similar to: dial plan and sip

Hi All, I'm having an issue where Asterisk continuously sends out a GAZILLION "SIP NOTIFY" messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf resolves this and everything works fine. I'd like to confirm the layout of the

Hello, I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The backend is a MySQL database running through the ODBC backend in Asterisk. At this point everything works in terms of phones registering, placing calls between them, etc. However, I am having a problem setting the Caller ID number whenever I am using the Realtime database for the SIP users/peers. If I use a

why does this not work? i simply want to hear the recorded message exten => s,1,Answer() ;exten => s,n,Record(zipcodegutter1.gsm) ;zcg1 exten => s,n,Playback(zipcodegutter1) exten => s,n,Dial(SIP/c000001s/12222222259,120,A,(demo-thanks))

Are there any internal DHCP or DNS services built-in to the Asterisk code? -- -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110427/2a4bff5d/attachment.htm>

Does anyone know how to send a text message from Asterisk?

This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses but I want to know in any case! Can a vb script run somehow on a Linux machine or does it only work on Windows? If I were to build a call file script (described in this link http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out ) then how does it work if my Asterisk machine is running on Centos 5.5? I simply

Is it possible to have Calls incoming to different DIDs? I want an AA that handles 100s of businesses. [Incoming-pizza] Exten => 4045551212,1,Goto(pizza,s,1) [Incoming-hvac] Exten => 8085551212,1,Goto(hvac,s,1) [Incoming-gutter] Exten => 6175551212,1,Goto(gutter,s,1)

DID number A. I have a DID (a regular line from Verizon). number A. Can I have A ported to my SIP provider? Then, interface the A DID to my system so that I can build a solution. I want to write an IVR for the A number and allow callers dialing A to interact with my Asterisk machine. I need to keep number A. Kindly advise

I read that I need to run 1.6.2.6 (at least 1.6) to use the MeetMe application since I don't have a zdummy timing driver. Anyway, I want to upgrade my machine to 1.6.2.6. Does anyone have the exact steps? I see a lot of references on the web but any other links from our community may be preferred! Thank you Tom

i am running 1.4.37 and am hosted on Rackspace. I feel like a took a step back by using the Cloud server service since I am having a little trouble proving that my basic configuration is working. Nevertheless, I want to upgrade to 1.8. I use Centos 5.5 Anyone know of a good link that can help please? I searched Google and got confused by the options. Upgrade to 1.8. How please?

Hi all, I am planning to do clustering for my company's asterisk servers. I dont know much about it, just read some articles on the internet and learned how to use DUNDi and some basic information about clustering. What I need to know is: 1. can i register end user with multiple asterisk servers at a time? 2. If not, Can I re-route registeration requests to different servers using 1 asterisk

Dear Philipp; Kindly find the part of the configuration as below: [general] allow=all disallow is comment by ( ; ). [bilal_sip] type=friend context=internal host=dynamic canreinvite=no dtmfmode=rfc2833 So where is the problem? The endpoint does not register and nothing appear on trace level 3. And the amazing thing that if the endpoint send wrong username (for example: bilal_sip100) then it

Where is everyone located? I am in Washington DC. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091106/7c73847d/attachment.htm

Hi List; I noticed that if I disabled secret in the context by placing ( ; ) before it, then at the asterisk the log will be: -- Registered SIP 'bilal_sip' at 0.0.0.0 port 5060 expired The IP address of the endpoint was not captured!!! Why? If secret enabled, then some endpoints can not register (maybe due to compatibility in reading the negotiation packets), so what is the solution?

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

Does anyone have a Dial script or a hint on how I can dial 10000 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me.

Dear R users, I have a function (below) which encompasses several tests. However, when I run it, only the output of the last test is displayed. How can I ensure that the function root(var) will run and display the output from all tests, and not just the last one? Thank you, b. root <- function(var) { #---Phillips-Perron PP.test(var, lshort = TRUE) PP.test(var, lshort = FALSE)

I want to create a script for IVR that compiles responses, aggregates them to a total number. Then, run an equation based on the result. Press 1 for X (X is a positive number 500) Press 2 for Y (Y is a positive number 200) Press 3 for Z (Z is a positive number 300) Press 20 to calculate the results = 500+200+300 =1000 then, exten => s,n,Read(NUMBER,,1000) exten => s,n,SayDigits(${NUMBER})

So, https://bugs.llvm.org/show_bug.cgi?id=32056 is an example showing our current TBAA tree for union generation is definitely irretrievably broken. I'll be honest here. I'm pretty sure your proposal doesn't go far enough. But truthfully, I would rather see us come closer to a representation we know works, which is GCC's. Let me try to simplify what you are suggesting, and what we

Hi I was using the delivered Ubuntu 1.6.x packages but I wanted to look at gtalk integration so I downloaded, compiled and installed the source (after removing the Ubuntu packages) have installed the following: asterisk-1.8.0 dahdi-linux-complete-2.4.0+2.4.0 libpri-1.4.11.5 I copied my config back into place and most seems to work, but I cannot get my phone that is plugged into the Wildcard