Thanks Tony. I compared a standard call with the TFN and the From Headers are the same. Anyhow, this morning I tested a couple of calls to some TFNs and guess what? They are "automajically" now working and going through. Me thinks Voxitas is reading this thread.. I did a siptrace again on a TFN call and the From Headers that they said that needed to be changed, still looks the same and the call goes through.
Ly Tran From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Wednesday, August 11, 2010 7:02 PM To: Tran, Ly V. Cc: michael.scheid...@secnap.com; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Calling Toll Free #s failing after 4.2.1 Update If your gateway is the same as your other calls, you should compare a standard call with this one. The frame where the call is given to the ITSP (frame 14) says: INVITE sip:+18773669...@netlogic.net <mailto:sip%3a%2b18773669...@netlogic.net> ;user=phone SIP/2.0 Call-ID: 94bc7610-c0a802f0-13c4-a882d-40e060bb-a8...@mydomain.com.0 CSeq: 1 INVITE From: "sipxbridge" <sip:sipfoou...@netlogic.net <mailto:sip%3asipfoou...@netlogic.net> >;tag=5577697804831999119 To: <sip:+18773669...@netlogic.net <mailto:sip%3a%2b18773669...@netlogic.net> ;user=phone> Via: SIP/2.0/UDP 24.153.175.204:5080;branch=z9hG4bKb5bb9d03eae14e51e9018f88d32dfb44393235 Max-Forwards: 70 User-Agent: sipXecs/4.2.1 sipXecs/sipxbridge (Linux) P-Asserted-Identity: <sip:+19724717...@mydomain.com <mailto:sip%3a%2b19724717...@mydomain.com> > Contact: <sip:sipfoou...@24.153.175.204:5080;transport=udp> Route: <sip:209.40.224.172:5060;transport=udp;lr> Session-Expires: 1800;refresher=uac References: 94bc7610-c0a802f0-13c4-a882d-40e060bb-a8...@mydomain.com;rel=chain;sipxe cs-tag=request-invite-z9hg4bk-xx-0070bcx8fsr5qz7dyymfikfxsw Allow: INVITE,BYE,ACK,CANCEL,OPTIONS Supported: timer Content-Type: application/sdp Content-Length: 246 So the correct invite the carrier wants for a TFN is there (+1 and 10 digits): Invite: INVITE sip:+18773669...@netlogic.net <mailto:sip%3a%2b18773669...@netlogic.net> From: "sipxbridge" <sip:sipfoou...@netlogic.net <mailto:sip%3asipfoou...@netlogic.net> > If a regular call shows the same thing then your carrier is toying with you. I suspect it WILL show the same thing. Besides they ACK it and the session tries to establish after that. It ultimately fails because of frame 24: Time: 2010-08-11T22:16:23.552000Z Frame: 24 /tmp/trace.mRQ20821/_.sipxbridge.trace.xml:1369 Source: 209.40.224.172:5060 Dest: sipx.mydomain.com-sipXbridge SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 24.153.175.204:5080;branch=z9hG4bK523b18e8a99e29f0969500b5d1ec1e44393235 ;received=24.153.175.204 From: "sipxbridge" <sip:sipfoou...@netlogic.net <mailto:sip%3asipfoou...@netlogic.net> >;tag=5577697804831999119 To: <sip:+18773669...@netlogic.net <mailto:sip%3a%2b18773669...@netlogic.net> ;user=phone>;tag=as55cb36ec Call-ID: 94bc7610-c0a802f0-13c4-a882d-40e060bb-a8...@mydomain.com.0 CSeq: 2 INVITE User-Agent: NetLogic Switch v3.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY Supported: replaces Contact: <sip:+18773669...@209.40.224.172 <mailto:sip%3a%2b18773669...@209.40.224.172> > Content-Length: 0 that frame is generated by the carrier. Why? I don't buy their response to you. Maybe you can compare a regular call against this one and see if "foouser" in sent and accepted by them. I noticed a lot of issues over the last several months with toll free calls and ITSP's FAILING because of the peering and connection costs in some areas going up. Ultimately I started using PSTN gateways and secondary trunk providers to use JUST for toll free calling. Note: I have never used voxitas, this is an industry problem. Tony On Wed, Aug 11, 2010 at 7:16 PM, Tran, Ly V. <lt...@rrtgi.com> wrote: Attached is a merged file of a failed call to an 877 TFN. I'm no expert at reading these logs with sipviewer. What can anyone tell that may be wrong with my setup? (actual phone number / account username has been changed for privacy). I'm concerned about the "407 Proxy Authentication Required" but normal phone calls are working ok. This TFN number is returning a "503 Service Unavailable" then "500 Internal Server Error" , Reason: ~~id~bridge;cause=213;text="Relayed Error Response" which I don't really know what that means. Voxitas is telling me what is wrong is that the From: Header is showing sipfoouser (my ITSP account username) and it should be displaying the +19724717777 number. I don't know if it was doing this as well before the update to SipX 4.2.1, but we were able to make TFN calls before the update 1 to 2 weeks ago. Thanks in advance! Ly Tran From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Wednesday, August 11, 2010 10:49 AM To: Tran, Ly V. Cc: michael.scheid...