On Wed, Jan 5, 2011 at 12:18 AM, Roy Walker wrote:
> Paul,
>
> I wish it was a configuration issue (I am crystal clear on how Caller ID is
> supposed to work in sipXecs now). I have tried EVERY possible configuration
> I can think of. I have deleted the Gateway, added it back... I can not get
>
Paul,
I wish it was a configuration issue (I am crystal clear on how Caller ID is
supposed to work in sipXecs now). I have tried EVERY possible configuration
I can think of. I have deleted the Gateway, added it back... I can not get
the transform or user specific Caller ID to work, it always jus
O.k. The only thing I would note is that user caller ID settings and transform rules are bypassed if you place your Caller ID string in the field indicated in the screen shot that you sent earlier.Its working on my system - default and user specified settings - with Vitelity just fine. Sounds lik
I have only used that (transform function) with physical gateways (PRI).
On Tue, Jan 4, 2011 at 10:06 PM, Roy Walker wrote:
> Paul,
>
> Yes the "Default Caller ID" option works. It is any user or transform
> settings that do not work, which is what I am trying to get working...
>
> Thanks,
> Ro
Paul,
Yes the "Default Caller ID" option works. It is any user or transform
settings that do not work, which is what I am trying to get working...
Thanks,
Roy
On Tue, Jan 4, 2011 at 8:55 PM, Paul Herron wrote:
> Roy –
>
>
>
> Try putting your caller ID in the “Default Caller ID” field for the
Roy –
Try putting your caller ID in the “Default Caller ID” field for the
gateway and leave all others blank. I do this on my 4.0.4 installation
with Vitelity as my ITSP and it works just fine. Per line/user caller
ID specifications work just fine as well. I have not used any transform
rules
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I don't know if you are just screwing with me or not. Yes,
the ITSP is pu
If you do a simple search of the attached file looking for "6102" you find it
multiple times. And not one time does it have a 1 in front of it. At least
not from what I could find. It's pretty basic, if sipXecs was putting that one
is for any reason, you would find it in your capture from the
I think you have a problem with the gateway settings. I don't use vitelity,
but I use other ITSP's who let me send what I want, and it works.
I do think the suggestion to remove the gateway, etc. and recreate it is the
best approach. I don't think you believe me, and that's fine, but you are
not g
I for one am lost.
You have manually mangled the log file. What is it you are trying to say?
I WOULD NOT use the TRANSFORM. I WOULD NOT use the gateway as you have
configured it. Remove the gateway. Remove the default callerid. Set the user
callerID to exactly the 10 digits you are trying to send
Thanks !
Gabrial Casey
Telecommunications
Franklin American Mortgage Company
501 Corporate Center Dr.
Suite 400
Franklin, TN 37067
Direct:615-468- 2945
Cell: 615-852-5015
Fax: 615-628-5698
Email:gca...@franklinamerican.com
From: "Nathaniel Watkins"
To: "Discussion list for user
They put it up for a vote for stack inclusion:
http://bitnami.org/product/sipxecs
--martin
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>>> Douglas Hubler 01/04/11 11:36 AM >>>
>>
>> I just pushed changes incl. mock to 4.4 release branch in git. I
>> assume you've followed my notes about OBS being unusable in the end.
>> Are there mock templates for suse, I cannot seem to find them. Cross
>> distro compiling has been a godsend.
Henry,
the timeout is something you can set, not sure what the default is,
but it was low enough for me to notice, also if the slave DNS server
is restarted, and it has no master to connect to, it stops serving the
zone! and yes sync is just files.. I was using heartbeat for high
availability DH
Just got this...passing it along :)
-Original Message-
From: VoIP.ms [mailto:nore...@voip.ms]
Sent: Tuesday, January 04, 2011 2:14 PM
To: Nathaniel Watkins
Subject: Important Security Tips
Dear Nathaniel Watkins,
We are contacting you with some suggestions on how to improve the security
On Tue, Jan 4, 2011 at 11:07 AM, Douglas Hubler wrote:
> On Tue, Jan 4, 2011 at 8:43 AM, Matt White wrote:
> Douglas Hubler 12/30/10 1:55 PM >>>
>Building using mock is currently on 4.5 codebase so instead of
>backporting it, I recommend we cut the new 4.4 release from 4.5 HEAD
>
On Tue, Jan 4, 2011 at 8:43 AM, Matt White wrote:
Douglas Hubler 12/30/10 1:55 PM >>>
Building using mock is currently on 4.5 codebase so instead of
backporting it, I recommend we cut the new 4.4 release from 4.5 HEAD
and begin to stabilize this branch.
>
> I second the recommend
Ok thanks,
Our client uses the failover setup as well, but it's setup a little different
than yours.
What would this time-out for the slave DNS be? Would it be days/weeks/months?
And syncing is just keeping the files the same? So wouldn't be a problem I
guess?
Henry
-Original Message-
Hi Henry,
We run our servers and the larger gateways all on one vlan and phones
are on other vlans based roughly on their native departmental subnets.
Cisco does per-vlan spanning tree which is handy in the case that
someone loops a switch, they only take down that vlan.
For Cisco we use the "ip
Was on holiday and had a lot of mails to catch up
Here my answers to the original questions:
Paul
P.S. I am not in the rest of the week, so if you have questions dont
expect a reply before monday.
Victor Williams wrote on 17-12-2010 23:40:30:
> I have successfully gotten this working
>>> Douglas Hubler 12/30/10 1:55 PM >>>
>>>Building using mock is currently on 4.5 codebase so instead of
>>>backporting it, I recommend we cut the new 4.4 release from 4.5 HEAD
>>>and begin to stabilize this branch.
I second the recommendation to stabilize 4.4 without mock for now.
We are cl
>>> "Henry Dogger" 01/04/11 3:45 AM >>>
>>>Hi Kyle,
>>>
>>>Thanks for the quick reply :)
>>>
>>>So you have DHCP and DNS on both sipx boxes? It would seem to me you
>>>only need one DHCP right?
>>>My DNS, DHCP, NTP and TFTP are all on the master server.
If DNS is only on your master node, and the
The fact that a "1" is being appended is bothersome. I would delete and
recreate the gateway.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8
Hi Kyle,
Thanks for the quick reply :)
So you have DHCP and DNS on both sipx boxes? It would seem to me you
only need one DHCP right?
My DNS, DHCP, NTP and TFTP are all on the master server.
The setup is not situated at our office but with a client, so I need to
check how the IP-helper is setup.
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