Does 4.1 have a release date? From looking at the roadmap, it appears 4.2 is the next proposed release. Maybe I'm reading something incorrectly. I have helped a few of the users get by with the personal attendant, but I would like to give them an idea as to when I will have something that could fix their issue.
On 1/26/2010 8:57 PM, M. Ranganathan wrote: > On Tue, Jan 26, 2010 at 9:16 PM, Tony Graziano > <[email protected]> wrote: > >> I don't know what "206 no peers available" means from sipxbridge. >> > Well, there are two problems at hand here. The 481 error from > sipxbridge on the re-INVITE -- I believe this to be a sipxbridge bug > that I have fixed already in 4.1. You can give that a try. > However, I am curious about why verizon issending a re-INVITE as soon > as the call is. Perhaps it does not like the fact that you are using > 10.87.20.5 address in your call setup. > > > > Regards, > > > Ranga. > > > > >> Perhaps someone else can help? >> ============================ >> 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: [email protected]<[email protected]> >> To: Tony Graziano<[email protected]>; >> [email protected]<[email protected]> >> Sent: Tue Jan 26 21:12:55 2010 >> Subject: Re: Can't forward from external number to external number >> >> Just 1. 95% of my users have 1 line assigned to 1 phone, and everyone is >> definitely having the issue. >> >> On 1/26/2010 8:01 PM, Tony Graziano wrote: >> >>> How many phones is your line registered on? >>> >>> On Tue, Jan 26, 2010 at 8:40 PM, [email protected] >>> <mailto:[email protected]> <[email protected] >>> <mailto:[email protected]>> wrote: >>> >>> Ok. Lets try this again. I think I have some better data now >>> thanks to some help from Tony. I waited until nobody was on my >>> system, moved all the logs out, immediately made a test call, and >>> then immediately moved those calls to a temp directory. I ran >>> merge-logs in that temp directory, opened it in sipviewer, and >>> attached the merged.xml that was created. >>> In this test, I was calling from 6155008073 (my cell phone) to >>> 6159253043 (my Sipx/Desk phone) which was set in the Sipx GUI to >>> forward to 6155914780 (my home phone). My home phone rung, and the >>> call was dropped as soon as I answered it. I didn't try to mask >>> any of my phone numbers in the logs this time. >>> >>> I have to guess the 2 parts from merged.xml below indicate a >>> problem. Googling '481 Peer dialog is null' doesn't get too many hits. >>> Sorry for not sending correct or helpful information earlier. >>> Hopefully what I'm sending now is a little more helpful. I didn't >>> think I needed to send a screenshot from sipviewer, but I will be >>> glad to if that would help. >>> >>> Thank you all for your help, >>> Matthew >>> >>> Time: 2010-01-27T01:16:12.311000Z >>> Frame: 29 sipxbridge.xml:378 >>> >>> Source: nshpbx1.sipx.voip-sipXbridge >>> Dest: 10.87.20.5:5060<http://10.87.20.5:5060> >>> >>> SIP/2.0 481 Peer dialog is null >>> Via: SIP/2.0/UDP >>> 10.87.20.5;branch=z9hG4bK-sipXecs-19e8a5983eb41c5558b4327c16988037cc75 >>> Via: SIP/2.0/UDP >>> 10.87.20.5:5090;branch=z9hG4bKeca98329a090bdd96d9d633f294be82f333631 >>> CSeq: 917280447 INVITE >>> Call-ID: [email protected] >>> From: "WIRELESS CALLER"<sip:[email protected] >>> >>> <mailto:sip%[email protected]>;user=phone>;tag=2099232641-1264554945912- >>> To: "DSI HOLDING COMPANY 251 DSI Corp" >>> <sip:[email protected]>;tag=9df1f5b6 >>> >>> Server: sipXecs/4.0.4 sipXecs/sipxbridge (Linux) >>> Contact:<sip:[email protected]:5090 >>> <http://[email protected]:5090>> >>> Supported: replaces,100rel >>> Content-Length: 0 >>> >>> Time: 2010-01-27T01:16:12.321000Z >>> Frame: 33 sipxbridge.xml:383 >>> >>> Source: nshpbx1.sipx.voip-sipXbridge >>> Dest: 172.30.209.62:5070<http://172.30.209.62:5070> >>> >>> SIP/2.0 481 Call leg/Transaction does not exist >>> Via: SIP/2.0/UDP >>> 172.30.209.62:5070;branch=z9hG4bK548gtc308ghhhoccp4g1cbm1bd9v2.1 >>> From: "WIRELESS CALLER"<sip:[email protected] >>> >>> <mailto:sip%[email protected]>;user=phone>;tag=2099232641-1264554945912- >>> To: "DSI HOLDING COMPANY 251 DSI Corp"<sip:[email protected] >>> <mailto:sip%[email protected]>>;tag=5102113 >>> Call-ID: [email protected] >>> <mailto:[email protected]> >>> CSeq: 917280446 INVITE >>> >>> Server: sipXecs/4.0.4 sipXecs/sipxbridge (Linux) >>> Supported: replaces >>> Contact:<sip:[email protected]:5080;transport=udp> >>> Reason: ~~id~bridge;cause=213;text="Relayed Error Response" >>> Content-Length: 0 >>> >>> >>> On 1/26/2010 3:26 PM, [email protected] >>> <mailto:[email protected]> wrote: >>> >>> This is similar to something I posted a week or so ago about >>> trying to forward at the handset level, but I'm assuming it is >>> a a completely different issue. >>> If a user sets a forward through the web gui and specifies an >>> external number, they have an issue if the inbound call to be >>> forwarded is also from an external number. The call rings on >>> the destination phone, but is disconnected with a click as >>> soon as it is answered. If the call is forwarded to an >>> internal extension, everything is fine. If the call is >>> forwarded to an external number and the caller is on an >>> internal phone, everything is fine. This sounds like a >>> permission issue, but if so, I don't understand why it makes >>> it as far as calling the destination phone , but then >>> disconnects when it is answered. >>> >>> The text below is from sipxbridge.log. I didn't want to post >>> the phone numbers in question for a automated routine of some >>> sort to grab at least, so I changed the 615 area code to 222 >>> in the logs. All area codes involved in this log are 615. In >>> this case, the polycom phone is at 4670142. I set it to >>> forward to 5008073. The inbound call came from 2439019. >>> 10.87.20.5 is my sipx server. pcelbcn0001.dsi.globalipcom.com >>> <http://pcelbcn0001.dsi.globalipcom.com> [172.30.209.62] is my >>> Verizon gateway. I would be more than happy to provide any >>> more information, but I'm not sure where I should be looking. >>> >>> Sipx 4.0.4, sixbridge, Verizon VOIP, No firewall (not needed, >>> private connection), Polycom 450s and 550s - bootrom 4.2.1, >>> firmware 3.1.3C split. >>> >>> Thanks as always, >>> Matthew >>> >>> >>> >>> >>> >>> -- >>> ====================== >>> Tony Graziano, Manager >>> Telephone: 434.984.8430 >>> Fax: 434.984.8431 >>> >>> Email: [email protected]<mailto:[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/ >>> >>> Why do mathematicians always confuse Halloween and Christmas? >>> Because 31 Oct = 25 Dec. >>> >>> >> _______________________________________________ >> sipx-users mailing list [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users >> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users >> sipXecs IP PBX -- http://www.sipfoundry.org/ >> >> > > > _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
