I think you are asking for a commitment or rough timeframe that developers are not providing, other than their rule that it is released soon after all issues for the release are resolved. They are extremely open about it, you can look at the roadmap to see what they are still working on.
However, if you read one of the emails on the DEV list today, you will see that they are pushing to fix the final few items, as well as negotiating on some compromises to meet their goal to release it soon, and that maybe they have a target date of the next 60 days or less. That is what I read into it, but I don't wear glasses when I read. From: [email protected] [mailto:[email protected]] On Behalf Of [email protected] Sent: Wednesday, January 27, 2010 12:53 PM To: [email protected] Subject: Re: [sipx-users] Can't forward from external number to external number 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. 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]> <mailto:sip%[email protected] <mailto:sip%[email protected]> >;user=phone>;tag=2099232641-1264554945912- To: "DSI HOLDING COMPANY 251 DSI Corp" <sip:[email protected]> <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]> <mailto: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]> <mailto: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> <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.
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