[asterisk-users] PJSIP outbound register and inbound calls

Nick Awesome jleed at me.com
Wed Jul 16 12:13:05 CDT 2014

Ok there is my test account from sipiko.net

username: cb5069
password: sqv664yqtp
domain: callme.sipiko.net

its using username/password authentication.
because its just website widget I need only inbound calls from this peer,
test call can be done from url: 

on my side I have an asterisk 12 using pjsip

Have configured IVR with number 5000 on context "dialmap", so I need forward all calls from this provider to number 5000 over "dialmap" context

help if you can please:)

On Jul 16, 2014, at 8:53 PM, Joshua Colp <jcolp at digium.com> wrote:

> Nick Awesome wrote:
>> I thought that
>>>> type=identify
>> will match an IP address and accept it,
>> well, in my example I can control both sides and able to configure it
>> without registration. in real life I have a provider that requires
>> username/password authentication
>> provider gives me - Username - Password - DomainName
> They may require it for *outgoing* calls to them but for incoming I
> highly doubt they'd want you to authenticate them. It's usually always
> IP authentication.
>> I have configure it like I showed before and have exactly the same
>> notice
>> [Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246
>> log_unidentified_request: Request from
>> '"cb5069"<sip:asterisk at>' failed for
>> '' (callid:
>> 173995aa2e25283807700d65055c9214 at - No matching
>> endpoint found is an operator,
>> so what I should add to my config to be able accept calls from
>> Registered peer ?
> The PJSIP functionality does not currently allow using the dynamic IP of a registration to match an incoming call. You either have to explicitly use the identify section or match as I previously described.
> Without further details of your setup (IP addresses, who are calling who) and how you want it to work I can't answer.
> -- 
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
> -- 
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>              http://www.asterisk.org/hello
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140716/5790aced/attachment.html>

More information about the asterisk-users mailing list