On Wed, Jan 27, 2010 at 3:53 PM, [email protected] <[email protected]> wrote: > I have seen answers in the past along the lines of "absolutely no clue" and > it looks like when the question was asked on 1/14 there was no answer. I'm > just asking if there is even a ballpark release time frame for 4.2. I do > understand "Like most software projects, sipXecs has approximate target > dates for when any given release will happen. Like most software projects, > those target dates are not always met." > I'm hoping someone has a general idea at least. It would go a long way if I > could tell my users that we think the forwarding problem may be resolved > around XYZ time frame.
Watch http://track.sipfoundry.org/browse/XX-7517 > > Thank you, > Matthew > > On 1/27/2010 1:53 PM, Tony Graziano wrote: > > 4.1-dev will be 4.2 stable when released. > > On Wed, Jan 27, 2010 at 2:35 PM, [email protected] > <[email protected]> wrote: >> >> 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/ >>>> >>>> >>> >>> >>> >> > > > > -- > ====================== > 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/ > > 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/ > -- M. Ranganathan _______________________________________________ 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/
