BTW, I am the ITSP and looking to test SipX as many of our Customers use it.
Should I create a Jira ticket or just send the snapshot to this list? From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Monday, August 09, 2010 10:54 AM To: Ujjval Karihaloo Cc: Michael Scheidell; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] SIP Trunk Setup questions You are not providing much information. What is the UA? I see the ITSP is sending G722, is that for real? Who is the ITSP? I see they are sending to port 5080, which is good. Is the UA at a.b.c.161:5060? If so, what do the CDR logs show? I would want to see the sipproxy logs at debug for a call like this. A call trace would be REALLY helpful. On Mon, Aug 9, 2010 at 12:39 PM, Ujjval Karihaloo <ujj...@simplesignal.com<mailto:ujj...@simplesignal.com>> wrote: It maybe something else. Theoretically just matching a UserID shud work. I will make the user ID 4 digits later on. For now just want the SIPX to answer. I Know I may be doing something wrong, so please bear with me. To make things even simpler In the SBC Routes, I entered Operator in the Incoming calls Destination field for the sipXbridge....based on step 2 for creating SIP Trunks in the link below. http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configuration#2._Configure_SipXbridge I see the Default "Operator" is defined but I don't hear the welcome WAV file. Could it be looking at the Domain in the Req URI..Here is my INVITE.. Also , where canI lookup SIP/routing logs on the Linux Box where SIPX is installed? INVITE sip:5625551...@a.b.c.20:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP a.b.c.161:5060;branch=z9hG4bKf5ebb9bedc886ec2074709fdc9edafa4 From: "Test"<sip:3035551...@a.b.c.20:5080;user=phone>;tag=0e1f7d3755a944a5117b7dbaa4160484 To: "Sipx"<sip:5625551...@simplesignal.com<mailto:sip%3a5625551...@simplesignal.com>;ssig=06b2f5eb36814eef830167df3b245373> Call-ID: db1771c414fc592e60a3847529c54476-e9f...@a.b.c.161 CSeq: 4217 INVITE Contact: <sip:a.b.c.161:5060;transport=udp> Supported: 100rel Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:multipart/mixed,application/media_control+xml,application/sdp Max-Forwards: 70 Content-Type: application/sdp Content-Length: 407 v=0 o=BroadWorks 82601 8260100 IN IP4 a.b.c.190 s=- c=IN IP4 a.b.c.190 b=AS:448 t=0 0 a=sendrecv m=audio 42114 RTP/AVP 115 102 9 0 8 18 101 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 From: Tony Graziano [mailto:tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net>] Sent: Monday, August 09, 2010 10:25 AM To: Ujjval Karihaloo Cc: Michael Scheidell; sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> Subject: Re: [sipx-users] SIP Trunk Setup questions I don't understand. Your userID DOE NOT need to be identical. Example, user 200 has an ALIAS of 5625551000 Creating users with the same value of a DID really hampers you. Someone leaves the company and someone else needs to get their calls creates way too much work and quite a management headache too. In the VERY ORIGINAL invite, is it 5625551000 or is is 15625551000 or +15625551000. This is important to know. Your ITSp should be able to provide guidance on what format they will send numbers to you in. I have one that sends me 10 digits, but another that sends me full +1(10digits). i would encourage you to keep a simple 3 or 4 digit dialplan to keep you from creating more pain for yourself than you need to. Similarly these users should be numeric only. On Mon, Aug 9, 2010 at 12:14 PM, Ujjval Karihaloo <ujj...@simplesignal.com<mailto:ujj...@simplesignal.com>> wrote: Thx, I am just trying to setup an inbound route to really anything right now to get it to work. I see the INVITE comin with 5625551000 in Req URI ; the user defined has the same userid...and I have call forwarding on that user to go to the Conf Bridge Extension 2000 - for a Conf I created... I see a trying back from SIPX and then nothing... From: sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.org> [mailto:sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.org>] On Behalf Of Michael Scheidell Sent: Monday, August 09, 2010 10:05 AM To: sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> Subject: Re: [sipx-users] SIP Trunk Setup questions On 8/9/10 11:37 AM, Tony Graziano wrote: OR you can assign the DID to a separate Auto Attendant and let people choose the conference they want (1 for sales conf, 2 for management conf, etc.). I set it up so that 'special' users were given their own conference 'rooms', with a two digit prefix(70). (the admin goes in and creates one. leaves it DISABLED) our conference 'DID' calls into a second AA (we use a dialplan that will convert the DID to a user). The conference AA .'extension' is the transformed user. example: DID is 1 561-922-3806, AA 'extension' is 3806. caller hears custom prompt, then dials a 6 digit conference number. (70+users extension) Note: our extension dialplan is 4 digits, so you can't directly dial the 6 digit conference number from the AA. the SECONDARY AA (the conference AA) is set up for 6 digit extensions. you can try it (if you have ISN/sip dialing its free). we have an 'idle' conference at 701259. I'll leave it opened till 3pm today if anyone wants to walk through. dial DID (or sip it, or ISN is: ISN is 3806*1300, sip:confere...@secnap.com<mailto:sip%3aconfere...@secnap.com>) Conference number is 701259. Access code is 8245 (till 3pm) -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > | SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ________________________________ _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net<mailto:tgrazi...@voice.myitdepartment.net> Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net<mailto:helpd...@voice.myitdepartment.net> Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec. -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net<mailto:tgrazi...@voice.myitdepartment.net> Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net<mailto:helpd...@voice.myitdepartment.net> Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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