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:[email protected] <sip%3a%[email protected]>;user=phone SIP/2.0 Call-ID: [email protected] CSeq: 1 INVITE From: "sipxbridge" <sip:[email protected]<sip%[email protected]> >;tag=5577697804831999119 To: <sip:[email protected] <sip%3a%[email protected]> ;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:[email protected]<sip%3a%[email protected]> > Contact: <sip:[email protected]:5080;transport=udp> Route: <sip:209.40.224.172:5060;transport=udp;lr> Session-Expires: 1800;refresher=uac References: [email protected] ;rel=chain;sipxecs-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:[email protected]<sip%3a%[email protected]> From: "sipxbridge" <sip:[email protected]<sip%[email protected]> > 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:[email protected]<sip%[email protected]> >;tag=5577697804831999119 To: <sip:[email protected] <sip%3a%[email protected]> ;user=phone>;tag=as55cb36ec Call-ID: [email protected] CSeq: 2 INVITE User-Agent: NetLogic Switch v3.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY Supported: replaces Contact: <sip:[email protected]<sip%3a%[email protected]> > 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. <[email protected]> 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:[email protected]] > *Sent:* Wednesday, August 11, 2010 10:49 AM > *To:* Tran, Ly V. > *Cc:* [email protected]; [email protected] > > *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.msaccount 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. <[email protected]> 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:[email protected]] > Sent: Tuesday, August 10, 2010 4:31 PM > To: Tran, Ly V.; [email protected] > Cc: [email protected] > > 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: [email protected] > > 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. <[email protected]> > To: Tony Graziano <[email protected]>; > [email protected] <[email protected]> > Cc: [email protected] <[email protected]> > 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: [email protected] > Ly Tran > > > > > From: Tony Graziano > Sent: Tue 8/10/2010 2:21 PM > To: Tran, Ly V. > Cc: [email protected] > 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. <[email protected]> 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 > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.984.8431 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > 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: [email protected] > Fax: 434.984.8431 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > 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: [email protected] Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] 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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