Mark update and try again... we did some changes to the core odbc
stuff today and waiting on the dust to settle.
Thanks,
Brian
On Nov 16, 2009, at 9:15 PM, Mark Campbell-Smith wrote:
> I installed libcurl4-openssl-dev, but this automatically removed
> libcurl4-gnutls-dev, which is required by
I installed libcurl4-openssl-dev, but this automatically removed
libcurl4-gnutls-dev, which is required by mod_dingaling. Now
mod_dingaling fails to build with:
Compiling mod_dingaling.c ...
mod_dingaling.c:309:78: error: macro
"switch_odbc_handle_callback_exec" requires 5 arguments, but only 4
gi
Hello folks!
I just wanted to let everyone know that there is a status update on the main
FreeSWITCH page. Here's a quick link for your convenience:
http://bit.ly/3RSY9F
The abridged version is this: we're working on it, there are some
outstanding JIRA reports that people need to review & test, a
On Mon, Nov 16, 2009 at 3:22 PM, wrote:
> I have three FreeSWITCH servers currently setup with perl modules using
> ESL to send call instructions and monitor events. On two of the servers,
> my modules execute without error, but on a third, I keep getting the
> following error:
>
> No matching f
Tim,
1.0.5 is coming soon... We were ready to release on the tuesday
morning we said but we woke up and Jira was flooded with tons of new
issues half of which we asked for more info on and the reporters
aren't responding. So the key is if you open a jira be ready to
respond because
I have three FreeSWITCH servers currently setup with perl modules using
ESL to send call instructions and monitor events. On two of the servers,
my modules execute without error, but on a third, I keep getting the
following error:
No matching function for overloaded 'new_ESLconnection' at
/usr/li
We are still working on 1.0.5. Right now the best place to be is that latest
trunk. More information is forthcoming...
-MC
On Mon, Nov 16, 2009 at 3:07 PM, Tim Uckun wrote:
> Where is 1.05? The trunk? Is trunk stable?
>
> Thanks.
>
> ___
> FreeSWITCH-u
Where is 1.05? The trunk? Is trunk stable?
Thanks.
___
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Hi!
I am trying to enable SSL support in FS. I have followed the wiki at
http://wiki.freeswitch.org/wiki/SIP_TLS
I already had libssl-dev installed, so I thought support should
already have been compiled into FS, however enabling
Internal_ssl_enable=true in vars.xml results in FS internal profil
Hi,
Since recently it's also possible to use lua *as* a dialplan:
http://wiki.freeswitch.org/wiki/Mod_lua#For_dialplan
regards,
Leon
On Mon, 2009-11-16 at 11:33 -0800, Michael Collins wrote:
>
>
> On Mon, Nov 16, 2009 at 9:36 AM, Jerry Richards
> wrote:
>
> I have a bit
Hello
thanks so much. The machines are on the same lan , 2 have static IP with
one on DHCP just for variation . I do get there errors on stating FS
1. Error stacksize too large 4194303 offers advise to run ./freeswitch
-wate
2. Error checking for PMP [GENERAL ERROR]
and
3. [WARNING] sofia_reg.
On Mon, Nov 16, 2009 at 9:36 AM, Jerry Richards
wrote:
>
> I have a bit of confusion about Lua scripting. When a script is invoked,
> should it always return an XML string that is used by FS? Or as in the
> case
> of dialplan examples, does it actually execute the dialplan (e.g.
> "session:answe
FYI,
I've added the skeleton of the agenda for this week's call:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_11_20
The agendas have been pretty light lately. I would like everyone to think
about questions that could be brought up for discussion. Also, I'd like to
take this time to say thank y
Hello Anthony,
I made a console trace today:
http://pastebin.freeswitch.org/11125
Different from the mail below, in this case A and C have voice.
Best regards
Peter
Anthony Minessale schrieb:
> if you provide a console trace of both situations with console
> loglevel debug and put them on pasteb
I have a bit of confusion about Lua scripting. When a script is invoked,
should it always return an XML string that is used by FS? Or as in the case
of dialplan examples, does it actually execute the dialplan (e.g.
"session:answer();")?
Best Regards,
Jerry
-Original Message-
From: Leon
On Sat, Nov 14, 2009 at 5:18 PM, Samuel Abekah-Mensah wrote:
> Hello
>
> Please pardon me if the solution to this is somewhere already that I
> have been unable to locate. I have just got a straight out-of-the-box
> build of FS. According to the wiki, I should be able to test using user
> IDs 1001
Hi Anthony,
Thanks for the input. I will try & reproduce the problem & give you
something more concrete to work with & log it in Jira.
Thanks again,
Michael
On Mon, Nov 16, 2009 at 5:25 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:
> That's a pretty small problem description to
I think you missed this step http://wiki.freeswitch.org/wiki/Dingaling#TLS
So your dingaling is failing to work properly... :P
/b
On Nov 16, 2009, at 9:45 AM, David Schwartz wrote:
> Thanks to everyone for all the help.
>
> I finally got gtalk to work in both directions - well almost.
>
> I hav
Thanks to everyone for all the help.
I finally got gtalk to work in both directions - well almost.
I have the gtalk client indicating that user 1000 1000 is trying to call him
only there is no button on the gtalk client to answer the call. (when I call
from a gtalk client to gtalk client there
That's a pretty small problem description to be so sure about something.
It would probably be better to capture some evidence of the exact problem
you are having since we are using computers and we need to see the computers
in action doing something specifically incorrect to diagnose any sort of
pr
Hi All,
I have an issue that when my call volumes on my FS IVR box > 30 calls DTMF
digits are lost (using RFC2833). It is definitely load related as it all
works perfectly under 30 calls.
Any pointers or a solution to the problem?
Thanks,
Michael
_
Just use the siemens to do the transfer. Hit the menu key, select
internal (if you want to do an siemens-siemens transfer), select the
extension, hit talk, then menu and choose conference or transfer. If
youw ant another extension, then choose external, dial the extension,
and the rest is the sam
Hi,
just a thing i noticed... the debug log and sip trace have different time
... one hour difference ... looks like UTC/GMT issue.
where do i set the time for siptrace correctly ?
2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:411
(sofia/external/00010038516659...@10.4.5.107:
I discovered that the problem was due to having the incoming PSTN line
connected to FS and the Siemens A580 base station. The Siemens A580
handsets were configured not to accept incoming calls from the PSTN but that
does not seem work. In any case, I have disconnected the PSTN line from the
Sie
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