I found this in the Diagnostics > Registration
sip:1...@domain.com1928
sipxgw.domain.comsip:1...@domain.com 311
sipxgw.domain.comsip:1...@domain.com
;x-sipX-nonat> 2670 0004f21e7cdfsipxgw.domaincom
sip: 114 and 112 = Cisco 7970
sip: 111 = Polycom 650
I noticed. in the second column for the polyco
I think the operability issues are with the Cisco phones... but if
you're stuck with them, you're stuck with them.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rhon
Sent: Wednesday, April 21, 2010 10:37 PM
To: Nathan Nieblas; sipx-us
Hi Mate,
Going back to 8.3.5 firmware for cisco 7970G doesn't solve the problem.
Polycom still unavailbe to dial any cisco phones.
Any chance for you share what you did with 7970G and 8.3.5 firmware to make
polycom communicate with Cisco?
Thanks in advance.
Rhon
On Thu, Apr 22, 2010 at 3:17 AM
I deleted and re-added my gateway and it showed up. Not the whole server
config, just the gateway.
Sent via BlackBerry from T-Mobile
-Original Message-
From: Robert B
Date: Wed, 21 Apr 2010 19:28:57
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] ITSP Account setting missing i
I see another post about this, but the resolution sounds like it
involves blowing away your config and re-doing it...
Can anyone come up with a better solution to restoring the ITSP account
setting tab/screen?
-- Robert
___
sipx-users mailing list
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To: <29d9927d35da807a7cfa2519a20be...@mail.gmail.com>
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <45234>
Message-ID:
I have 7940's and 60's running P0S3-08-12-00 (latest)
against
:-)
they never liked SIP.
they wanted us to buy their $50K skinny based call manager. It is NOT
suprizing that many 'BYOP' (bring your own phone) voip providers won't
support the cisco's, and why you can get them on ebay cheap. But I have
dozens of them on the old broadsoft platform that we
Firmware versions should never start with the letter "POS". What were they
thinking?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax:
OpenACD seems to be the one that I'm hearing a bit about lately. It's
FreeSWITCH based, but not integrated or affiliated with sipX.
http://wiki.github.com/Vagabond/OpenACD/
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/21/2010 3:40 PM, Flatfender wrote:
On Tue, Apr 20, 2010 at 7:20 AM, Picher, Michael
wrote:
> Agents not staying logged in or appear logged in through GUI but not
> receiving calls and the ACD does not like it when you transfer calls out
> of queue. There may be others, but those are my major 2.
>
> Check the tracker.
>
So what ar
On 4/20/10 9:00 AM, Matt White wrote:
We are testing 79xx series with P0S3-08-11-00, the only issue we are seeing
right now is that the MWI never turns off with 4.2
But we have not seen any of the polycom calling issues or dial over trunk
issues in testingyet
P0S3-08-9-00 seems to wo
I was wondering if user-level music on hold (music on hold defined for
each user) worked like the old music on hold method in that playback
started from the beginning of the file each time. This is what I'm
experiencing in my test environment. Anyone else?
--
Josh Patten
Assistant Network Admi
On Wed, 2010-04-21 at 17:00 +0400, Nikolay Kondratyev wrote:
>
> > Support for re-INVITE (no SDP) in order to solicit a SDP
> > offer is mandatory. There is no way to avoid this.
> Can you please point me to the appropriate rfc? 3261?
RFC 3261
Section 13.2.1 Creating the Initial INVITE (page 79
On Wed, 2010-04-21 at 11:08 -0500, Ken Fulmer wrote:
> When we attempt to forward an internal extension to an external number
> and someone calls that extension from the PSTN, our provider is
> returning a 604 Does Not Exist Anywhere message.
> The provider is PaeTec and they need a SIP Diversio
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nikolay Kondratyev
[k...@nstel.ru]
> Support for re-INVITE (no SDP) in order to solicit a SDP
> offer is mandatory. There is no way to avoid this.
