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A second (virtual) number and *Need help or have questions about BroadVoice? BroadVoice is here to answer your questions and concerns: technical support, how-to guides, troubleshooting, and general assistance. |
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| I have my Broadvoice service connected to an asterisk server and it is working fine (both in and outbound). Now I added the second, "virtual" number, but would like it to be routed to a different extension. I read here that I can setup DID routes, so I just tried adding another "register" statement. As you can probably guess - it did not work - I was getting registration timeouts. Here is the register line (working for my first number) register => 678XXXXXXX@sip.broadvoice.com:dontte...broadvoice.com/101 I tried adding this to sip.conf (ext 102 is defined in extensions.conf) register => 678YYYYYYY@sip.broadvoice.com:dontte...broadvoice.com/102 Any ideas what I might be doing wrong? Thanks |
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| You are not doing anything wrong. What you describe requires the use of a brand new feature of Asterisk. When you have more than one BroadVoice phone number you can elect (on your Account Portal) to invoke Distinctive Ring for these numbers. BV then, for your second and third numbers, includes an ALERT_INFO parameter in your Incoming Call SIP packet, which your ATA would decode and ring your phone differently for the different inbound numbers. Asterisk versions through 1.0.9 did not have the ability to trap and decode the ALERT_INFO parameter, so what you describe was incredibly hard to achieve. With the release of version 1.1.0, however, this feature is included in the release. There is a link, http://www.voip-info.org/wiki-Asterisk+cmd+SIPGetHeader, which describes the command you need to fold into your extensions.conf file to trap and process this information. You will need an exten line on the order of exten => s,3,SIPGetHeader(RingTone=ALERT_INFO) to trap the code and then you will need to craft a couple of GotoIf commands to send the different numbers in their different directions. If you are more adventurous you can patch your chan_sip file according to this thread and report your success either here or there.
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| No, I don't know better - I just wish they would pass DNIS - seems like an easy thing to do. I have to report a success here I grabbed a patch, patched chan_sip.c, compiled asterisk 1.0.8 (was using 1.0.7), added configuration as suggested in the thread you pointed me to, captured the "Alert-Info:" header and... it did not work. What was wrong? The thread suggests to set the whole Alert-Info header line: dring1=Alert-Info: <http://127.0.0.1/Bellcore-dr4> When I did that it just would not work. When I changed it to: dring1=<http://127.0.0.1/Bellcore-dr4> It started working and works fine now. Of course, I broke music on hold and voicemail while recompiling, so I have got some more work to do, but at least the second number is routed differently than the first. And yes, I did check that my primary still rings fine Thanks for your help! |
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| I am also frustrated by the fact that, despite the announcement of the imminent release of Asterisk 1.2, I have not been able to locate a copy of Asterisk 1.1 for download. Just as you killed MoH and VM when you compiled the dring patch into Asterisk, I am scared of what I would kill if I went that route.
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I think you can get 1.1 from the tip of the Asterisk CVS. # cd /usr/src # export CVSROOT= # cvs login - the password is anoncvs. # cvs checkout zaptel libpri asterisk They don't have a tag for release 1.1, so 1.1 must be on the tip.[/url] |
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