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How can I let asterisk immediately dials a trunk when off hook?

Anyone can recommend a commercial large scale IVR with easy + pro management for asterisk? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091124/53055c0c/attachment.htm

Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source & destination? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/48076c6b/attachment.htm

When Asterisk establish a call through an outbound trunk, Is there any way I can know who hang up the call first? The caller or the party called? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091022/c8beaabb/attachment.htm

Hi I have asterisk 1.4 I want to make a MGCP trunk as a client to connect to a provider who is using MGCP protocol, he provided me with user & password, I tried a custom trunk: MGCP/$OUTNUM$@user:password at 66.152.163.106:4000 Not seems to help, Any suggestions plz?

How can I mute the message "please enter the number you wish to call and press the # key" in a2billing??? I tried use_dnid = YES but still I keep getting the message prompt... thanks

Not sure if a2billing can be shared here, but ill give a shot If the credit < min_credit the IVR play: sorry you have 0 credit and hangup, I want it to FW me to the IVR to add voucher, please let me know: here is log: [18/10/2010 07:01:12]:[file:a2billing.php - line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1] [18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE : standard]

Was my question not understood? Hello, I would like to connect 2 asterisk boxes together, so this is my scenario: Asterisk Main: it is connected to many sip providers and its main purpose as a call termination forwarder. Asterisk B: it?s connected to E1, and its purpose to terminate calls. It will receive SIP messages from Asterisk_Main, but there will be no voice traffic

Hello, How can I reset CDR time , let's say after 30 seconds of answer signal, reset CDR to 0 , any idea ?? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090909/8c47a03a/attachment-0001.htm

Hi After a day of running asterisk, I got choppy sound when fw ip->pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091009/be8bbf2f/attachment.htm

Hello all, Is there anyway that I can configure Asterisk to start dialing out from fxo after (xx) seconds from getting the dial tone? I don't want tdm card to send the number immediately because it fails many times. Thanks for any help. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am not sure why I am getting this message, I have an outbound route that goes to asterisk gateway1 then asterisk gateway2 When all lines on asterisk gateway1 are full, I get the message " all our circuits are busy now" then few second later, the phone rings, going to the second route! And the call can be established, how can I get rid of this message?? thanks --------------

Hello, I have : answeronpolarityswitch=yes on chan_dahdi.conf but it's making all my lines answer on polarity reversal, this causes a problem for PSTN lines, so how can I set these lines to answer immediately (when it rings)? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone I would like to know how possibly to make an announcement with asterisk. i.e. I have a phone registered with extension 100 when I dial (for example) 500, then all the phones with extension 501,502,.510 will automatically answers my call, and I speak my announcement. Any help in the issue very appreciated. -------------- next part -------------- An HTML attachment was

Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/af1f9656/attachment.htm

Hello, I would like to know how Asterisk deal in this case: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk

Hello, Sometimes the call gets answered for 1 second, but actually the phone has not rang, it?s just the CDR, and asterisk hangup automatically, I cought the log of 1 call like this, I hope you can help me with this. My setup is : <vendor> ----SIP--? <Asterisk> ?----IAX2---? <Asterisk with Dhadi channels> Here: -- Executing [966505103150 at from-internal:1]

I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. -------------- next

Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI> -- Hungup 'IAX2/99999-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on