similar to: Asterisk voicemail from mysql no longer working

Hi, i have a problem with iax2 and ringtone. Here is the call path pstn -> asterisk -> iax -> firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really

Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom --------------

Hi, Has anyone done any AGI scripting in pascal. I would appreciate help anyone can offer. My understandin on AGI scripting is very flaky, I am assuming whatever language is used the application needs to be compile and made executable. So if I write a script in pascal, I would compile it with something like freepascal and make it executable. Thanks Umar Sear

Hi Guys. I've had a lay off from Asterisk for 12 months but I am starting to look into it again. I am not very Linux savvy and found it hard going the last time. I've started playing with it in the last 3 weeks and I have to admit to making more head way this time. The first problem I'm stuck on and I cant find a solution to is that sound files that I have recorded (be it by

Hi: Is DynExtebDB module still working?? -- JO

Hi I am trying to use SetCDRUserField in an agi script but with no success. I am using the CDR mysql addon, however I can't see it being at fault as my attempt is not doing anything to the CVS CD either. has anyone used this, any hints guidence would be greatly appreciated. The syntax I am using is like so .. res=DoExec('SetCDRUserField','12345'); and then dialing the

Hi List, Forgive me if this has already been covered. I did go through past messages but could not find anything. I want to setup a queue like scenario where users don't need to login/logout. Basically I want to define a list of extensions that will be rung when a call comes in. The sequence in which the extension are rung needs to be intelligent, in the way queues are. For example, it

How do you prepend. I want to be able to dial 7 digits instead of of 11 for local calls. Can someone post there extensions.conf part that is relavent?

I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki mailboxnumber@context ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a

Hi I've need to reinstall my asterisk software (hard drive failure). I'm back and running to a make samples state. I have backed up all of my conf files (ok so they were about a week old but much better than starting from scratch), the problem I am having is with WS_FTP Pro. Basically I used to connect to my asterisk server using this software no problems just using root as

Does anyone know if the E1 cards that digium sells work in India. Also are there any distributers for those cards in India. By E1 cards I mean E100P, TE410P or TE405P -- regards Vikram (http://www.vicramresearch.com)

Hi, I am trying to implement a dialplan in which the user is notified of a missed call, if no voicemail is left. Basically what I would like to achieve is something like ... exten => _0207XXXXXXX,1,DIAL(SIP/${EXTEN},15) exten => _0207XXXXXXX,2,HasNewVoicemail(${EXTEN:4}@default:INBOX|msgcount) exten => _0207XXXXXXX,3,Voicemail(u${EXTEN:4}) exten =>

Hi I have looked through the wiki and search the mailing list, but I cannot find a way to intrude on a call, can asterisk do this feature? if so how? Thanks for your help Robb

Hi Guys, I am having trouble trying to get format_mp3 working to play music on hold. I have followed the instructions in the read-me and the wiki however it seems after un-installing mpg123, asterisk is not even attempting to play MOH. My musiconhold.conf is ; Music on hold class definitions ; [classes] [moh_files] default = >/var/lib/asterisk/moh-native ;default =>

One of the connections my asterisk PBX has is an analog extension from a Comdial hybrid. On the Comdial system, message waiting is turned on by dialing *3 and then the station number. It is turned off by dialing #3 and the station number. I was wanting to have Asterisk (or Comedian mail) set the message lamp in the Comdial system when a new message arrives for a user, and extinguish the lamp

Hi all, Is there a way to get the "called id" (the B number) with AGI perl ? I know how to get the caller id which is working fine and is just below: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $callerid = $input{'callerid'}; $AGI->say_digits($callerid); } Thanks in advance, Angel.

Hi all, I am trying to write an application to monitor queues using the Asterisk Management API. So far I have had some level of sucess, basically reverse engineering the protocol and the event messages using ethereal etc. I know there are a couple of pages on the Wiki that attempt (no dis-respect to who ever did it as it has been a great help) to document the API and was wondering if there

Checkout http://www.voip-info.org/wiki-Asterisk+variables I believe that should have the answer for you. furthermore assuming that your number is always going to be 12 digits. exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number. Hope this helps. Umar On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote: > Hi, > > this

Hi, i have succesfully configured Xen 3.0.4 with 2.6.16 kernel on some C2D hw with currently 2GB RAM. Only hitch was with the integrated Intel Gigabit network card which is not supported by by default xen kernel - so there is e1000 module compiled from intel sources. Server is connected on 100Mbit ethernet line. Network setting is currently bridged. At the moment 4 domUs with various linux PV

Hi, Has anyone manage to comile app_dbodbc or ast_data with the latest stable release (1.09). If so can you give some guidence on howto do it as I have trouble getting either working. Umar