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.
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