[Asterisk-Users] Incoming SIP connection

Joshua Colp - Asterlink joshnet at nbnet.nb.ca
Sun Oct 16 10:25:05 MST 2005


Hi Joseph,

Here's a basic entry for you that you should be able to adapt.

[mypeer]
Type=peer
Host=ip or hostname
Context=where to send the call
Disallow=all
Allow=ulaw
Insecure=very

The insecure=very causes Asterisk to not do any authentication and trust it
based on the IP.

Joshua Colp

On 10/16/05 1:22 PM, "Joseph Rothstein" <jrothstein at comcentrixs.com> wrote:

> Geetings to all.
> 
> I am having a hell of a time getting incoming SIP connections to work
> properly, and am hoping that someone can help me. Here is what I am using as
> a guide (from the wiki):
> 
> "Incoming SIP Connections
> 
> 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."
> 
> I am mainly concerned with the second point. I want to match an incoming SIP
> connection to a particular IP address.
> 
> I have tried just about everything, and the connection always goes to the
> default context, or the context defined at the top of the sip.conf file. I
> would like to be able to direct incoming SIP connection to a particular set
> of extensions. There is no username and password involved as there will be
> many users coming from this one IP.
> 
> This is what I have tried recently:
> 
> [sipin_test]
> type=peer
> defaultip=195.27.242.120
> context=test_trunk
> deny=0.0.0.0/0.0.0.0
> permit=195.27.242.120/255.255.255.255
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw
> nat=no
> 
> I have also tried changing what is inside the brackets to the IP address. I
> have tried many many different combinations of the above, but the IP address
> never seems to get picked up correctly.
> 
> I am testing the SIP connection using sipsak.
> 
> I realize that Asterisk is probably not the best SIP server to use, and plan
> on migration to SER, but if anyone can offer any suggestions I would really
> appreciate it.
> 
> Regards to all,
> Joe
> 
> 
> 
> 
> 
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