[asterisk-users] PJSIP configuration question

[Dan Cropp]

Thanks George.

That was the ip address I was given.  Unfortunately, my contact at Vitelity is gone for the day so I can’t verify it with him.

I added the qualify_frequency as you suggested and it does appear that I have something configured incorrectly….

<--- Transmitting SIP request (483 bytes) to UDP:0.0.19.196:5060 --->
OPTIONS sip:64.2.142.93 at 5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xx.xxx:5060;rport;branch=z9hG4bKPjcea63914-b8d1-483d-96db-11968abab704
From: <sip:e31d5809-f26a-4219-8365-70931428072b at xxx.xxx.xx.xxx>;tag=7cfab3ba-73de-4243-9967-d1e6a5e7b0b4
To: <sip:64.2.142.93 at 5060>
Contact: <sip:e31d5809-f26a-4219-8365-70931428072b at xxx.xxx.xx.xxx:5060>
Call-ID: 7ba766bf-363b-47d0-a388-62a58d1df88d
CSeq: 33778 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Length:  0


[Dec 17 19:22:31] WARNING[49476]: pjsip:0 <?>:    tsx0x3c501e8 .Failed to send Request msg OPTIONS/cseq=33778 (tdta0x32c7c90)! err=120022 (Invalid argument)
[Dec 17 19:22:31] ERROR[49476]: res_pjsip.c:2532 endpt_send_request: Error 120022 'Invalid argument' sending OPTIONS request to endpoint <unknown>


The 64.2.142.93 is the exact value I was given to use for the outbound trunk (works with sip.conf)
host=64.2.142.93
Any thoughts?
I was really hoping they had worked with the PJSIP, but apparently the latest Asterisk version any of their customers are using is Asterisk 11.

Have a great day!

Dan

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of George Joseph
Sent: Wednesday, December 10, 2014 2:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question


On Wed, Dec 10, 2014 at 1:27 PM, Dan Cropp <dan at amtelco.com<mailto:dan at amtelco.com>> wrote:
Not sure why, but Vitelity changed the settings to IP based authentication on me.  Here's the new sip.conf settings they sent me.

type=friend
dtmfmode=auto
host=64.2.142.93
allow=all
nat=yes
canreinvite=no
trustrpid=yes
sendrpid=yes

When I use these settings to originate calls using the sip.conf they sent me, everything works.

Action: Originate
ActionID: S8
Channel: SIP/outbound.vitelity.net/8005555555<http://outbound.vitelity.net/8005555555>
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 60000
CallerID: John Doe <1234>
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true


I translated those settings to the following for pjsip.conf...

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

[outbound.vitelity.net<http://outbound.vitelity.net>]
type = aor
remove_existing = yes
contact = sip:64.2.142.93 at 5060

You might want to set a qualify_frequency here  to see if the server responds to OPTIONS messages.  Also 64.2.142.93 isn't currently one of their outbound servers.  Are you using one of their inbound* servers as outbound?  IIRC unless you ask them, they don't allow it.

[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
allow = all
direct_media = no

[identify1]
type = identify
endpoint = outbound.vitelity.net<http://outbound.vitelity.net>
match = 64.2.142.93

When I attempt to use AMI Originate, it's failing.  I am not seeing anything with pjsip logging turned on, so it seems to be something with the settings.

Action: Originate
ActionID: S8
Channel: PJSIP/outbound.vitelity.net/8005555555<http://outbound.vitelity.net/8005555555>
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 60000
CallerID: John Doe <1234>
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true

NOTE: I am able to use AMI Originate to other PJSIP endpoints.

Action: Originate
ActionID: S9
Channel: PJSIP/1003/1003
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 60000
CallerID: John Doe <1234>
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true

Anyone have any suggestions as to what I am doing wrong?

Have a great day!

Dan

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