I don't know what "206 no peers available" means from sipxbridge.
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/
