Thats exactyl what I did.
UserName of the extension in this case is
200
Alias is the full pstn number.
Still, the call does not get completed.
Got to check the logs ...
Damned...
From the logs
INVITE 6409...@192.168.2.250:51154 tid=4c4a.540a0962.1 cseq=INVITE
Just for testing purposes,
I did enter the user-ID as an alias and ... worked like a charm, which at
least
Removes some of my headaches.
Thank you all for the great advice.
Now on to outbound calling.
-Ursprüngliche Nachricht-
Von: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Please correct me if I am wrong.
I believe that sipX is agnostic to which codec the endpoints negotiate
when the endpoints
are e.g. soft phones or SIP trunks (external phones). Therefore despite
sipX not supporting g729,
a g729 media stream can be relayed through sipX, to soft phones or SIP
Well, there is some information on hot to do that on the FS site. I've
never tried it. The problem you will have is with the media server
prompts.
We would use an SBC with transcoding to handle this normally.
On Thu, Jun 24, 2010 at 9:03 AM, Sven Evensen sven.even...@onrelay.com wrote:
Please
Yes this means all calls from that provider will hit that one user, no
matter what number was actually called.
On Thu, Jun 24, 2010 at 4:18 AM, maybelater maybela...@gmx.de wrote:
Just for testing purposes,
I did enter the user-ID as an alias and ... worked like a charm, which at
least
On Thu, Jun 24, 2010 at 4:06 AM, maybelater maybela...@gmx.de wrote:
Thats exactyl what I did.
UserName of the extension in this case is
200
Alias is the full pstn number.
Still, the call does not get completed.
Got to check the logs ...
Damned...
From the logs
INVITE
Hey Mike,
Here is some more information.
No cisco phones were used. I tried it with those phones:
Our Softphone Client (isPhone)
Polycom IP670
Had the same problem with both.
Whenever user A calls user B the call get's routed trough sipXbridge to a
sip trunk (Dialogic Diva Card).
Here you
If both are having the same exact problem, I would look to the firewall
(connection timeout sort of problem) or trunk provider.
Can you call internally without disruption?
Mike
From: cyrill.rei...@iscoord.com [mailto:cyrill.rei...@iscoord.com]
Sent: Thursday, June 24, 2010 7:27 AM
Can you detail the call? your sipxproxy.log is a 14MB text file.
Why is the call going through a siptrunk from sipxbridge? Normally if
dialogic (assuming this is the model using the ingate trunking software) is
doing siptrunking, why is sipxbridge involved in the call at all? Are either
of the
Ok. I am dredging this back up... sorry.
In 4.2 I can do Internet dialing by default using the default sbc,
sipXbridge-1. I do not have to enable Internet dialing by default to
dial by sip URI.
What I do find is that ISN dialing, which used to be defined allow
ISN dialing was located under
On 6/24/10 8:58 AM, Tony Graziano wrote:
What I do find is that ISN dialing, which used to be defined allow
ISN dialing was located under Domain or Internet calling. Now it is
under the registrar.
where is 'registrar'?
I don't remember doing anything to enable or disable ISN/SIP calling
I think ISN is a great idea. We implement it and so do a lot of
customers. I'm just trying to make it easier to enable for others. ISN
assignments are free, and very handy to dial a sip uri eqivalent from
a numeric keypad.
On Thu, Jun 24, 2010 at 9:56 AM, Michael Scheidell scheid...@secnap.net
I think we had tried that, not sure now, I'll have to look at the notes. Got so
involved in troubleshooting why we aren't able to get the logs alone that we
got sidetracked. We need to get back to this test, just haven't been able to
yet.
Mike
On Mon, 14 Jun 2010 16:35:26 -0400, WORLEY,
On 6/24/10 10:00 AM, Tony Graziano wrote:
where is 'registrar'?
I don't remember doing anything to enable or disable ISN/SIP calling
(you can sip me on my extention @, or any alias, including alphanumeric
aliases)
so you are saying this is a function of your ITSP, when you register
Paul Scheepens wrote:
Subject: RE: [sipx-users] CDR not working
Snip...
Raymond Dans wrote:
In an HA environment, there should be a connection established
between the Call Resolver on the primary and the Call Resolver Agent
on the distributed.
Have you turned on DEBUG logging
Thanks, I think I upgraded - just some seconds before - the end of april.
