file downloads for console messages (below) came from: http://downloads.xiph.org/releases/cortado/cortado-ovtk-debug-0.6.0.jar i have included

I have been trying to track down the cause/fix for a problem and I am out of ideas... I am hoping one of you can point me in the right direction. The symptom is that when a calls is placed from an internal extension through an analog line to a number on the pstn the caller can hear the callee but the callee can not hear the caller for as long as ten seconds. The problem appears to happen fairly

Hi, I?m working on a rails application that uses data from an existing ms sqlserver 2000 database. Unfortunately I can?t get the connection to work properly. I used http://wiki.rubyonrails.com/rails/pages/HowtoConnectToMicrosoftSQLServer to make the connection. - Inserted the ado.rb - Changed my database.yml to development: adapter: sqlserver database: database_name host:

GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can

Hi all, I've got some questions to post in regard to running asterisk in a production-grade environment, specifically targeting high-density IVR applications. No VoIP involved, just straight PSTN -> * and perhaps the occasional outdials or agent-based predictive dialing. 1) Which user would you run * under? 2) What other security-related issues do you have to resolve? 3) How do you handle

Hi all, Does anyone have an application/script or extensions.conf file which will do the following? "When a new VoiceMail is left for a user, the asterisk system will place a call to a cellphone/pstn number(via some provider). When the user answers his cell/home phone, comedian mail will ask for his password and he can check his Asterisk VM?" Anyone have any examples of it