After further checking, it does not seem like the authentication after
the challenge is being sent...
Are there any other settings I should be aware of other than placing
the file in external and setting register to true?
Scott
2008-12-18 13:32:28 [NOTICE] sofia_reg.c:265 sofia_reg_check_gat
I have a click2click app that originates both legs of the call. After
the call is bridged if you press the # key it records the call. Only
the original person should be able to click # to enable records. That
is why i want to filter dtmf on 1 leg of the call.
Thanks,
Stephen C
-All of my email add
By ignore do you mean filter out? Or do you mean don't do anything but
do audiblize the tones? Do you have some sort of application that does
something with dtmfs?
-MC
Sent from my iPhone
On Dec 17, 2008, at 10:00 AM, "stephen at stephenjc" wrote:
> I have a click to click system written i
Hi,
I have multi domain, multi tenant setup configured and working.
Did you add something like
to one of the profile configs for multi-domain so FreeSWITCH can look its
configs for those domains?
Also, check if you specified domain_A for "domain_name" param in the
domain_A.xml file.
D
Hi,
Fresh freeswitch user. Installed freeswitch with default installation a
week ago. Need more information on multi domain usage. Followed the wiki
pages with multi tenant examples. All working well but I do have a
problem when calling to an internal extension.
Scenario:
domain_A created by
Seems pretty sleazy to me... what are they going to commercialize the
results?
/b
On Dec 17, 2008, at 9:41 AM, freeswi...@davidnicol.otherinbox.com wrote:
> http://www.google.com/search?q=cisco+linux+contest
>
>
> Although cisco already does VOIP stuff so they might have trouble
> awarding p
I have a click to click system written in javascript, so i call out
both legs of the call then bridge them. I am looking for a way to
ignore dtmf tones on 1 leg of the call.
Thanks,
Stephen C
-All of my email addresses go to the same place
-Save Paper, think before you print.
___
Helmut,
Can you turn on full debug and capture the output? It's a lot so put
it in a pastebin.
-MC
Sent from my iPhone
On Dec 17, 2008, at 7:30 AM, Helmut Kuper
wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi Anthony,
>
> thx, but that doesn't work very good. Outgoing call
FreeSWITCH already logs into your jabber server as a component if you
cant communicate with other domains then your jabber server is not
configured correctly.
/b
On Dec 17, 2008, at 11:15 AM, Kirk Bateman wrote:
Kriko,
I have been looking at the same sort of thing, but I'm planning to
i
Kriko,
I have been looking at the same sort of thing, but I'm planning to implement
an ejabberd bot component (so I can hopefully use the new mod_erlang_event
freeswitch interface).
It seems to me that bits of the current dingaling / jingle interface are
having problems, like not liking sending m
FreeSWITCH/openzap is completely uninvolved at this point, you might
try asking on the zaptel mailing lists?
Mike
On Dec 17, 2008, at 5:17 AM, fidibus83 wrote:
I did a reinstall but there is the same error!
Is there something else I can do to remove the error?
__
I think the best way to confirm all this is to load a full pcap in
wireshark and have it pull the wav file of the individual audio
streams to see what is going on.
Mike
On Dec 17, 2008, at 3:06 AM, mszla...@aol.com wrote:
Hi Mike,
That does get the audio go between the softphone and the a
On Dec 17, 2008, at 2:40 AM, Jason White wrote:
> The code in bind6only_check in libs/sofia-sip/libsofia-sip-ua/tport/
> tport.c looks
> correct to me, but I can't find where the result is tested (it's in
> mr_bindv6only). When bind6only_check() is called in
> tport_bind_server(), the
> return
http://www.google.com/search?q=cisco+linux+contest
Although cisco already does VOIP stuff so they might have trouble awarding prizes to a technology which would compete with themselves, but what are they expecting, putting Linux boards in Cisco backplanes?
___
I have an idea which is takes too many characters for irc.
I'm relatively new to telephony and such stuff, I managed to get
freeswitch running, but I don't fully uderstand
my problem in detail and how to solve it, so I need a bit of directions.
Briefly, my idea is to have a jabber contact, which
On Dec 17, 2008, at 10:05 AM, Tamas Cseke wrote:
> Helo,
>
> with register-proxy registrations use SRV.
> but it I use register=false param, and dial the gw it lookup only A.
>
> I figured out if I don't specify the port in proxy param the SRV
> lookup
> is working,
> but if I put ":5060" it do
Helo,
with register-proxy registrations use SRV.
but it I use register=false param, and dial the gw it lookup only A.
