I did not see anywhere for the option to time limit the conference calls. Did
I miss something?
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Ah, thanks for clarifying.
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Friday, December 04, 2009 6:53 PM
To: Todd Hodgen
Cc: Burden, Mike; Tony Graziano; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Disabling Music on Hold (Was: Re: Hold/Resume with
On Fri, 4 Dec 2009 18:24:59 -0500, Tony Graziano wrote:
> If everythibg is running, its not a cert issue. Does the server have the
> right gateway? Does your client need their sip password reset?
That won't be a solution anyhow as I think about that because I'll have
multiple SBC's so
can't give
Please Note that there is an issue with the Polycomm phone firmware 3.2.1.0054
Please upgrade to 4.0.4. Leaving MOH ON for both bridge and phone was
not a release goal ( i.e. has only been developer tested ) in release
4.0.x. It is the default configuration for 4.1, however.
Regards,
Ranga
> Did you use the same realm name (it defaults to the SIP domain name).
> If not, your clients are probably registering with the old one, which
> won't work.
Yes, that is set correct also.
Tony;
>Ah. Your remote client should shutdown/restart their app.
When remote phones authenticate, do they
One other thing to consider is the release of sipx you are running. You can
leave that on with 4.0.4, at least that is what I understand from a previous
note from Ranga. Are you running 4.0.2 by change, or something earlier?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun
There are several of us on the list that provide SipX solutions, so maybe
someone will chime in with a support offering.
We offer customers that purchase a SipX solution through us with an emergency
contract. But its difficult for most providers to provide support for system
they have been cus
Ah. Your remote client should shutdown/restart their app.
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
Helpdesk Cont
If everythibg is running, its not a cert issue. Does the server have the
right gateway? Does your client need their sip password reset?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Co
>does sipxconfig work?
Yes, I've got access to everything.
>can you run preflight tests?
Yes, everything returns correct settings for everything, DHCP and DNS.
>did you see what is running with sipxproc command?
Everything is running.
On Fri, 4 Dec 2009 17:33:26 -0500, Tony Graziano wrote:
On Fri, 2009-12-04 at 16:37 -0600, m...@grounded.net wrote:
> 2009/12/04 16:31:37.712 INF: [CID=0x30eddca5] <<< REGISTER sip:mydomain.com
> SIP/2.0
> Method(REGISTER) SRC: xx.xx.113.30:49073:UDP enc=0 bytes=565
> 2009/12/04 16:31:37.713 DTL: [CID=0x30eddca5] Found
> NIST|ZjNjYTMwN2EwZTdjODY1NGI
I too would be interested, though I would probably never use it. As far
as I know though the only way to get paid support is to also buy Nortel
SCS and all the voicemail and extension licenses that go along with it.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-467
2009/12/04 16:31:37.712 INF: [CID=0x30eddca5] <<< REGISTER sip:mydomain.com
SIP/2.0
Method(REGISTER) SRC: xx.xx.113.30:49073:UDP enc=0 bytes=565
2009/12/04 16:31:37.713 DTL: [CID=0x30eddca5] Found
NIST|ZjNjYTMwN2EwZTdjODY1NGI4NWYyMjU0NjRkNjA1ZmI.|z9hG4bK-d8754z-610834723241ef77-2---d875
4z-|REG
On Fri, 4 Dec 2009 17:02:22 -0500, Tony Graziano wrote:
> When you first do the certs, a reboot is on order. There are some notes on
> here from the last week from someone with an issue trying to get an HA
> system working. There was a note about clearing out the existing certs
> first. you might w
When you first do the certs, a reboot is on order. There are some notes on
here from the last week from someone with an issue trying to get an HA
system working. There was a note about clearing out the existing certs
first. you might want to look for it.
On Fri, Dec 4, 2009 at 5:01 PM, m...@ground
Got it... found it... disabled it... and it seems to have fixed my
hold/resume problem!!
Thank you VERY much!
Mike Burden
Lynk Systems, Inc
e-mail: m...@lynk.com
Phone: 616-532-4985
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Friday, December 04, 20
I finally fired up the new server I've been wanting to, including dhcp/dns this
time. I
was sure to take the steps to re-gen the certs but now am seeing that remotes
cannot log
in again.
Last time I tried switching the a new server, this was the problem and I was
told to make
sure to follow
MOH is a sip setting in polycom
moh uri needs to be erased in sipxconfig for the phone, then resend the
profile.
On Fri, Dec 4, 2009 at 4:20 PM, Burden, Mike wrote:
> Am I missing something in “Phone Groups?” I’m not finding a setting to
> disable Music on Hold under Polycom IP550. For that
Am I missing something in "Phone Groups?" I'm not finding a setting to
disable Music on Hold under Polycom IP550. For that matter, I can't
find an option to do that on the phone itself, either.
