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Anyone have a cookbook on configuring sendmail to work with Asterisk? Or,a few config examples.

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

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)

I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5" hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? -- Thanks in advance,

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

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

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})

I have Sipura 841's talking to a CVS-HEAD of August 4. If I disconnect the power to the Sipura, Asterisk does not hang up the channel. My sip.conf for this phone looks like: ; [super1] ; Sipura 841 disallow = all allow = ulaw callerid = "super1"

Any reason why I don't get audio on the channel after it rings and the end user picks up. Here are my files. CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus [default] include => stdexten exten => s,1,Answer() exten => s,n,Wait(1) exten => s,n,Dial(SIP/callwithus/1111444444,120,A,(demo-thanks)) exten => s,n,Wait(2)

I want to distribute a random number to each of the first 100 callers to my IVR. This random number will be matched to their telephone number. Where in Asterisk can I do this? And, how? Logic. Call arrives. Context for announcement begins. You will receive a authentication code at the end of the message. Then, if they press a certain digit to confirm then I simply pass a code to them. These

My client wants to use my service that I will host. It is an IVR application. I have the solution worked out on the server side. I will port his current POTS line phone number to a DID service where I can control it via SIP. Question relates to his current phones. Forgive me as I am new. Does he need his current phones? How will they ring if I port the number? Should I simply have him remove

Does anyone have any short answers on how I can fix this problem: A calls B. B rings Says connected. But the call is not bridged and therefor no audio passes. very simple dial plan. Frustrated. v 1.8

I have a very lite layout and attempting to get the SIP configuration set up initially before proceeding into other areas. VMware is running my Asterisk 11 on Ubuntu 12. Shouldnt I be able to at least ping the SIP provider IP? I run command "sip show registry" and do not see it set up. I run sip show peers and I do see an entry. I have not configured anything other then entries in the

Hello, Reading through the Wiki: "Asterisk supports the ability to write dialplan instructions in the Lua programming language. This method can be used as an alternative to or in combination with extensions.conf and/or AEL. PBX lua allows users to use the full power of lua to develop telephony applications using Asterisk" My question is, what is the benefit of using Lua? I recently

I am trying to make sure my DID and SIP account details are working properly and engaging the extensions.conf and dial plan. I have a successful SIP session registered: Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922) Asterisk*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time sip3.voipvoip.com:5060

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.

Zitat von jg <webaccounts173 at jgoettgens.de>: > Yes, it is called "core set verbose 42", the other options is "core > set debug 42". Enjoy the show! OK, thanks, but with this option I can just debug what happens if I call an extension right now... I'd like to have a command to ask Asterisk how it will handle a call... > Once you are more familiar

Hi All, I have installed astriesk 6 and am able to make calls using sip x-lite. Its working as I expected. Now I want to make call from sipx-lite to PSTN using asterisk. can any please suggest me which Hardware card that I can buy? and use ( pl give me all ths list of cards.... which are good.). 2) what is that I need to do after buying the card to make it talk to the real world PSTN network?

Do you see any syntax errors? Positive comments welcomed. The short version of the logic is as follows: create a file based on the NUMBER create a an audio file move to a new extension (label) and play the results exten => 621,1,Answer() exten => 621,n,Read(NUMBER,enteryournumberstartingwithaone,12,,5) ; create a variable from a DTMF entry / 12 characters long exten =>

Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3234 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090226/a46e68fa/attachment.bin