On Mon, May 31, 2010 at 6:15 PM, Sven Evensen wrote:
> Hi Ranga,
>
> I did not quite understand the conclusion. Where is the DNS fault? In sipX?
> And what is it actually causing?
>
> Why does sipX ignore 200 OK after two 401?
How do you conclude that it is being ignored?
> Why does sipX sto
Is there any guide about how to open a JIRA in sipx ? I never did it before
...
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Monday, May 31, 2010 5:58 PM
To: Wen Jun
Cc: thod...@verizon.net; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-user
portahk.cordiaip.netx
It can't resolve the A record for the above entry. If it can't find it, it
can't register to it. Check the DNS resolver for your amazon host, you might
also check the dns timing (latency) from your host there.
Tony Graziano, Manager
Telephone: 434
Hi Ranga,
I did not quite understand the conclusion. Where is the DNS fault? In sipX? And
what is it actually causing?
Why does sipX ignore 200 OK after two 401?
Why does sipX stop registration of that trunk?
Regards,
Sven
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.co
On Mon, May 31, 2010 at 2:33 PM, M. Ranganathan wrote:
> On Mon, May 31, 2010 at 1:55 PM, Sven Evensen
> wrote:
>> Hi Ranga,
>>
>> The problem is a little more subtle, I will try to explain. And I will send
>> you a snapshot too.
>>
>> sipX sends REGISTER cseq=1.
>> after 500ms sipX resends REG
On Thu, May 27, 2010 at 7:56 AM, Matt White wrote:
> I have a customer inquiring about the ability for some sort of sip based
> door lockie the ability to unlock a door by using a key code on a phone.
>
> I did find one here: http://www.abpsec.com/blog/open-doors-with-sip/
>
> Looks neat. Ha
I understand that, and am inputing the dial string correctly, but it
seems any call that requires permissions isn't working with tel:
forwarding. I will do further testing tomorrow.
On 05/31/2010 10:35 AM, JOLY, ROBERT (ROBERT) wrote:
>> OK I have found this only works on internal calls meaning:
On Mon, May 31, 2010 at 1:55 PM, Sven Evensen wrote:
> Hi Ranga,
>
> The problem is a little more subtle, I will try to explain. And I will send
> you a snapshot too.
>
> sipX sends REGISTER cseq=1.
> after 500ms sipX resends REGISTER cseq=1.
> a few ms later 401 cseq=1 is received for first REGI
Hi Ranga,
The problem is a little more subtle, I will try to explain. And I will send you
a snapshot too.
sipX sends REGISTER cseq=1.
after 500ms sipX resends REGISTER cseq=1.
a few ms later 401 cseq=1 is received for first REGISTER.
sipX sends new REGISTER with nonce cseq=2.
sipX receives 401 c
If sipxbridge sees two 401 or 407 for two successive REGISTER
transactions that implies it tired to authenticate, failed, tried
again, failed and hence there is a problem with the credentials. If
such is the case, then sipxbridge will retry again in 60 sec.
This has nothing to do with the 500 ms.
You can rule out the caller-id header brokenness by setting the SPA to not
provide caller ID (or to provide a static caller ID) and seeing if it works.
If the corrupted caller-id headers are the issue you're having, you can look on
the JIRA for a patch I made to work around this
(http://track.s
> thanks,
> I'll try it out an let you know.
>
> It matters if you have (theoretical speaking) the remote user
> trying to call an outside number, then the media goes like this :
>
> remote user internet - sipx -- internet --
> voip provider
>
> and i want the media to go dire
(and like the click-to-call application, a "dash" in the number will fail
the call)
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: 4
>
> OK I have found this only works on internal calls meaning:
>
> set forward to outside number (eg 5551212) for IM user (lets
> say 6001) call from internal extension to 6001 - the call is
> forwarded as it should be call from outside line to 6001 -
> the call is not forwarded
>
> Could thi
I have a hunch that you may need to look at the source files.
From: Irena Dolovčak [mailto:irena.dolov...@gmail.com]
Sent: Monday, May 31, 2010 10:18 AM
To: Tony Graziano
Cc: Paul Herron; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Remote office problem
hmm..
so is there a wa
hmm..
so is there a way to turn off the * *?
or was that just an option before?
*
*
2010/5/31 Tony Graziano
> All the xsd files are in: /usr/share/sipxecs/schema
>
> However, there is no sipxbridge.xsd file there.
>
> 2010/5/31 Irena Dolovčak :
> > Hi Paul, thanks for your answer.
> >
> > I have
I have never used one, but am considering it.
There are "two parts" to this to function properly.
1. Assuming the SIP phone registers properly, they simply "autodial"
an extension (hunt group, etc.). Once the call is answered the live
person at the other end can choose to open the door with a key
You need to explain what kind of phone it is, because it does make a
difference how to address this.
You can adjust these settings for remote users with a group, but not
all phones have the same options.
You need to look at the "keepalive" settings, and it "matters" whether
the phone is communica
We are having an unusual problem with our phones not re-registering after it
expires. It has happened several times before and always during the weekends.
I'm wondering if has anything to do with no phones being used for an extended
amount of time causing this. For example, this being a long h
All the xsd files are in: /usr/share/sipxecs/schema
However, there is no sipxbridge.xsd file there.
2010/5/31 Irena Dolovčak :
> Hi Paul, thanks for your answer.
>
> I have read the post that you have send me, but I cannot find the files..
> Where are they? at least the sipxbridge.xsd I can't fin
You can create a "speed dial with presence" for the call park
locations on a Polycom 650. Simply blind transfer to the preset button
the park it, etc.
On Mon, May 31, 2010 at 3:03 AM, Hiral Patel wrote:
> Hi There,
>
>
>
> Has anyone configure the Polycom 650 with shortcuts so phone users can pla
A JIRA with the system snapshot and log files would be the best next step.
On Sun, May 30, 2010 at 10:41 PM, Wen Jun wrote:
> I turned on the proxy log in debug mode and found NAT was referred by
> SipXProxy.
>
> Pls let me know if you like to receive the whole logs.
>
> _
Hi Paul, thanks for your answer.
I have read the post that you have send me, but I cannot find the files..
Where are they? at least the sipxbridge.xsd I can't find.
And in the file sipxbridge.xml, I can't see anything about
..
Want somebody give me a hint?
*It is called Look in the sipxbridge.x
The other side of this question is that Audiocodes gateways can handle those
"status 407" messages.
Try the following:
Configure a user, say M1000, on the sipx.
On the mediant side configure user (M1000) for "whole gateway registration",
but don't enable registration. Then mediant will use that us
Hi There,
Has anyone configure the Polycom 650 with shortcuts so phone users can
place and retrieve parked calls with dialling a code? If so, what did
you do and how?
We would like to setup the Polycom phone to have line buttons configured
4 call short cuts.
Any input would be great!
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