The big, orange box on the first page of the Wiki page for using Polycom
phones with sipXecs seems
Me too.
It would go a a long way with management.
Burden, Mike wrote:
> I'm interested in the same thing. Especially if the support provider
> is also knowledgeable about Polycom phones.
>
>
>
> Mike Burden
> Lynk Systems, Inc
> e-mail: m...@lynk.com
> Phone: 616-532-4985
>
>
>
>
>
>
>
>
> -
I'm interested in the same thing. Especially if the support provider
is also knowledgeable about Polycom phones.
Mike Burden
Lynk Systems, Inc
e-mail: m...@lynk.com
Phone: 616-532-4985
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@li
We have enjoyed experimenting with SIPX over the last year. One road block to
a full deployment is the availability of a paid support option as an emergency
measure. Normal operation can be handled but management is concerned about the
weird cases (database corruption or other nastiness) that
Picher, Michael wrote:
> Hey Dimitris,
>
> Did they get rid of the lag on connecting to an outside phone number?
>
> Mike
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of dimitris
> Sent: Thursday, December
Sen-
We can't seem to track down where the MOH setting is in the ciscoplus
plugin. Do you know more specifically where it is located? We have gotten it to
work on the Polycoms and Linksys phones, but the not cisco.
Do you know if paging works? When someone tries to page the phone rings
ins
Consult the wiki...
This page should help:
http://sipx-wiki.calivia.com/index.php/SipXecs_4.0.2_Upgrade
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Dan White
Sent: Friday, December 04, 2009 2:44 PM
To: Scott
Where can I download the latest version of SIPX and how do I run the
upgrade? Is there instructions for this?
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: Friday, December 04, 2009 2:41 PM
To: Dan White
Cc: sipx-users@list.sipfoundry.org
Subject: Re
On Fri, 2009-12-04 at 14:25 -0500, Dan White wrote:
> OK, here is my problem
>
> I use vitality.net as my sip provider, I have a problem with call
> forwarding,
>
>
>
> Here is the setup, If I have extension 200 and dial from within the
> system, and I call forward an extension, it works, So I
Thanks Staffan.
I've posted this to the wiki:
http://sipx-wiki.calivia.com/index.php/HowTo_configure_Cisco_SIP_Gateway
_with_sipX
Mike
From: Staffan Kerker [mailto:ietf-li...@kerker.se]
Sent: Thursday, December 03, 2009 11:07 AM
To: Picher, Michael
Cc: Dylan Ebner; Mike Ketchum; sipx-u
OK, here is my problem
I use vitality.net as my sip provider, I have a problem with call
forwarding,
Here is the setup, If I have extension 200 and dial from within the
system, and I call forward an extension, it works, So I dial extension
206, it forwards and works to a cell phone. But, I hav
Also, have you figured out how to dial out to skype users?
Dial in works great from a skype user to your skype account that your
sip # is registered to.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of dimitris
Se
Sure. They have international dialing rates.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Sen Heng
Sent: Friday, December 04, 2009 4:37 AM
Cc: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Mobile gatew
Hey Dimitris,
Did they get rid of the lag on connecting to an outside phone number?
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of dimitris
Sent: Thursday, December 03, 2009 4:40 PM
To: sipx-users@list.sip
On Fri, 2009-12-04 at 13:25 -0500, Dale Worley wrote:
>
> Exactly what sort of load problem are you seeing? As a rule of thumb,
> you should have under 80% CPU utilization and zero paging activity.
Important point there - zero swapping. Once the system starts to swap,
things get bad fairly quic
No I meant on the firewall that connects to the Internet.
-Original Message-
From: an...@iguanait.com [mailto:an...@iguanait.com]
Sent: Thursday, December 03, 2009 7:33 AM
To: Picher, Michael
Cc: Robert Joly; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Calls goes through but
On Thu, 2009-12-03 at 08:53 +, Gabor Paller wrote:
> My problem is that the SipX server under load produces a large number
> of retransmissions which further decrease the server’s performance. My
> idea is that I start to fiddle with retransmission timers.
Exactly what sort of load problem are
Dylan,
1. I am not sure what do you mean change boot-up image to sth else ? But
there is option for you to change firmware version on GUI.
2.MOH, you need put MOH number (Sipx default is 101 ?), you can find from
GUI I think.
3. For NTP server, can you try use 0.us.pool.ntp.org or 1.us.pool.ntp.o
Sen-
A couple more questions.
1. Is their any way to change to boot-up image from Cisco to something else?
2. Music on Hold doesn't seem to be working. Any ideas their?
3. The phone pulls time from NTP, but it doesn't seem to offset it to our
timezone. We have tried using two different ntp serv
Dylan,
To change the background logo, it needs to create another xml file. More
details please follow this link.
http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7961g_7961g-ge_7941g_79
41g-ge/4_1_3/english/administration/guide/7961cus.html#wp1030672
Thanks,
Sen
-Original Message-
Sen-
I know this is trivial, but is it possible to update the background logo on
the 7941G? The wiki article says the url needs to be at Phone
parameters/logo_url, but I cannot seem to find this in the 7941 config. Is it
just hiding or is it currently not an option?
