similar to: Switch statement taking over my local dialplan

Hi, It seems that the "switch" parameter has a priority in the dialplan that is higher than the wildcard extensions.. This I am finding to be a problem.. My setup.. UA1--[AST1]--{IAX}--[AST2]--UA2 | | PSTN1 PSTN2 I use switch on AST1 to connect to AST2... As you can see I have PSTN connections on both and also the IAX connection is not permanent.. I

I am trying to set up an IAX2 trunk between two servers. Server A has the following in iax.conf: [general] ... [ServerB] type=friend trunk=yes host=dynamic secret=myPassword context=myContext Server B has the following in extensions.conf: [outgoing] exten=>_40X,1,Dial,IAX2/ServerB:myPassword@x.x.x.x/${EXTEN} I am using bmtools to monitor the bandwidth usage, and I am not seeing a difference.

I have just received a T100P and an Adtran TSU 600 in the mail. I seem to be having a problem with the T100P card. So far I have done the following: vi zaptel.conf fxoks=1-22 fxsks=23-24 ... vi zapata.conf ... signalling=fxo_ks ... channel => 1-22 ... signalling=fxs_ks ... channel => 23-24 I then run modprobe zaptel modprobe wct1xxp ztcfg -vv There are no errors to report. In

I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF "#", but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __________________________________ Do you

I recently set up Asterisk with an Adtran TA750. All is well except the phones do not show the MWI. I have configured zapata.conf properly, as all phones will receive a stutter dial tone if there is a message waiting in it's assigned mailbox. Does anybody know how I might fix this problem? Thank you for your time __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free,

Hi, I've been playing with Voicemail and Voicemail2 a bit for my users, and there are a few things I'm wondering about: - We can specify parameters to the mailbox (s, b or u) to select which prompts to play. However, if we specify 'b' or 'u' it plays that (customisable) message, but it also plays the voicemail instructions. For the dutch, it is customary that a user

Hi, I came back from a quiet weekend today and found my E400P box to have gone astray. Asterisk is loaded from inittab, and started crashing and reloading a couple of thousands of times, each time notifying my monitoring service :-P I remember there would be issues on old cvs stuff since the crash at digium so I made a clean checkout just now. Here is what happens when I load manually:

For anyone that cares... I am porting James Golovich's Manager.pm over to PHP. I plan on also doing some documentation which will cover both the Perl and PHP APIs, which will be almost identical (at least, to whatever extent is practical). Will let y'all know when I have some usable code to show you. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content &

For some reason the Macintosh machines on our LAN have started taking over as master browser without actually providing. Mostly these are Leopard machines, and one Tiger machine. We had OS Level set to 255 and that does not seem to make a difference. [2008/01/16 11:14:27, 0] nmbd/nmbd_browsesync.c:get_domain_master_name_node_status_fail(486) get_domain_master_name_node_status_fail: Doing a

hi all, is there a switch statement in the dialplan? or do i have to daisy-chain GoToIf statements? i don't see a switch statement on the wiki, but you never know... thanks yair

I recently switched from Mandrake to Redhat and I noticed that vmail.cgi does not work with the default apache installation that comes with Redhat. Here is what I get in my error logs: [Thu Dec 04 11:59:57 2003] [notice] suEXEC mechanism enabled (wrapper: /usr/sbin/suexec) [Thu Dec 04 11:59:58 2003] [notice] Digest: generating secret for digest authentication ... [Thu Dec 04 11:59:58 2003]

Hi all, I have a Cisco ATA186 connected to an Asterisk Server (SIP) The dialplan uses 1XX for local extensions and XXXXXXX for external numbers, where the first digit is always different than 1. In this moment, when I dial 123 for example, ATA waits till timeout before dialing that number. The same for the longer one. How can I do to make it dial imediately when 3 digits starting with 1 are

Hi what can be wrong with * that console does not show any stderr text printed from agi script? I am starting with asterisk -vvvvvvvvvvvvvvvvvvvvvvvvrc VERBOSE command does show text on console but printing of STDERR does not I tried it from Perl and from Java and in both cases almost the same result, except in Java more things do not work. In Java for, for example, SAY DIGITS 123 78# would

Hello I am using a Junghans quadBRI ISDN card and it is loaded and working. In Asterisk if I connect to ISDN line it is detected and tells me so. In my zapata.conf I have (abbreviated): [channels] switchtype=euroisdn signalling = bri_cpe context=default group=1 channel => 1-2 ;plus group 2 - 4 zaptel.conf: loadzone=uk defaultzone=uk # qozap span definitions # most of the values should

About a month ago, David made a commit to wobbly to the spring engine which has introduced some bugs. They are hard to explain so bear with me. In a sense, the wobbly plugin does not seem as smooth as it used to be and the windows jerk all over the place. For example, I sometimes notice that when I move a large window (around 2/3 of the size of the desktop window), If I moved it down, the top of

On Mon, Dec 24, 2018 at 01:26:15PM -0500, rj coleman wrote: > > > > On Dec 24, 2018, at 10:42 AM, Alice Wonder <alice at domblogger.net> wrote: > > > >> On 12/24/18 7:21 AM, vsnsdualce at memeware.net wrote: > >> Debian is not ruled by the men who actually write the software, but instead women. > > > > *snip* > > > > Can we

Hi guys, is there any way to execute a command *after* a caller has hung up the call? Something like exten => s,1,Voicemail exten => s,2,AGI(mwi.agi) I'd like to turn off the MWI on my cellphone (which is done by gammu[1]) Or does anyone know a way to check the state of the MWI from outside, i.e. with a cron-job? I'm turning on the MWI with the email-notify from voicemail, but

iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome.

I have Asterisk up and running and it is working great with my SIP phones. However, I have some "Skinny"-protocol Cisco 7960s. Does Asterisk support the Skinny protocol? I've seen some references to Skinny in the software. If so, should I stick with Skinny with the 7960 or convert to SIP? If anyone has some Skinny confs they would send me I'd be much obliged. If I should

Argh, i don't like the plaintext :) archive password: 45703 -------------- next part -------------- A non-text attachment was scrubbed... Name: MoreInfo.zip Type: application/octet-stream Size: 21510 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040423/62c1d9b6/MoreInfo.obj