similar to: Having Trouble With Wait Command in Callback Context

When using a SIP phone with Asterisk, hitting the # key (pound or hash depending on where in the world you happen to be) tells Asterisk that there are no more digits coming, and to put the call through immediately based on the digits already entered. This is the same functionality as the PSTN (at least in North America). However, DISA just sees the # as another digit, and therefore pressing #

I am having some hardware problems with the Linux machine where I have Asterisk installed and want to use a different machine. Assuming I install Asterisk on machine number 2, is it possible to just move files over from the old machine to the new machine and the new machine will behave like the old? Anyone have a list of the files that would need to be moved? (Obviously the *.conf files in the

I want to essentially transplant my existing Asterisk server to a new machine, and take the old sever out of service. Assuming I install Asterisk on the new machine, does anyone know what files I would have to copy over? What comes to mind are the *.conf files in /etc/asterisk, as well as the voicemail audio files. Anything else? -------------- next part -------------- An HTML attachment was

Dear list, I have problems with DISA on an specific server with Asterisk 1.4.26.2. After starting DISA I can only press one key and DISA is jumping direct into the context without waiting for further digits. In dtmf.log I found this: [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin '7' received on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin passthrough

New problem, I figured out how to get the extension working and internally it works just fine. If I pick up a phone and hit 501 my cell starts ringing. However if an inbound caller dials that extension Everything seems to stop when it trys to bridge the two trunks together. Sound familiar to anyone? exten => 501,1,Macro(dialout-trunk,1,5551212) exten => 501,2,Wait,1 exten =>

When I use DISA I get congestion when I try to reach 1-800-number: Here is the context: [disa] exten => 087,1,Answer exten => 087,2,DigitTimeout,8 exten => 087,3,ResponseTimeout,20 exten => 087,4,Authenticate(985) exten => 087,5,DISA(951|disa-access) [disa-access] include => tollfree include => outgoing-voipjet [tollfree] ; ; terminate toll-free no.'s via fwdnet ; US

Hi/ I am newbe in asterisk. I try to setup callback with Disa on my home server Anybody help me, pls

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123

I am wondering if it is possible for an extension that is served by a zaptel device to revert to dial tone once a call disconnects. For instance, if I make a call to another extension, talk with them, and THEY hang up, can I then be presented with a new dial tone rather than a congestion tone? Further, can an extension be set up so that, once the call goes back to dial tone, if the user does NOT

Actually doing both sounds good to me. Can you explain further about ringing them all at once? Here is how I tried to make mine work and failed... {global} PHONES0=SIP/2000 PHONES1=SIP/2001 [local] exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I

I have a friend traveling overseas. I want to allow him to call a number which will give him a busy signal (so no charge), but will then send me an email that he has called. I know how to use a call file to trigger a call (I created a callback system for myself when I traveled overseas a few months ago), but I don't know how to trigger an email. All I want is a simple message like

Hi All. I've been experimenting with SLA on Asterisk 1.4.13 (patched up to 1.4.14). I am using a SIP channel for my "trunk" line. On the whole things are good, but I have noticed that if I misdial an outgoing call, i.e. I get 404 "Not Found" in the SIP trace, then the trunk line just drops, rather than presenting an error tone or message to the user.

I've got a new setup (different building) where Asterisk is sitting between the PBX and phone company on a E&M T1 line. Mitel PBX <-> Asterisk <-> Phone company Inbound works. Asterisk gets the in-band digits from the phone company and hands the call off to the Mitel just fine. Outbound is weird. Asterisk seems to expect that the mitel will send routing information

Greetings Asterisk users! I'm trying to setup Asterisk system to act as a callback system together with callcentric (http://callcentric.com) but it appears that I hit common DTMF issue and I want to workaround this problem. Basically my current setup is the following: 1) I have dedicated Asterisk server that it is linked to my callcentric account 2) I have US phone number (DID) from

I''m running a site that gets ~30k to 40k page hits per day. In the last 4 days my mongrel processes have been jumping into high CPU usage a couple of times a day to the point where my site becomes unresponsive (database on a diff machine with no load). The only way for me to resolve the problem and reduce load on the machine is to delete my rails cache directory (I have plenty of space

Anyone know of a good dial plan example for call in / call out? I want to be able to call my Asterisk server, auth, and then call out any number. Michael

We tried to give Mongrel a go running our application which has moderate traffic. We got mongrel up and running easy enough but 3 times in a short period(2hours) mongrel just died with the following error when we went live to production with it. As you can see we are running mongrel-0.3.13.3. /usr/lib/ruby/gems/1.8/gems/mongrel-0.3.13.3/lib/mongrel.rb:576:in`peeraddr'': Transport endpoint

Admittedly this is probably pretty stupid of me, but there are just some things I can't understand by reading documentation. Any suggestions or recommendations about how to handle my problem are greatly appreciated. I'm trying to achieve the same functionality as my Nortel PBX, without rewriting much 'C' code. In my dialplan I'd like to compare two variables as a means of

hello- i was messing around with a did from ipkall.com, and asterisk seems to be ignoring the context specified in the sip config. in sip.conf, i've added: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = "ipkall incoming" <7508> nat = no in extensions,conf, i have: [remote] exten => 7508,1,DISA(1111|internal) [internal] exten =>

Our telco has setup toll access account codes for outgoing calls. I would like to include these account codes in the dialplan for certain extensions (fax lines, modems) so that they are not prompted for the 4 digit code when making a toll call. I have played around with the 'w' command with ZAP channels, commas, and DISA, all with no success. I see a lot of examples of Asterisk with