search for: somethingwrong

...finding its always 0. Here's my test code: exten => *244*,1,Dial(Local/test@custom-tests/n,,g) exten => *244*,n,Noop(after dial duration is ${CDR(duration)} billsec is ${CDR(billsec)}) exten => *244*,n,Hangup [custom-tests] exten => test,1,Answer exten => test,n,Playback(tt-somethingwrong) exten => test,n,Hangup The actual CDR record that gets posted in Master.csv looks like so: "","200","*244*","default","""Exten 200"" <200>","SIP/200-94dd","Local/test@custom-tests-0255,1","H...

...You can see the content only by viewing the message source. instead of "- np4e68592849da7 Content-type: text / plain, charset = utf-8 " appear "- np4e68592849da7 Content-type: text / plain, charset = utf-8 " , and that blank line spoil everything. You can check if there is somethingwrong ? -------------- next part -------------- A non-text attachment was scrubbed... Name: adrian_stoica.vcf Type: text/x-vcard Size: 195 bytes Desc: not available URL: <http://dovecot.org/pipermail/dovecot/attachments/20110908/4036437e/attachment-0004.vcf>

...angup() exten => 3,1,Dial(SIP/402,60,rg) exten => 3,n,Hangup exten => 9,n,Hangup() exten => i,1,Set(Loop=$[${Loop}+1]) exten => i,n,Goto(LoopEnd) exten => t,1,Set(Loop=$[${Loop}+1]) exten => t,n,Goto(LoopEnd) [nighttime] exten => s,1,Wait(2); exten => s,n,Playback(tt-somethingwrong); exten => s,n,Hangup; [to_SIPtrunk] exten => _[2-9]XXXXXXX, 1, Macro(dial-trunk-sip,${EXTEN}); exten => _0XXXX, 1, Macro(dial-trunk-sip,${EXTEN}); [incoming] exten => s,1,Noop(); exten => s,n,Verbose(Call ${EXTEN}); exten => s,n,Dial(SIP/501); exten => s,n,Hangup(); [ma...

Hi all. I've been reading through Wi-Ki and at the extensions.conf file description (http://www.voip-info.org/wiki-Asterisk+config+extensions.conf) The author says this: "One day, someone is going to write a proper scripting language for Asterisk that can understand a simpler, easier (and more traditional) scripting syntax. All it would need to do is translate the "high

...llow=gsm dtmfmode=rfc2833 ;; allows use of pushbuttoms nat=no externip=64.4.127.106 localnet=10.0.0.0/255.0.0.0 canreinvite=no extensions.conf [general] autofallthrough=no [default] [testofidea] ;exten => br549,1,Dial(SIP/151,20) exten => br549,1,Answer() exten => br549,2,Background(tt-somethingwrong) If you don't put that, it defaults to, well default! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100420/cc9ffad5/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Anyone have the telemarketer torture prompts? I would seriously like to revive this..... - -- James Finstrom -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH3I8qdloC7YyaIOoRAlAjAJ9Hp+3SS2Z8179HecWIETp4RVDzWQCeMizp fW2JPZdYl/uxG1ziUwYnHGo= =QPbv -----END PGP

I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and converted form SIP to PJSIP using the python script as a start and then mofiying from there.  I ran into an issue when testing that incoming calls from MagicJack would go silent after about 10 seconds.  This happened while in the automated attendant area.  This problem did not occur with Asterisk 13 LTS.  I reverted PJSIP

I am attempting to write in incoming context for calls. 1. If the caller id is given and it is not black listed it will Playback a greeting and then right the phone or go to voicemail under busy or unavailable conditions 2. If no caller id is given, then Privacy Manager will ask for the number. I am testing 6145551212 to see if the black list will work 3. If a caller id is given, and it is

Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using

I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to "incoming" so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten => s,1,Wait(1) exten

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C