Can you
I have an issue with outbound calling on a system recently upgraded to 4.0.4
When dialing out from any extension, I get a dialtone.
A siptrace reveals a 407 Proxy Authentication Required msg.
An edited copy of the trace is here:
Time: 2010-04-21T15:58:16.242263Z
Frame: 130 sipXproxy.xml:11178
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
<47ab18ac0f23934383f2bba7ee3d8d7421fd834...@dc-us1mbex4.global.avaya.com>
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <45213>
Message-ID:
Confirmed by our Notes team that domi
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <45211>
Message-ID:
Hi All,
I have been testing the new vmail system in 4.2 and had a
few queries:
1/ I saw on the feature list that the admin
When we attempt to forward an internal extension to an external number and
someone calls that extension from the PSTN, our provider is returning a 604
Does Not Exist Anywhere message.
The provider is PaeTec and they need a SIP Diversion Header. Is there any
way to generate the diversion in sip
The voice talent is 'Karen' from www.gmvoices.com.
As in, "My name is Plankton, and this is my computer wife, Karen".
On Wed, Apr 21, 2010 at 12:00 PM, Todd Hodgen wrote:
> If you search the archives, I think you will find a discussion in the past
> about the voice used for sipXecs. I seem to
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <45206>
Message-ID:
All,
Good news, I managed to get MWI working between 4.2 and
CUCM6. SipX is defined over a SIP trunk on CUCM a
If you search the archives, I think you will find a discussion in the past
about the voice used for sipXecs. I seem to recall it about 6 months ago.
They are available, so you should be able to use them for customer messages
that match.
-Original Message-
From: sipx-users-boun...@list.sip
On Wed, 2010-04-21 at 11:13 -0400, Joseph Modi wrote:
> I am unable to upgrade to 4.2, it installs but then when I refresh, it
> defaults back to 4.1, any ideas.
Please be more specific.
What did you do to upgrade?
___
sipx-users mailing list sipx-user
Yeah, don't know if it is supported. The schema doesn't seem to like it.
Export and import.
I would remove any users from the administrators group (with the exception
of superadmin), and remove superadmin from the export file before importing
between 4.1.x and 4.2.
On Wed, Apr 21, 2010 at 11:19 A
If you were running a development release you may need to install
fresh... I haven't tested upgrading from a dev version... only from
4.0.4. Others may have different experiences.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf
Thanks guys!
I'll definitely use that command to clear my log files.
Ken
From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
Sent: Wednesday, April 21, 2010 10:11 AM
To: Ken Fulmer; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] sipXproxy.log
http://sipxecs.blogspot.
I am unable to upgrade to 4.2, it installs but then when I refresh, it
defaults back to 4.1, any ideas.
Joseph
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://list
Thanks. I deleted the file on a test machine and it reappeared when I
restarted the SIP Proxy service.
Ken
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, April 21, 2010 10:05 AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] sipXpro
http://sipxecs.blogspot.com/2010/03/using-sipviewer-on-your-pc-for-windo
ws.html
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ken Fulmer
Sent: Wednesday, April 21, 2010 11:00 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-use
You can delete it. Depends on how much you want to rely on those logs later
on.
logrotate -f /etc/logrotate.d/sipxchange
is a little more graceful.
Services Logging LevelsSupervisor select level... DEBUG INFO NOTICE WARNING
ERR CRIT ALERT EMERG SIP Proxy select level... DEBUG INFO NOTICE WARNING
We need to view calls in SIPviewer. However the sipXproxy.log file is 400 MB
and we can't make heads or tails of the information inside. Can we delete
the sipXproxy.log file? Will the system create a new one?
We had the general logging level set to INFO. We lowered it to EMERG and set
the SIP
> yes and yes. Pidgin works fine, so does spark.
Just tried Pandion against my 2 sipXecs 4.2 boxes.
I can register fine against the sipXecs that does not use DNS SRV but I
*CANNOT* register it against the sipXecs that uses DNS SRVs.