Thanks
Dylan Ebner, Netw
Ok, Thanks Dylan clarified everything is ok.
Cisco has recently re-badged Linksys phones to Cisco SPA 50XG. There are few
more features Cisco has added in like SSL VPN client, mobility,codec etc
I am writing script for these phones and will update community in two weeks
time. The testing and
On Fri, 2009-12-04 at 17:10 +0800, Wen Jun wrote:
> Woo, my mistake in the contact address. The 1500 is registered with
> incorrect sip:1...@sipx.net in contact address.
sipXecs will handle the situation correctly, that is, as the SIP
specifications define it. But the results are likely to confu
Hey everyone-
Thanks for all the input. The processnodename is the problem. When we add our
sipx server IP address to the prcessnodename section in CiscoPlus gui the phone
registers. We have not tried using the domain name yet. At least it looks like
the gui module is working so the phones can
you should do a call trace so you can understand if the issue is the xlite
beta or at sipx. no point in opening a tracker issue here if it is a
counterpath problem. if it is a counterpath issue, the ticket maybe could be
filed with them. does it work with the stable version of xlite?
2009/12/4 Bry
On Fri, Dec 4, 2009 at 8:39 AM, Josh Patten wrote:
> Speaking of the wallboard application, does anyone have a link to it or
> to images of it?
>
http://wiki.voiceworks.pl/display/vwost/sipXworks+example+queue+stats+screen
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sipx-users mailing list s
SIP keepalives enable and seesion timers are disabled and I have this loaded on
a MAC and A windows machine and I get the same result when calling into a
conference bridge. Calls drop at around 30 seconds whether I am the only one in
the conference or I am in the conference with others. Think I
Well I didn't write the ACD wallboard application, but someone else
already has and it is slated to be included with the 4.2 release. What I
have to offer is the configuration for setting up a large display system
for call centers that can display the wallboard application that has
already been
OK, I will deble check the "processnodename" to see is this caused issue.
Because I know there are few people in community here have successed use
Cisco phone through GUI.
Thanks,
Sen
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfound
Sen Heng tcd.ie> writes:
>
>
>
> I think I got 7961 and 7970 working. Jes tested as
> well.
> I didn't get change to test 79x5 phones, if someone
> menaully get 79x5 phones working, please post the difference part of cnf
> file.
>
>
> Thanks.
>
>
> From: sipx-users-bounces list.sipf
Hi,
I would certainly be interested in what you have done here.
We are still considering SipX as a enterprise wide solution but previously the
ACD functionality has prevented us from implementation, this (ACD
queue/wallboard) was one of the limiting factors.
Cheers,
Grant
-Original Messag
I think I got 7961 and 7970 working. Jes tested as well.
I didn't get change to test 79x5 phones, if someone menaully get 79x5 phones
working, please post the difference part of cnf file.
Thanks.
_
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfound
We have a similar issue, but only with certain end-points.
We have the issue with X-Lite-Beta (XLB) and a Yealink T26P (good phone,
but unfortunately no SRV support yet, so wait...).
XLB with other products is OK, Yealink with others as well.
It turns out that XLB sends RTCP data to the wrong port
Yes, it's in the tracker:
http://track.sipfoundry.org/browse/XX-5358
"This next part is required, otherwise the phone claims to be unprovisioned.
It just requires your SIP server's DNS name. It also appears that the phone
will not pick up some XML configuration changes if the
element is not pres
Since you have so many NXX's maybe this is a solution (partly borrowed
from Scott):
Say you have area codes 111 and 222 and NXX's 888 and 999 for both local
area codes and ooo is all long distance then (in this order):
StripArea_111
Prefix 111 and 7 digits
Dial matched suffix
no gatewa
Hi,
Can I ask for a solution. If I have one GSM mobile gateway and one Skype
Trunk connect with Sipx.
When I use my mobile phone call to GSM gateway, once the call connected, is
that possible I take control Skype trunk and call internation number through
Skype trunk ?
Thanks in advance,
Sen
Woo, my mistake in the contact address. The 1500 is registered with
incorrect sip:1...@sipx.net in contact address.
Thanks
Jun
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: Thursday, December 03, 2009 9:10 PM
To: Wen Jun
Cc: sipx-users@list.sipfoundry
Scott Lawrence nortel.com> writes:
>
> On Thu, 2009-12-03 at 10:05 +, Nitin wrote:
> >
> > I am unable to use sipviewer - It read and does nothing(the xml file)
>
> You only included a fragment of the file.
>
> Attach the file to the mail (you may need to compress it first to get
> under
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