[Asterisk-Users] Dual Line SIP config to the same provider

Soren Christensen soren at silikone.com
Wed May 24 14:20:13 MST 2006


Hi,

I have a setup where I have multiple lines to the same provider - in 
this case broadvoice.

I have created the entries in sip.conf for both accounts - and 
independently they work fine. When they both are in use, incomming calls 
are placed to the one that's the last in the sip.conf file.

On voip-info I found the following:
*Quote:*

When Asterisk receives an incoming SIP call, the SIP Channel Module

* first tries to find a [user] section matching the caller name
(From: username),
* then tries to find a [peer] section matching the caller's IP address.
* If no matching user or peer is found, the call is sent to the
context defined in the [general] section of sip.conf.



This would imply that one had to split the entry into a inbound and an 
outbound entry ? Did anyone try this and got it to work ?

Is there anybody that has gotten this to work, such that the correct 
context based on the phone number is activated when a call comes in.

My sip.conf structures are:

*Code:*

[broadvoice-1178]
type=friend
host=sip.broadvoice.com
username=<Number>
fromuser=<Number>
authname=<Number>
fromdomain=sip.broadvoice.com
context=1178-incoming
secret=<secret>
canreinvite=no
insecure=very
;dtmfmode=inband
;dtmf=inband
dtmfmode=rfc2833
dtmf=rfc2833
qualify=0

[broadvoice-4633]
type=friend
host=sip.broadvoice.com
username=<Number>
fromuser=<Number>
authname=<Number>
fromdomain=sip.broadvoice.com
context=4633-incoming
secret=<secret>
canreinvite=no
insecure=very
;dtmfmode=inband
;dtmf=inband
dtmfmode=rfc2833
dtmf=rfc2833
qualify=0



Thanks
/S
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