[asterisk-users] PJSIP configuration question

Dan Cropp dan at amtelco.com
Mon Dec 15 16:33:36 CST 2014

Hi George,

Thank you for looking into this.
This is behind a nat…

type = global
debug = yes

type = transport
bind =
protocol = udp

type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:

type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.net
dtmf_mode = rfc4733
force_rport = yes
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
trust_id_inbound = yes
disallow = all
allow = ulaw
direct_media = no

type = identify
endpoint = outbound.vitelity.net
match =

Have a great day!


From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of George Joseph
Sent: Monday, December 15, 2014 3:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question

On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote:
Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0.

Same problem is happening with both of them.

Could this be caused by PJPROJECT 2.3?

Anyone have any suggestions for what I can try?

My boss is giving me until tomorrow to get the PJSIP support working with Vitelity.  Otherwise, he’s told me to go back to using chan_sip and wait a year or two for PJSIP to be in the field more.

I have a Vitelity account I can try.  Re-post your pjsip config and I'll try it now.

Have a great day!


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