I see that click the "help" link should take me to:
http://www.sipfoundry.org/doc/sipxhelp/4.0.html
which comes back as:
*Page not found*
Recently the SIPfoundry site got re-organized and updated. Please check your
bookmarks.
Is there a manual way to fix the link in an installed system? If so
On Wed, Dec 9, 2009 at 9:24 PM, Jake Ballamis <
jballa...@alliancetechsolutions.com> wrote:
> Polycom by default does this...
>
> If you pickup the handset THEN dial, it dispatches when it finds the first
> match (3 extensions for internal dial plan, as an example). You can set the
> polycom to "
Jake,
You can make adjustments to fix this in the dial plan under the phone itself
- there is a digit mapping form. You need to make some considerations here.
Most likely you will want to do this under the Phone Group, so that it
applies to all phones of the same type. I'd recommend you find
The Viking RAD-1A is a device that will auto answer a call from an ATA
and send the sound as RJ11 or 600ohm pair of wires to the amplifier.
The key is using an ATA that supports CPC. The Linksys PAP2T-NA is one
that I used. The Linksys ATA device has Calling Party Control (CPC)
disabled by de
I am trying to enable click to call for an existing contact list application
written in PHP. I have done some initial trials to place calls from this app
using the sipx REST interface for this purpose, but have not yet been
successful.
Does anyone have some example code that calls this interfa
> Polycom by default does this...
>
> If you pickup the handset THEN dial, it dispatches when it finds the first
> match (3 extensions for internal dial plan, as an example). You can set the
> polycom to "allow user to dispatch" I think in sipXconfig.
>>
> Normally, I tell users to dial then HIT
On Wed, Dec 9, 2009 at 7:29 PM, Jake Ballamis <
jballa...@alliancetechsolutions.com> wrote:
> All,
>
>
>
> I have two questions, the first of which I’m sure is a quick fix and the
> other I’m not so sure.
>
>
>
> #1 – I, for the life of me, can’t figure out how to configure my handsets
> to immed
All,
I have two questions, the first of which I'm sure is a quick fix and the
other I'm not so sure.
#1 - I, for the life of me, can't figure out how to configure my
handsets to immediately send dialed digits when it's off-hook. All I
can do right now is dial the digits and click "Send".
I've come across interop issues with various SIP UAs and now a commercial
SBC (Cisco ASR 1000 Series SBC SP Edition) and SIPXecs, which stems from the
SIPXecs use of the "Tag" parameter in the "From" Header for generating nonce
values for SIP digest authentication.
Basically the position of SIP Fo
Scott,
Thanks for that - will give it a try with the upgrade to 4.0.4!
Many Thanks
Abdul
> Subject: Re: [sipx-users] Manual Software Updates
> From: scott.lawre...@nortel.com
> To: abdul_ma...@hotmail.com
> CC: sipx-users@list.sipfoundry.org
> Date: Wed, 9 Dec 2009 19:04:41 -0500
>
>
On Wed, 2009-12-09 at 23:41 +, Abdul Mayat wrote:
> Hello,
>
> I have a system installed running v4.0.3 which was intalled from an
> ISO running on CentOS.
>
> My server does not have any external internet connectivity, so when I
> check for new software it always says no updates are avail
Hello,
I have a system installed running v4.0.3 which was intalled from an ISO running
on CentOS.
My server does not have any external internet connectivity, so when I check for
new software it always says no updates are available.
Is there a method of manually downloading a patch (
Dial any user's voicemail in your system and listen to the ENTIRE
message and you will hear "to reach the operator, dial 0". The poster is
asking how to disable this message and if possible disable pressing 0.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On
You can change the alias for operator from the AA.
What is it you are trying to do?
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:
Is there a simple way to not play the prompt which says to press zero for
the operator? Along with that, is there a way to ignore zero?
If there is not a simple way, could you please tell me where the VXML files
and wav files are for voicemail?