@secnap.com; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Calling Toll Free #s failing after 4.2.1 Update Like I said, it could be a voxitas change. If you sign up for a voip.ms account and add a dialplan to send certain calls there, do toll free calls work? does outbbound callerid work? On Wed, Aug 11, 2010 at 11:44 AM, Tran, Ly V. <lt...@rrtgi.com> wrote: Well, tried removing the +1 from the gateway and added to the individual dialplans. Local / LD calls still works, but toll free numbers still doesn't work. It's time for Voxitas to dig deeper to see if they have made any recent changes or something in Sipx from the recent updates. Outgoing caller ID is not working at this point to external phones or cell phone. Ly Tran -----Original Message----- From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Tuesday, August 10, 2010 4:31 PM To: Tran, Ly V.; michael.scheid...@secnap.com Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Calling Toll Free #s failing after 4.2.1 Update Well, the rule to take +1 at the gateway should be removed and +1 should be added to the individual dialplans or if your gateway does not require registration you can create anotgher instance of it and not add +1 and send 10 digit toll free calls to that gateway. Or you can get a voip.ms account just sending toll free calls there. ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: Tran, Ly V. <lt...@rrtgi.com> To: Tony Graziano <tgrazi...@myitdepartment.net>; michael.scheid...@secnap.com <michael.scheid...@secnap.com> Cc: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> Sent: Tue Aug 10 17:13:08 2010 Subject: RE: [sipx-users] Calling Toll Free #s failing after 4.2.1 Update I am using the same gateway with Voxitas for local, LD and tollfree. Because they are using +1, I've set up the dial plan as you suggested to enable dialing from the missed calls on the phone since the incoming caller ID shows +1 as well. I have a basic dial plan of 10 digits, 1 appended then the gateway adds +. So all calls made by the users are 10 digits on the phone (local, LD and TFN). Results in dialing an external and mobile phone using different iterations of the caller ID number on the gateway are quite strange. +1XXXXXXXXXX - no caller id on external; cell phone says forwarded call and my cell number shows as the incoming 1XXXXXXXXXX - caller id shows truncated CompanyName and blank numbers on external phone; cell phone does the same as above XXXXXXXXXX - same as above. I notice that setting Caller ID for user does not over ride the Caller ID set on the gateway anymore as well. The majority of the phones are set to display the main office number as caller ID. A few individuals had their assigned DID set as the caller id, but that's not working now. Looking at the sipproxy.log, the invite is sip:NPANXXYYYY What does the Call ID suppose to look like, mine shows Call-ID: 94bbdfc0-c0a80148-13c4-eeb28-50c9fab8-ee...@mypbx.com Ly Tran From: Tony Graziano Sent: Tue 8/10/2010 2:21 PM To: Tran, Ly V. Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Calling Toll Free #s failing after 4.2.1 Update Well, it would help to know what your gateway is doing and how your dialing rules work. I don't use voxitas so I can't help with anything that might be peculiar there. Is your gateway set to add any digits to "all call" through the gateway? Is you dialplan doing this either? Do you have a separate dialplan for toll free (depending on how your configuration is done, you might not need to). If you tail the sipproxy log can you see what is in the invite you are sending to the gateway? invite sip:1NPANXXYYYY or sip:NPANXXYYYY ?? On Tue, Aug 10, 2010 at 3:14 PM, Tran, Ly V. <lt...@rrtgi.com> wrote: Just noticed that we are unable to make any outbound calls to toll free numbers after this latest update. We were able to on the previous version. We are using Voxitas as the ITSP. Has anyone else seen this or tested TFN dialing after the update? Normal local and long distance phone calls are working. Voxitas tells us that our From URI is incorrect, and we need to provide a valid 10 digit number when dialing out to TFNs. I'm not sure what that means since our default caller id is set to with our valid main office number on the gateway. When we dial a TFN, after about 10s the display on the phone says "disconnected, temporary failure". Ly Tran _______________________________________________ sipx-users mailing list 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 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: 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 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: 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 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: 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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