Soren Christensen
2006-May-24 14:20 UTC
[Asterisk-Users] Dual Line SIP config to the same provider
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060524/36b10d5b/attachment.htm