Hi Huijun,
I have tried SIP 3.1.3 revision C (tried bot split and combined
separately) with ROM 4.1.3 and noticed few issues and intermittent freezing.
Then I've lowered the firmware 3.1.1(base version) and rom to 4.1.1(base
version), which solved the hanging on transfer issues but the
inter
>Subject: [sipx-users] How to diagnose Polycom Phone Freezing
> Following the update to sipx 4.0.1 and SIP 3.1.1 with ROM 4.1.1 (base
versions, not revisions) device files for Polycom 330 phones, users are
> reporting very frequent freeze - which occurs either during
conversation or simply when
Dave,
The allow connections sip to sip does not exist in the firmware for
the 5350. Ideas?
Thanks,
Max
On Tue, Jul 28, 2009 at 3:06 PM, Max Clark wrote:
> David,
>
> That makes perfect sense... except the command doesn't exist in my IOS
> version. I'm searching for the equivalent and will post b
David,
That makes perfect sense... except the command doesn't exist in my IOS
version. I'm searching for the equivalent and will post back the
results.
Thanks,
Max
On Tue, Jul 28, 2009 at 9:40 AM, Dave
Deutschman wrote:
> On the Cisco router, I did not see the following option set in your
> conf
Following the update to sipx 4.0.1 and SIP 3.1.1 with ROM 4.1.1 (base
versions, not revisions) device files for Polycom 330 phones, users are
reporting very frequent freeze - which occurs either during conversation
or simply when they come back to office.
1) which SIP & ROM would be more stable
I've long wanted to be able to go to an issue in the SIPfoundry issue
tracker by entering the issue ID into Firefox's search engine box. It
turns out to be fairly simple: Put the attached file,
sipfoundry-tracker.xml into the searchplugins directory
(~/.mozilla/firefox/*.default/searchplugins on
I just got a bunch of new polycom's for testing which all have the latest
firmware.
When the phone registers it does so with only the extension number and does not
include the users name.
So the line on the phone just has the extension instead of the Users name as a
label.
I can force this on
I am new to this product, and I need some over the phone help, I have
installed it and have it working through the web interface, I need
someone to walk me through the setup, configuring sip lines, configuring
soft phones and regular phones as well as setting up autoattendant and
all other features
I am trying to connect a Sipx box to a Trixbox using a SIP account on the
Trixbox. The trixbox is connected to the PSTN via PRIs. I created a SIP
account on the Trixbox and then went to Devices > Gateways and created a SIP
trunk. I can't see any registration attempts on the Trixbox from the
Manuel A. Pombo | CONTACT (DSI) wrote:
> Hi,
>
>
>
> Is there a programmatic way of logging in/out the ACD other than the
> User Portal?
>
>
Yes. ACD Presence server offers XML/RPC API and this is incidentally what
User Portal is using as well.
It's kind of defined here:
sipXcallLib/include
Hi,
Is there a programmatic way of logging in/out the ACD other than the
User Portal?
Thanks,
Manuel Pombo
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On Tue, 2009-07-28 at 08:48 -0700, Max Clark wrote:
> Scott,
>
> Thank you - the nameservers that the 5350 was set to use could not
> resolve the SRV records. I updated the gateway to point to nameservers
> that properly resolve with no change. I then went one step further and
> set an "ip host" c
Scott,
Thank you - the nameservers that the 5350 was set to use could not
resolve the SRV records. I updated the gateway to point to nameservers
that properly resolve with no change. I then went one step further and
set an "ip host" command on the gateway so that the lookups for the
domain would r
On Mon, 2009-07-27 at 17:39 -0700, Max Clark wrote:
> Hello,
>
> I am having a similar problem to the issue described here:
>
> "Incoming PSTN call through Auto Attendant transfer failure"
> http://xrl.us/be6g8b (Link to www.mail-archive.com)
>
> My configuration is as follows:
>
> TI/PRI -> Ci
2009/7/28 Chris Rawlings :
>
> Seems to be a very common problem... is it always going to be like this ?
The "problem" is the way in which the NAT is configured. If you do not
mind stray packets, you can allow for relaying of stray packets and
then you will not have the "problem"; but you will als
Seems to be a very common problem... is it always going to be like this ?
Thank You,
Chris Rawlings
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of M. Ranganathan
Sent: Tuesday, July 28, 2009 6:53 AM
To: Але
2009/7/28 Александр Горбунов :
> I'm trying SipX installed from sipfoundry-4.0.1-015823-i386.iso.
> I configure SIP trunk to local ITSP exactly by
> http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration
> document, but I do not apply patch4.zip. After that I c
I've recently updated 3.10.2 installation to latest stable 4.0.1 via yum
(as per the wiki tutorial )
I am running Polycom SIP 3.1.1 and ROM 4.1.1 (all base version, not the
revisions)on polycom 330 phones.
( spip_ssip_3_1_1_release_sig.zip and BootROM_4_1_1_release_sig.zip)
However, users repor
Patton SmartNode Series. For PRI/T1/E1 connectivity the 4960 is
available in several different configurations.
http://www.patton.com/products/pe_products.asp?category=354&MiDAS_Sessio
nID=eeaf53ea7c374be89058eb2fe2b9b8ba
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.o
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