[asterisk-users] pjsip.conf question

[Dan Cropp]

Thank you Joshua.

That did in fact solve the problem I was seeing.

I am now experiencing another issue.  The 3rd party sends their messages from various port numbers.
However they only read messages sent to them on port 5060.

For example, right now I receive the first INVITE with port 1234.
Asterisk sends the Trying to port 1234
Third party doesn't monitor this port so it eventually times out and sends another INVITE, this time with port 1235.
Asterisk sends the Trying to port 1235
Third party doesn't monitor this port so it eventually times out and sends another INVITE, this time with port 1236
Asterisk sends the Trying to port 1236

Would this be achieved via the transport external_signaling_port setting?

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: Tuesday, July 14, 2015 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] pjsip.conf question

Dan Cropp wrote:

<snip>

>
> My pjsip.conf looks like...
>
> [transport1]
>
> type = transport
>
> bind = 0.0.0.0
>
> protocol = udp
>
> [3400]
>
> type = aor
>
> max_contacts = 1
>
> remove_existing = yes
>
> contact=sip:xxx.xxx.xxx.xxx
>
> [3400]
>
> type = endpoint
>
> context = DEF
>
> transport = transport1
>
> aors = 3400
>
> accountcode = 1
>
> dtmf_mode = inband
>
> device_state_busy_at = 32

chan_pjsip has few defaults. As a result the above endpoint has no codecs configured. Therefore the SDP negotiation fails. Adding "allow=ulaw" should allow it to work.

--
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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