Thanks,
Derek
___
Ranga,
I wasn't really asking you to speak for OpenVZ, my reply was information
that may provide insight into the issue -- the fact that it creates
these venet0 and venet0:0 interfaces using private routing and
forwarding by default could explain where 127.0.0.1 was coming into the
picture.
I would want to know what ifconfig shows.
In a lab vmware testing environment you would use a bridged connection using
a standard ethernet interface.
I think you are on the right path.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...
I brought up a similar issue a while ago. As I recall I had to add the
primary and redundant server to DNS on both boxes before continuing the
install (I just ssh'ed into the box while the install was still running
and added the records and restarted the named service.
This isn't a problem if
On Wed, Dec 9, 2009 at 12:46 PM, M. Ranganathan wrote:
> On Wed, Dec 9, 2009 at 12:13 PM, Scott Lawrence
> wrote:
>> On Wed, 2009-12-09 at 11:00 -0600, Robert B wrote:
>>> Scott,
>>>
>>> There's only one interface on the server, so -i would be redundant -- at
>>> least that's my understanding.
>>
Ranga,
I am still confused as to why this happens with any installation I do...
However, I have a theory that may or may not be correct.
On my system, since I run sipXecs in an OpenVZ container, I do not have
an eth0 interface. In my current setup, I have venet0 and venet0:0
interfaces:
venet
On Wed, Dec 9, 2009 at 12:13 PM, Scott Lawrence
wrote:
> On Wed, 2009-12-09 at 11:00 -0600, Robert B wrote:
>> Scott,
>>
>> There's only one interface on the server, so -i would be redundant -- at
>> least that's my understanding.
>
> No... you want the messages on the loopback interface too. Not
On many of our sites, I can block admin access using apache configurations that
don't allow users to go to http://www.site.com/admin for example.
On sipx, it seems that everyone lands on
https://sipx.site.com:8443/sipxconfig/app which is the same path that the
superadmin takes. It seems that I
On Wed, 2009-12-09 at 11:00 -0600, Robert B wrote:
> Scott,
>
> There's only one interface on the server, so -i would be redundant -- at
> least that's my understanding.
No... you want the messages on the loopback interface too. Note that
you see the messages in this trace between the proxy and
On Wed, 2009-12-09 at 11:00 -0600, Robert B wrote:
> There's only one interface on the server, so -i would be redundant -- at
> least that's my understanding.
There's 'eth0' and then there's 'lo'. Messages between the sipXecs
components are often sent on 'lo'.
Dale
___
On Wed, 2009-12-09 at 10:12 -0600, Robert B wrote:
> Still fighting with this. I took a few days off to work on other
> things but the time is rapidly approaching that I need to start
> testing real calling.
>
> Attached to this email is a packet capture that shows the issue.
>
> Scott Lawrence p
Sorry my fault when writing the steps.
Of course I add the second server first in Web UI and then enter the data in
setup screen on second server...
But I installed both servers in parallel and stopped on second server until
I added it on Master in Web UI.
As this is not the first time I installed
On Wed, 2009-12-09 at 05:55 -0800, Jordan Turner wrote:
> It is nice due to its organizational layout. To the people who comes
> from the "Windows" world, the UI is welcoming. It reminds me of the
> MMC or management console Microsoft provides which alot of good
> software company uses for their
Jordan Turner wrote:
> It is nice due to its organizational layout. To the people who comes
> from the "Windows" world, the UI is welcoming. It reminds me of the MMC
> or management console Microsoft provides which alot of good software
> company uses for their UI management. In the IP world, th
The SIPfoundry issue tracker, code browser, user admin system, and
possibly also the mailing lists and subversion will be off line for no
more than 5 hours starting at:
Location
Local time
Time zone
Universal Time
Coordinated
Thursday, December 10,
2009 at 00:00:00
UTC
Boston / Ottawa
Wednesday
Hi
I'm using SipX together with OpenSIPS and the parts I'm using it for works
great. I'm using OpenSIPS as a "core" SIP proxy for
ENUM lookup, policy routing and more together with SipX as the end user system.
