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  #1 (permalink)  
Old June 13th, 2005, 08:40 PM
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fatcat
Default Multiple Broadvoice DIDs on one account and Asterisk

I have 1 Primary number and 2 alternate numbers with my Broadvoice account, is there any way to map these 3 numbers with 3 different extensions one for each extensions
Like , xxxxxxx will ring ext x
yyyyyyy will ring ext y
and zzzzzzzz will ring ext z

I tried placing 3 entires in extensions.conf under [bvin]

ext => xxxxxxxxxx,1,Dial(SIP/100)
ext => yyyyyyyyyy,1,Dial(SIP/200)

But call is not going to any extension just got Called party is Unreachable IVR


My Sip.conf settings

register => 1234567880:bbbbbbbb@sip.broadvoice.com

[Broadvoice]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=1234567890
secret=bbbbbbbbb
username=1234567890
insecure=very
authname=1234567890
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
context=bvin
disallow=all
allow=ulaw
allow=alaw



[100]
username=100
type=friend
secret=password
nat=never
mailbox=100@default
host=dynamic
dtmfmode=rfc2833
context=bvout
canreinvite=no


[200]
username=200
type=friend
secret=password
port=5060
nat=never
mailbox=200@default
host=dynamic
dtmfmode=rfc2833
context=bvout
canreinvite=no
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Old June 13th, 2005, 09:22 PM
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Default RE: Multiple Broadvoice DIDs on one account and Asterisk

I'm going to give you some suggestions and make a request of you. First, the request. If you get this working, please post a sanitized version of your successful configuration (like you did here), because I would love to do the same thing, just haven't gotten around to figuring it out yet.

Now, the suggestions. I understand that in your Account Portal to use a Distinctive Ring for each of your alternate numbers. There appears also to be a way for Asterisk to capture this Distinctive Ring tag and act upon it. Putting these two bits together, you should be able to send the inbound call in different directions based upon which Ring is being requested, much like a modern day Party Line.
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Old June 14th, 2005, 06:55 AM
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fatcat
Default

Thanks for your suggestion
AS far as i understand Distinctive ring pattern matching works only with zap hardware i didnt saw/discoverd any appplication or method which do ring detection with SIP. Might be there is a method some one might help on this, or there may be some AGI to do this.
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Old June 14th, 2005, 03:25 PM
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What I read about (I don't remember where) is talk about an Alert-Info packet coming in with the incoming call. This Alert-Info packet tells the ATA which Alert (ring pattern) to use when ringing the attached phone. It seems to me that we should be able to get Asterisk to look at this Alert-Info information and make a routing decision based upon its contents. I also don't remember reading of anyone who has actually succeeded, though.
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Old July 28th, 2005, 08:14 PM
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fatcat
Default

Finally i got some time to get some more data about it
The dring header that comes from BV is
Alert-Info: <http://127.0.0.1/Bellcore-dr3>

Now the configuration which can be done is (got after googling)

[peername]
..
..
dring1=Alert-Info: <http://127.0.0.1/Bellcore-dr3>
dring1context=bvoice-dring
..
..

I have seen some comments on it web that they suceesfully did it with broadvoice. I tried it but unable to get the alternate number in bvoice-dring context


The other thing i've seen is asterisk cmd sipgetheader()
Which is defined as
With this app, you can pick any header from an incoming invite and
stuff it into a channel variable. It is a generic way of supporting any header
a vendor or service provider may add that you want to use in your dial
plan.


I never played with variables in asterisk and only a average user of asterisk.
So any ASterisk GURU can help out by setting a variable if get dring header and then route call according to it

Is there any way to it ? ?

I am also looking in to it but have very few little for these experiments
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Old July 28th, 2005, 08:17 PM
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fatcat
Default

forgot to include refrences for my points

http://bugs.digium.com/bug_view_page.php?bug_id=0002838

http://bugs.digium.com/view.php?id=2924
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Old July 28th, 2005, 10:14 PM
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fatcat
Default

Finally I am sucessfull routing the alternate number to another context based on dring string.

I patched my chan_sip.c with the bugid = 0002924
Added

dring1=Alert-Info: <http://127.0.0.1/Bellcore-dr3>
dring1context=bvoice-dring

in my broadvoice peer


[Broadvoice]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=1234567890
secret=bbbbbbbbb
username=1234567890
insecure=very
authname=1234567890
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
context=bvin
disallow=all
allow=ulaw
allow=alaw
dring1=Alert-Info: <http://127.0.0.1/Bellcore-dr3>
dring1context=bvoice-dring


and my call to virtual number goes to bvoice-dring context and call to primary number is going to bvin


Thankyou mberlant to show me the right direction
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Old July 29th, 2005, 08:41 PM
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Thank you, fatcat, for making this work. My job now is to figure out how to apply the patch to chan_sip.c and get this working on my Asterisk. Thanks again.
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Old August 8th, 2005, 04:44 AM
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jshaffer
Default

I am new to asterisk and using it as a Gateway for my Cisco CallManager. I was wondering how to route the call just by sending it to a different context since it will show up as coming on the same line as the primary? Is there anyway you could post a snippet of your ext.conf? I would be very helpful.

Many Thx....
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Old August 9th, 2005, 08:37 PM
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lionvision
Default FatCat.. please help

Can you post the process of patching chan_sip?

im using AAH 1.3 (asterisk 1.0.9) and am a little lost at the point of patching (and the patch says its for 1.0.2)


Any help would be hugely appreciated as ive been trying to get the alternate number thing to work for a while.

thnks.
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