It is very possible the one created 14 months ago has some custom user
settings in it from the menu/interface of the phone. these override
the generated config. Wipe the phone to defualts (i.e., 4,6,8,* at
bootup with larger polcyoms, try again).
I erased the polycom 650 to default, using
I've just start a preliminary work on this issue, and i start working on
this use case:
call forwarding to external number,
so A, B and C where C is not local extension, reachable through an
unmanaged gateway (so without sbc)
what i see is that:
A call B,
B sends 302 to A but this message is
From what i saw, the capture does not make any sense. I don't see the
183 and the 200 you were mentioning.
On 02/07/2012 05:49 PM, Domenico Chierico wrote:
I've just start a preliminary work on this issue, and i start working
on this use case:
call forwarding to external number,
so A, B
Sorry the filter was too strict maybe try with this one
On Tue, Feb 7, 2012 at 11:25 AM, Joegen Baclor jbac...@ezuce.com wrote:
From what i saw, the capture does not make any sense. I don't see the
183 and the 200 you were mentioning.
On 02/07/2012 05:49 PM, Domenico Chierico wrote:
I've
-- Forwarded message --
From: Luciano Berardi luciano.bera...@sip2ser.it
Date: 2012/2/7
Subject: [sipx-dev] (no subject)
To: sipx-users@list.sipfoundry.org, sipx-...@list.sipfoundry.org
Hi, I just did a patch (sipxecs 4.4) to show in the ListGateway page the
Enabled flag.
It
Ok I see it now. Indeed the responses are never propagated. I can tell
from the record routes taht your unmanaged gateway is another sipx. I
do remember posts in the past from the old maintainers that 302
initiated by a third party is not advisable because it may conflict with
sipx internal
On 2/7/2012 2:12 AM, Tony Graziano wrote:
realize you cannot do call transfers with the above mentioned method,
so your mileage may vary.
Yes you can! You have the have the DID configured to come in on 5080.
Example voip.ms SIP URI:
{DID}@pbx1.example.com:5080
or
{DID}@youripaddress:5080
In sipXecs the non-default settings all have dashes around them.
the other thing you could do is check out the config files for differences.
Mike
On Tue, Feb 7, 2012 at 4:33 AM, Claas Hilbrecht
claas.hilbrecht+maillinglists.sipx...@linum.com wrote:
It is very possible the one created 14
Change the order of the lines on the phone and see if the problem flips
lines.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Claas Hilbrecht
Sent: Tuesday, February 07, 2012 1:33 AM
To: Discussion list for
--Am Dienstag, 7. Februar 2012 08:46 -0500 Michael Picher
mpic...@ezuce.com schrieb:
In sipXecs the non-default settings all have dashes around them.
Yes, this is a good thing and I used this already to find something, but
without luck so far.
the other thing you could do is check out the
Change the order of the lines on the phone and see if the problem flips
lines.
Good idea, I will try this.
And just to make sure. Is sending a busy on DND something that is supported
by sipxecs? Reading the help text on the DND setting I would think so.
--
Claas Hilbrecht
you'll find the phone configs in
/var/sipxdata/configserver/phone/profile/tftproot (or something close to
this...
Mike
On Tue, Feb 7, 2012 at 11:41 AM, Claas Hilbrecht
claas.hilbrecht+maillinglists.sipx...@linum.com wrote:
--Am Dienstag, 7. Februar 2012 08:46 -0500 Michael Picher
I do recall the #of calls per line key needs to be 1, and voicemail
needs to be disabled. If you find NO difference in the configs of the
phone, you should peruse the web interface of each phone to see if one
(that has not been wiped yet) has a manual setting. I seem to recall
that was something I
Is the provider sending calls to port 5080 or 5060? Who is the provider.
Calls need to be sent to port 5080. Does this mean transfers don't work? If
so check or ask the provider to send invites to port 5080.
On Feb 7, 2012 1:54 PM, Mark Roseboom mark.roseb...@38media.net wrote:
Using sipXecs
Mark,
You will want to provide the name of the ITSP, the model of phones you are
using, and the version of firmware they are running on. Brand of router can
be helpful as well.
Do a search on the wiki at wiki.sipfoundry.org and look up the page on the
sipviewer. It will have instructions
Currently, the provider is sending calls to 5060, however in our network
configuration, the sipxec server is behind a firewall, so I am forwarding
port 5060 requests to port 5080 on our sipxec. The provider is IIS
Group. You are correct on hold or transfers don't work for inbound calls.
Is this
Todd,
Provider is IIS Group. Phones are Grandstream GXP2000. Firmware is
1.2.2.26. Router is a DD-WRT v24. The download link for the sipviewer .jar
file found at
http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer
does
not work unfortunately so I am unable to use
Ask them to send invites for inbound calls to port 5080 and make sure the
firewall does the same (5080:5080).
On Feb 7, 2012 2:06 PM, Mark Roseboom mark.roseb...@38media.net wrote:
Currently, the provider is sending calls to 5060, however in our network
configuration, the sipxec server is
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