It comes to mind that this approach may have strange effects in
transfer scenarios. What is the expected behavior in regards to the
permissions on these calls when ttansfered.
Mike
On Aug 6, 2008, at 3:54 PM, "Michael Collins" <[EMAIL PROTECTED]>
wrote:
I just grepped the entire source
He is saying to add the var to the user in the variables so when we
auth that user, the variable is set.
Mike
On Aug 6, 2008, at 11:58 AM, UV <[EMAIL PROTECTED]> wrote:
I have to agree with Roberto on not understanding what you’ve done,
Brian.
Is toll_allow an undocumented channel varia
It should already work with what is in opezap using boost but we have
not done the testing on it yet. If someone has cards and a line I
would love to hear results.
Mike
On Aug 6, 2008, at 1:24 PM, Michal Bielicki <[EMAIL PROTECTED]
> wrote:
It would or it will ?
On Aug 6, 2008, at 8:14
Does this work? What about when the phone IS behind nat? We should
use the same transport for the options that we got on. The register or
invite. If that is not the case can you please file a bug on
jira.freeswitch.org with all the traces so we can look into this
further?
Mike
On Aug 6
I have used ixia gear in the past to do this sort of testing but it's
not cheap. Do you have access to any commercial test gear?
Mike
On Aug 8, 2008, at 7:38 PM, Brian West <[EMAIL PROTECTED]> wrote:
> I think this one has rtpecho on. You really can't do pcap replay sipp
> crashes before you
Eyebeam works well on mod_conference as does the grandstream video
phones.
/b
On Aug 7, 2008, at 10:33 PM, David A. Horner wrote:
Hey everyone,
So I've been thinking about how to get video conferencing going with
freeswitch.
When I first started, I was thinking about developing a video
OpenVZ is a zero issue it'll work perfect under OpenVZ.
I wrote this up about it http://wiki.freeswitch.org/wiki/FreeSWITCH_in_OpenVZ
/b
On Aug 8, 2008, at 5:23 PM, Tamas wrote:
>
> I've missed one fact: FS runs in OpenVZ guest.
> I don't know whether it is important or not
>
> t38 reinvit
I think this one has rtpecho on. You really can't do pcap replay sipp
crashes before you can really get it up to speed.
/b
On Aug 8, 2008, at 7:23 PM, Brian B wrote:
> Thanks for the pointer.
>
> Does that pipe any voice over the channels? We're testing a very
> unique configuration of mod
On Aug 8, 2008, at 7:22 PM, Lee JJ wrote:
Dear All Guru :
1.
I am curious what's the relation between alias "name", "type" ?
Does it affect the dialplan routing call ?
You have profile, gateway and alias. You can setup profiles with
multiple aliases. The default comes out of the box wit
Dear All Guru :
1.
I am curious what's the relation between alias "name", "type" ?
Does it affect the dialplan routing call ?
How to know more inside of this ?
[INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 1526->[EMAIL PROTECTED]
^
F
Thanks for the pointer.
Does that pipe any voice over the channels? We're testing a very unique
configuration of mod_conference, and if the lines are all quiet and don't
hit the Energy thresholds, I'm afraid the test won't meet our needs.
On Fri, Aug 8, 2008 at 5:02 PM, Brian West <[EMAIL PROTEC
www.freeswitch.org/eg/load_test/ thats what we have used for testing.
/b
On Aug 8, 2008, at 6:36 PM, Brian B wrote:
We're wanting to test our very particular FS conference
configuration, 1st on a small VPS slice and then on a dedicated
server.
To do that, we have written our requirements
We're wanting to test our very particular FS conference configuration, 1st
on a small VPS slice and then on a dedicated server.
To do that, we have written our requirements for a testing script, to
generate outbound SIP calls.
Is anyone interested in getting involved? This could be paid work, o
I've missed one fact: FS runs in OpenVZ guest.
I don't know whether it is important or not
t38 reinvite seems to be ok.
Regards,
Tamas
Michael Jerrie írta:
> This should work fine, just as well ad bypass_media except it can also
> handle nat. What exactly is not working? Is it fail
This should work fine, just as well ad bypass_media except it can also
handle nat. What exactly is not working? Is it failing on the t38
reinvite or negotiaoton or is that part working fine but the fax
machine is still not syncing up or is a fax starting then failing part
way through?
M
Hello,
I'm trying to make T.38 SIP working:
PSTN-->Cantata IMG-->FS--->CW
Unfortunately this does not work. I've set both the late negotiation and
proxy_media.
When I leave out FS from the chain, the fax receipt works well, also
when I set bypass_media in FS, it works too, so something might be
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