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@5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xx.xxx:5060;rport;branch=z9hG4bKPjcea63914-b8d1-483d-96db-11968abab704 From: <sip:[email protected]>;tag=7cfab3ba-73de-4243-9967-d1e6a5e7b0b4 To: <sip:64.2.142.93@5060> Contact: <sip:[email protected]: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: [email protected] [mailto:[email protected]] 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 <[email protected]<mailto:[email protected]>> 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@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 -- _____________________________________________________________________ -- 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
-- _____________________________________________________________________ -- 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