My investigation shows that DNS SRV records are set up properl
Yes, we are probably going to use a separate FS box as a SBC / B2BUA.
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, April 20, 2010 6:07 PM
To: mra...@gmail.com; kenful...@icstechnologysolutions.com
Cc: sipx-users@list.sipfoundry.org
Subject:
On Wed, Apr 21, 2010 at 9:00 AM, Nikolay Kondratyev wrote:
>
>
>> Support for re-INVITE (no SDP) in order to solicit a SDP
>> offer is mandatory. There is no way to avoid this.
> Can you please point me to the appropriate rfc? 3261?
>
>> Ranga
> Thanks and regards,
> Nikolay.
Please search the a
> Support for re-INVITE (no SDP) in order to solicit a SDP
> offer is mandatory. There is no way to avoid this.
Can you please point me to the appropriate rfc? 3261?
> Ranga
Thanks and regards,
Nikolay.
___
sipx-users mailing list sipx-users@list.si
On Wed, 2010-04-21 at 07:10 -0500, Robert B wrote:
> I am getting pretty sick of Sun's JVM using up so many resources...
>
> What about alternatives such as IBM's j9 or Apache Harmony? Does
> anyone have any experience with these and SipX?
That's really a topic for the sipx-dev list, not this one
I am getting pretty sick of Sun's JVM using up so many resources...
What about alternatives such as IBM's j9 or Apache Harmony? Does anyone
have any experience with these and SipX?
-- Robert
___
sipx-users mailing list sipx-users@list.sipfoundry.org
> On Wed, Apr 21, 2010 at 2:10 AM, Nikolay Kondratyev wrote:
> > Hi all,
> > i have a question regarding "late media" use in sipxbridge...
> > When incoming call is going through sipxbridge and is transferred by the
> > phone or by AA, sipxbridge converts Refer into re-Invite without sdp.
> > I h
On Wed, 2010-04-21 at 10:02 +0200, r.vanv...@raffel.nl wrote:
> You are absolutely right, that was my initial thought too. I checked
> with the ITSP today and they do not support re-INVITE's. On the other
> hand, they DO support REFER's.
Who is this and what SIP system are they using (answer off
On Wed, 2010-04-21 at 07:31 -0400, Tony Graziano wrote:
> To go from 3.10 to 4.2 would require two steps? 4.0 first right?
In theory, no, but that's the way I'd do it.
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list
On Wed, 2010-04-21 at 05:29 +0100, Hiral Patel wrote:
> What improvements have been made to the voicemail system in sipXecs
> 4.2?
The list of issues has been posted several times in the last couple of
weeks, and is easily found in the tracker.
The most important change is that it is now based
I've used www.pbxprompts.com for professionally recorded AA's. If you
have long prompts, they can get expensive, however I'm generally
impressed with overall quality.Tiffany's voice is close to the
existing voice used in sipexcs, but not exact. It would be great if
voice talent used in the si
To go from 3.10 to 4.2 would require two steps? 4.0 first right?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpde
On Wed, 2010-04-21 at 11:30 +1000, Graeme Allen wrote:
>
> Some further information:
>
> I got SipX to start, by changing /etc/init.d/sipxpbx
All that you did was disable the test that was designed to keep you from
hitting the problems you're hitting now.
As Josh said in another post, 4.2 has a
On Wed, Apr 21, 2010 at 2:10 AM, Nikolay Kondratyev wrote:
> Hi all,
> i have a question regarding "late media" use in sipxbridge...
> When incoming call is going through sipxbridge and is transferred by the
> phone or by AA, sipxbridge converts Refer into re-Invite without sdp.
> I have installat
You are absolutely right, that was my initial thought too. I checked with the
ITSP today and they do not support re-INVITE's. On the other hand, they DO
support REFER's.
Therefore, the compensation sipXbridge does in order to overcome problems with
REFER's is causing trouble with this particula
46 matches
Mail list logo