I figured out if I don't specify the port in proxy param the SRV lookup
is working,
but if I put ":5060" it doen't work. So it is a problem if I don't want
to use the default po
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Anthony,
thx, but that doesn't work very good. Outgoing calls ring only once and
then this error rises in console:
2008-12-17 16:23:19 [DEBUG] Span:0 Q.931() Sending message to Layer4
(size: 103)
2008-12-17 16:23:19 [DEBUG] ozmod_isdn.c:320 zap_is
On 12/17/2008 8:24 AM, Carole O. wrote:
> It would be unique you are right but I am not sure I can get its value if A
> puts the call on hold, calls C and wants to add it to the conference whose
> name dependent of the uuid of another session.
> I think if I use ${uuid} to add C I will have the uu
it might not.
try putting the value in register-proxy as well
sip:host.tld
On Wed, Dec 17, 2008 at 8:49 AM, Tamas Cseke wrote:
> Hello,
>
> We'd like to use DNS SRV for failover.
>
> if we are bridge sofia/profile/whate...@domain.with.srv it works perfectly
> but with gateway wich has this reco
try
in the in openzap.conf.xml
On Wed, Dec 17, 2008 at 8:53 AM, Helmut Kuper wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi,
>
> I updated the jira bug. I did a Q931/Q921 trace. Currently there is no
> direct hint, that FS is doing something wrong. NT side is just not
> anwe
you could make up a uuid just for the conference name in the original call
Now this channel and any other channel created by this channel will inherit
this var
On Wed, Dec 17, 2008 at 8:24 AM, Carole O. wrote:
>
> It would be unique you are right but I am not sure I can get its value if A
>
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I updated the jira bug. I did a Q931/Q921 trace. Currently there is no
direct hint, that FS is doing something wrong. NT side is just not
anwering the SETUP of FS, buuut, I was asked if FS is able to allow NT
side the channel management, so NT say
Hello,
We'd like to use DNS SRV for failover.
if we are bridge sofia/profile/whate...@domain.with.srv it works perfectly
but with gateway wich has this record in its proxy parameter it doesn't
work.
Once we set up an A record too it works, so we assume dialing gateway
doesn't use SRV records.
I
ok
Thanks a lot,
Carole
Anthony Minessale-2 wrote:
>
> I said i unblocked the ones in mod_commands
>
> mod_conference was it's own module.
> I changed it to work in latest trunk as well.
>
>
>
> On Wed, Dec 17, 2008 at 4:15 AM, Carole O. wrote:
>
>>
>> (I have just read the post again, I
It would be unique you are right but I am not sure I can get its value if A
puts the call on hold, calls C and wants to add it to the conference whose
name dependent of the uuid of another session.
I think if I use ${uuid} to add C I will have the uuid of the session
between A and C and not A and
I said i unblocked the ones in mod_commands
mod_conference was it's own module.
I changed it to work in latest trunk as well.
On Wed, Dec 17, 2008 at 4:15 AM, Carole O. wrote:
>
> (I have just read the post again, I have written application="bind_meta_app" data="1 a s conference::conf1 lock"
On 12/17/2008 7:34 AM, Carole O. wrote:
> My main problem is the name of the conference. Since everybody should be
> able to convert a simple call into a conference, the conference's name has
> to be unique each time. I have chosen to make it dependent on the caller
> number which is not perfect be
thankz!
ill set my openldap to provide these information..
but these about these binding settings... where should i set them?
best regards
John Skopis (Lists) wrote:
vinicius wrote:
hi ppl.. i tried to find something at google, but i couldnt manage to find
anything.
i still don
Hello,
I have done a small change in my dialplan which works but since I am new
with FreeSWITCH I was wondering if this solution goes with the philosophy of
the software or if it is absurd and there is a solution more adapted .
I try to reproduce the following functionality:
"A and B are on a si
I did a reinstall but there is the same error!
Is there something else I can do to remove the error?
_
Von: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Im Auftrag von
Michael S Collins
Gesendet: Dienstag, 16. Dezember 2008 16:53
(I have just read the post again, I have written but I
meant please don't pay attention for that, I made the mistake when I have
copied it in the post, not in the configuration.)
Carole O. wrote:
>
> Thanks, this works fine.
>
> But I try to use some other API commands and something goes wro
Thanks, this works fine.
But I try to use some other API commands and something goes wrong: I would
like to be able to use the API commands for the conference like lock,
unlock, say, etc... from the dialplan.
I try to add in my dialplan
but it did not work, I presse F8 and I have got:
2008-12
Hi Mike,
That does get the audio go between the softphone and the application (Voxeo's
Prophecy ASR) "around" FreeSwitch but I would like the audio going "through"
FreeSwitch. I plan to do something to it before passing it on.
Support from Voxeo had this to say about the "bypass media" settin
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