Yeah! I can confirm that this forwarding issue between Sipx/Verizon is resolved in 4.2. Thank you for all the hard work. It looks great from what I have seen so far.
On 1/27/2010 2:59 PM, M. Ranganathan wrote: > 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/ >> >> > > > _______________________________________________ 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/
