[asterisk-users] PJSIP configuration question

[Dan Cropp]

Yes, everything is behind the same NAT.

For the application I’m working on, the only endpoint is the endpoint to Vitelity.
We use AMI to Originate calls from Asterisk endpoint through Vitelity to phones.
After that, we control the call through AMI to perform the work we need.



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 4:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question



On Mon, Dec 15, 2014 at 3:33 PM, Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote:
Hi George,

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


Just to be clear...both the pbx and local endpoints are behind the same NAT?


[global]
type = global
debug = yes

[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[outbound.vitelity.net<http://outbound.vitelity.net>]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:64.2.142.93

[outbound.vitelity.net<http://outbound.vitelity.net>]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.net<http://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

[outbound.vitelity.net<http://outbound.vitelity.net>]
type = identify
endpoint = outbound.vitelity.net<http://outbound.vitelity.net>
match = 64.2.142.93

Have a great day!

Dan

From: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com> [mailto: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!

Dan

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