similar to: Prepending for 9NxxNxxx - adding the area code for 7 digit dialing

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 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

What I am trying to do is in the 3rd option dial my cell# thru voicepulse I just can't figure how to construct the line [inevans] exten => s,1,setcallerid(${CALLERID}) exten => s,2,Dial(MGCP/aaln/1@Egraph-1,10,tr) exten => s,3,Dial(iax2/passwod@voicepulse.com/ Where do I put the # to dial 18708573287 thanks James Schenck Egraph Design Inc. Arkansas Online Internet Services (870)

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 All, I hope someone can help. I have a system that I have recently upgraded to latest CVS and my voicemail is not working from mysql database. I get an error on the console saying " No entry in voicemail config file for 'number'" whilst there is an entry in the database for the specified number. It seems like app_voicemail is no longer checking the database even though

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 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

Until recently, I was using Broadvoice for my in/out calling thru Asterisk. I was extremely pleased to see that Broadvoice was actually passing the callerid info (number and text) that I had set up on each device in my SIP.CONF file. I had PSTN users tell me that they were actually seeing name and extension info when I called them from the Asterisk box. Last week, due to numerous user quality

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 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

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, 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 --------------

I have an inbound and outbound account with Voicepulse (I am very happy with the service, btw). But I have a weird IAX2 problem. When I get a inbound call on my Voicepulse DID, the call hits my asterisk server correctly with the correct callerid (the DID phone number 617902xxxx). when the call gets passed on to a softphone (X-lite), the caller id that shows up on the X-lite softphone as Lee ,

I cannot find any means of passing my own Callerid using Voxee. It always comes across as NO ID, or nothing, or unknown. I could not find anything on their website about setting your own caller id in the system either. (their web account pages). Is anyone here using their own Callerid information through Voxee? thanks

I have been picking at Asterisk for about a week, and I think I'm close. I was hoping for a little guidance to bring this on home. I want to be able to make outgoing calls from my SJPhone clients using my VoicePulse Connect account. I have the two requisite items from Voice Pulse, but I've had no luck successfully integrating the VoicePulse settings into iax.conf. My current config:

Hey I just setup service with IPComms and they are telling me to setup such as this: iax.conf: [IPCommsNet] type=user host=69.15.xxx.xx context=voicepulse-in ;(changed by me) nat=yes dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=gsm When I'm calling once of my numbers it's giving me this though: Oct 5 12:11:06 NOTICE[49584]: chan_iax2.c:5476 socket_read: Rejected connect

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.

Is any one else having problems with Voicepulse today? Suddenly, I can't register and calls to my Voicepulse numbers get a fast busy. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815