How many phones is your line registered on? On Tue, Jan 26, 2010 at 8:40 PM, [email protected] < [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 > > 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]<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> > 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 > > 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]<sip%[email protected]> > ;user=phone>;tag=2099232641-1264554945912- > To: "DSI HOLDING COMPANY 251 DSI Corp" > <sip:[email protected]<sip%[email protected]> > >;tag=5102113 > Call-ID: [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] 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 [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] 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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