I really like the flexibility of OpenSIPS, you can use that thing for basically
anyt
I tried once OpenSIPS as load balancer, it integrated nicely with SipX.
OpenSIPS is a very powerful tool (it means configuring it properly is a
challenge ;-)).
Regards,
Gabor
-Original Message-
From: Robert B [mailto:d...@spudland.com]
Sent: 08 December 2009 22:44
To: sipx-users@list.sip
Hi
Glad I could help.
In the Cisco router the following two debug commands are very useful:
For SIP debugging,
# debug ccsip messages
For ISDN debugging,
# debug isdn q931
This combined with the SIPVIEWER included in SipX should be enough to verify
that SIP signaling is successfully
moving
Give that man a prize.
I chose the wrong selection when the install asked me if I wanted the
sipXecs server to be the DNS server.
Thanks!
Mike Burden
Lynk Systems, Inc
e-mail: m...@lynk.com
Phone: 616-532-4985
-Original Message-
From: Picher, Michael [mailto:mpic...@cmctechg
2009/12/3 Gabor Paller :
> Where are the retransmission timeout values for SipX? (e.g. in what config
> file?) Also, if there is a JAIN-SIP expert listening, I have the same
> question for JAIN-SIP.
You can set the base timer interval on a per-transaction basis.
http://hudson.jboss.org/hudson/j
It is nice due to its organizational layout. To the people who comes from the
"Windows" world, the UI is welcoming. It reminds me of the MMC or management
console Microsoft provides which alot of good software company uses for their
UI management. In the IP world, the closest I've seen do a g
On Wed, 2009-12-09 at 14:24 +0100, Romano Ramdas wrote:
> Hello everybody,
>
>
>
> I’m a beginner with VoIP and SIP systems, and although I understand
> not much how to set up SipXecs and how to configure it, I fully
> understand the advantages that it will bring.
>
> In the last 3 months I lo
On Wed, 2009-12-09 at 13:03 +, Gabor Paller wrote:
> Update:
> Two important findings: the test calls were placed with the alias and
> not the domain (INVITE sip:4...@10.1.9.45 and not INVITE
> sip:4...@onrelay.local). This caused an extra iteration in the proxy and
> increased its memory/CPU u
Hello everybody,
I'm a beginner with VoIP and SIP systems, and although I understand not much
how to set up SipXecs and how to configure it, I fully understand the
advantages that it will bring.
In the last 3 months I look every where on the internet where to get good
information about SipXecs
Update:
Two important findings: the test calls were placed with the alias and
not the domain (INVITE sip:4...@10.1.9.45 and not INVITE
sip:4...@onrelay.local). This caused an extra iteration in the proxy and
increased its memory/CPU usage.
But after having corrected this problem, the bottleneck be
On Wed, 2009-12-09 at 13:36 +0100, Rene Pankratz wrote:
> Aren't there some phones, that are able to use 2 DNS A-Records for the
> domain?
Probably, but that doesn't mean that it's the right way to do it.
> But back to topic: Today I did another installation and again I had
> problems when adding
Aren't there some phones, that are able to use 2 DNS A-Records for the
domain?
But back to topic: Today I did another installation and again I had problems
when adding a server. Mappingrules.xml coult not be generated...
After a few tries and deleting and re-adding the server suddenly everything
w
On Wed, Dec 9, 2009 at 4:43 AM, Todd Hodgen wrote:
> It is nice that they summarize the calls in progress and the trunks used.
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Scott
> Lawrence
> Sent: Tues
It is nice that they summarize the calls in progress and the trunks used.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Scott Lawrence
Sent: Tuesday, December 08, 2009 7:10 AM
To: Jordan Turner
Cc: sipx-users@li
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