Will give this a try tomorrow, have to get home now.
Paul
DANS, RAYMOND (RAYMOND) rgd...@avaya.com wrote on 24-06-2010 17:18:45:
Snip...
Raymond Dans wrote:
In an HA environment, there should be a connection
No. If you dont use an ITAD it does not matter.
It's a global DB (you configure the records in their system after a
free application is approved). Optionally you can have the lookups
done in your DNS. There's a lookup and DNS component from their
registry when they host the records, and points
On 6/24/10 1:39 PM, Tony Graziano wrote:
No. If you dont use an ITAD it does not matter.
interesting. yes, I remember this from a while back, forgot all about it.
oh, I can't dial 1234*256
we got a 4 digit extension and 1234 belongs to the CFO :-(
It's a global DB (you configure the
Um. If you go to the registrar in sipx and turn on ISN dialing, you
can dial 1234*256. The * means @ and 256 will resolve to
loligo.com. That's what its all about!
On Thu, Jun 24, 2010 at 1:48 PM, Michael Scheidell scheid...@secnap.net wrote:
On 6/24/10 1:39 PM, Tony Graziano wrote:
No. If
gimme a break... are you really asking that? Where are all the
services in sipx listed? I know you know the answer.
On Thu, Jun 24, 2010 at 1:57 PM, Michael Scheidell scheid...@secnap.net wrote:
where is the registrar in sipx?
On 6/24/10 1:54 PM, Tony Graziano wrote:
Um. If you go to the
On 6/24/10 1:57 PM, Michael Scheidell wrote:
where is the registrar in sipx?
never mind, found it.
--
Michael Scheidell, CTO
Phone: 561-999-5000, x 1259
*| *SECNAP Network Security Corporation
* Certified SNORT Integrator
* 2008-9 Hot Company Award Winner, World Executive Alliance
On 6/24/10 2:00 PM, Tony Graziano wrote:
gimme a break... are you really asking that? Where are all the
services in sipx listed? I know you know the answer.
brain fried, looking for ^Registrar.. didn't see SIP registerar.
O
--
Michael Scheidell, CTO
Phone: 561-999-5000, x 1259
Coincidentally, you made my point that I was suggesting enabling that
should be in DIAL PLAN, it logically makes more sense (at least from
enable/disable) to be placed there.
On Thu, Jun 24, 2010 at 2:03 PM, Michael Scheidell scheid...@secnap.net wrote:
On 6/24/10 2:00 PM, Tony Graziano wrote:
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
[tgrazi...@myitdepartment.net]
Coincidentally, you made my point that I was suggesting enabling that
should be in DIAL PLAN, it logically
Which is why I was looking for input.
My post earlier today...
What I do find is that ISN dialing, which used to be defined allow
ISN dialing was located under Domain or Internet calling. Now it is
under the registrar.
Hurray it works! But, I am wondering why it is under registrar? I
On 6/24/10 2:31 PM, Tony Graziano wrote:
Coincidentally, you made my point that I was suggesting enabling that
should be in DIAL PLAN, it logically makes more sense (at least from
enable/disable) to be placed there.
I did find that in the 'advanced settings', I needed to put in something
Mine works without 012. [?]
Feel free to comment on the JIR. Though I hate to use prefixes that are
system specific, I never think its a good idea. I can dial 1234 it stays
local, I dial 1234*256, it knows its an ISN number, does the lookup and
routes the call. I don't think it should be that
On 6/24/10 2:59 PM, Tony Graziano wrote:
Mine works without 012.
Feel free to comment on the JIR. Though I hate to use prefixes
that are system specific, I never think its a good idea. I can dial
1234 it stays local, I dial 1234*256, it knows its an ISN number, does
the lookup and
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Scheidell
[list-s...@secnap.com]
I did find that in the 'advanced settings', I needed to put in something
(like recommended on isn site) a prefix
I don't see where licensing would be an issue: From mp3licensing.com If
your server will encode mp3 purely for internal use (i.e., you are not
planning to distribute mp3 codecs outside of your organization), you could
do so without needing to take our license. We would only assert our patents
if
OK I think I have figured out why this is happening and I think it's a
bug. What I think is happening is that sipX is processing the non-shared
branch gateways that match the branch of the calling user BEFORE
processing the shared gateways even though the shared gateways are
listed before the
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