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 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] type = aor remove_existing = yes contact = sip:64.2.142.93 at 5060 [outbound.vitelity.net] 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 allow = all direct_media = no [identify1] type = identify endpoint = 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 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
On Wed, Dec 10, 2014 at 1:27 PM, Dan Cropp <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 > 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] > 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] > 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 > allow = all > direct_media = no > > [identify1] > type = identify > endpoint = 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 > 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141210/6c8e1285/attachment.html>
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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<snip>> > I translated those settings to the following for pjsip.conf... > > [transport1] > type = transport > bind = 0.0.0.0 > protocol = udp > > [outbound.vitelity.net] > type = aor > remove_existing = yes > contact = sip:64.2.142.93 at 5060This is incorrect. The contact should be: contact = sip:64.2.142.93 It will use a default port of 5060. I also believe I've covered your origination issue in a separate email. Your dial string should be: PJSIP/8005555555 at outbound.vitelity.net 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
Thank you Joshua. I will make the modifications this morning and give it a try. 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 7:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question <snip>> > I translated those settings to the following for pjsip.conf... > > [transport1] > type = transport > bind = 0.0.0.0 > protocol = udp > > [outbound.vitelity.net] > type = aor > remove_existing = yes > contact = sip:64.2.142.93 at 5060This is incorrect. The contact should be: contact = sip:64.2.142.93 It will use a default port of 5060. I also believe I've covered your origination issue in a separate email. Your dial string should be: PJSIP/8005555555 at outbound.vitelity.net 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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users