[asterisk-users] PJSIP configuration question

[Dan Cropp]

I should mention, I am actually sending this via AMI in both the chan_sip and the pjsip case.

Pjsip originate...

Action: Originate
ActionID: S8
Channel: PJSIP/outbound.vitelity.net/1234567890
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 60000
CallerID: Dan Cropp<1234>
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true


Chan_sip based originate...

Action: Originate
ActionID: S8
Channel: SIP/outbound.vitelity.net/1234567890
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 60000
CallerID: Dan Cropp<1234>
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true

Have a great day!

Dan

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Cropp
Sent: Wednesday, December 10, 2014 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question

Thank you for the speedy reply.

My originate string is something like the following where xxxxx is really the sip provider's supplied IP address
1234567890 is really the phone number I am dialing

PJSIP/outbound.vitelity.net/1234567890

In the chan_sip based solution, it's...
SIP/outbound.vitelity.net/1234567890

Have a great day!

Dan

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, December 10, 2014 9:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question

Kia ora,

Dan Cropp wrote:
> I'm working with a SIP provider to try and transition our sip 
> connection with them to PJSIP. I thought I had transitioned the 
> settings correctly, but whenever I attempt an Originate it never even 
> tries to send any PJSIP messages.

What dial string are you providing to Originate?

> I'm currently running Asterisk 13.0.0.
>
> Anyone have any suggestions as to what I am doing wrong?
>
> The SIP provider says the latest version of Asterisk they have anyone 
> using is Asterisk 11, so they have no PJSIP configuration experience.
>
> The only setting that I believe I haven't found a PJSIP settting for 
> is the "insecure=invite" from sip.conf

That functionality exists in the form of the "identify" object. It does IP based matching of incoming traffic and to associate it with an endpoint.

>
> I thought that would be the equivalent of no authentication object, so 
> I tried that. However, that did not work either.

Authentication controls authentication, it doesn't control how PJSIP associates traffic with a specific endpoint. They are separate things.

I think before we get into config we need to see the dial string for your origination.

Cheers,

--
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

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