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]
<mailto:[email protected]> <[email protected]
<mailto:[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]
<mailto:[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]
<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/
----- Original Message -----
From: [email protected]
<mailto:[email protected]><[email protected]
<mailto:[email protected]>>
To: Tony Graziano<[email protected]
<mailto:[email protected]>>;
[email protected]
<mailto:[email protected]><[email protected]
<mailto:[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]>
<mailto:[email protected]
<mailto:[email protected]>>
<[email protected]
<mailto:[email protected]>
<mailto:[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><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]>;tag=9df1f5b6
Server: sipXecs/4.0.4 sipXecs/sipxbridge (Linux)
Contact:<sip:[email protected]:5090
<http://[email protected]:5090>
<http://[email protected]:5090 <http://10.87.20.5:5090>>>
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><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]>
<mailto:[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]>
<mailto:[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>
<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]><mailto:[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]
<mailto:[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] <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/
