Hi noob and Michael, thanks for your answers, I'll try to use
mod_xml_curl.
Hi Henry, http://wiki.freeswitch.org/wiki/Mod_xml_curl has mentioned, FS
will post a request to webserver when it get a registration request. you can
refer to the doc for more detail.
Lei.Tang
lei.tl...@gmail.com
I have used a SIP provider for more than a year. A few days ago, he
said he was moving to a new server, and asked me to reconfigure. I
did, and everything seemed to work fine, until I did an outgoing call
to an external telephone. I found out I had no audio, in neither
direction. Incoming
2009/10/27 lakshmanan ganapathy lakindi...@gmail.com:
Hi all,
I've done experimenting with the uuid_displace and mux. What mux does is
playing a file when the conversation is also happening. But I've a different
requirement.
Is posible to use uuid_displace in dialplan ? i want to do
Thanks Brian, it works now. I'll try to learn to say NO next time :)
/Peter
-Ursprungligt meddelande-
Från: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 26 oktober 2009 19:40
Till:
Hi all, I run FS on a machine with two net interface, each interface has a
ip addr, one of the them connect to public network(has ip addr A), the
other connect to a private network(has ip addr B), FS server as a SIP
server for public through A, all outbound call will bridge to a softswitch
in
Hi,
how can i call reload from managed code? and i called
Csharp_switch_xml_open_root, this is working first time, then onwards it is
not working.
is this the correct function to call reload?
--
Srinivasula Reddy K
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Hi,
when i am calling switch_xml_open_root(1,err) . i am getting this warning
message.
HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap( 0016,
100E9FD0 ). can any know. please help me.
Thanks
Srinivasula Reddy K
___
I tried the uuid_displace by using the Event Socket Outbound.
Not tried within dialplan, and I'm not sure whether that could be done in
dialplan itself.
On Tue, Oct 27, 2009 at 12:37 PM, Dome Charoenyost d...@tel.co.th wrote:
2009/10/27 lakshmanan ganapathy lakindi...@gmail.com:
Hi all,
Try setting ext-rtp-ip and ext-sip-ip on both profiles.
On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang lei.tl...@gmail.com wrote:
Hi all, I run FS on a machine with two net interface, each interface has a
ip addr, one of the them connect to public network(has ip addr A), the
other connect to a
Make sure you let their media IPs through your firewall. Also, if you
are behind a NAT, check you have things passing to the correct
internal address.
On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold i...@myrvold.org wrote:
I have used a SIP provider for more than a year. A few days ago, he
said
Sorry, trunk does not compile on win7, here are the details:
rev.15247
---
Microsoft Visual C++ Debug Library
---
Debug Assertion Failed!
Program: ...ev\Projects\External\FreeSWITCH\Original\Debug\FreeSwitch.exe
File:
won't compile or won't run?
maybe you should try rebuilding it.
On Tue, Oct 27, 2009 at 9:55 AM, Maciej Aniserowicz
maciej.aniserow...@gmail.com wrote:
Sorry, trunk does not compile on win7, here are the details:
rev.15247
---
Microsoft Visual C++ Debug Library
Is it necessary for you to call the reload function directly? Why not
make an api call using switch_api_execute and the command reloadxml.
On Tue, Oct 27, 2009 at 4:32 AM, srinivasula reddy
srinivas.ksvre...@gmail.com wrote:
Hi,
how can i call reload from managed code? and i called
The issue is you're not behind nat-pmp or upnp so it can't figure out
what your public IP is... you'll have to fill that in manually or
enable those options on your router if possible.
/b
On Oct 26, 2009, at 10:03 PM, Lars Zeb wrote:
I haven’t changed anything since v15183, where it
The server is on a public IP, so there is no nat issue here.
I can also see the rtp messages on wireshark starting just after the
183 Session Progress message on the server, but just in one direction,
coming in to the server.
So it looks like Freeswitch is stopping the rtp.
Is this because
Thanks for the reply, Brian.
Did something in FS change between v15183 and v15225 to make this occur? I
ask because this same configuration worked OK in the earlier version.
Lars
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org]
Let me review that and see I can't off hand think of anything that
would cause that.
/b
On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:
Thanks for the reply, Brian.
Did something in FS change between v15183 and v15225 to make this
occur? I ask because this same configuration worked OK in
No, the IP address the media originates from does not need to be tied
to the SIP IP address. Can you send a Wireshark capture taken on the
FreeSWITCH server of both call legs? Or, if you can, pastebin a debug
log from FreeSWITCH console with sofia loglevel set to 9 and siptrace
on for any Sofia
Is the following the correct command to force registration of an endpoint
below which is not showing up in 'sofia status profile internal' currently,
but when it does it looks like the following?
Call-ID:mjkg8c3zlqwjzfd.hrt5
User: 1...@192.168.10.29
Contact:user
Greetings all!
I just wanted to say thank you to those who have been contributing to the
documentation and cleanup efforts. Please keep up the good work. FYI, we
still have lots of janitorial stuff to do. Just keep checking the latest
conf call agenda for updates. This week's agenda is here:
From FreeSWITCH you can't force and endpoint to register... thats the
endpoints job... what you're doing is just flushing the reg out of
FreeSWITCH so now it can't find the endpoint... Try adding a reboot
on there maybe i'll reboot the device and cause it to register again...
/b
On Oct
Did the voting booth close? I was unable to vote. I'm not sure what link
to click and I have had some strange issues with my FS account today.
I would be interested in paid training. Do you have plans for offering a
training session at your locale? Or would you travel onsite to provide
A little off-topic, but since call-capacity is the subject, what are
people using to analyze their CDR's to discover this? I'm handling
about 30k calls per day but have only a bandwidth-based guesstimate of
the peak number of concurrent calls I'm handling.
If there's an open source solution,
I'm fond of Vqmanager from ManagEngine. It is a passive SIP monitor. I.E.
you mirror the ports that your FS or Asterisk boxes, and VQmanager sniffs
the mirrors, tracking all sorts of good data.
You can install it on a Centos box, and get a free trial.
Cliff,
Try using xml_rpc ... status or show channels will give you what you need.
SDR
Cliff Wells wrote:
A little off-topic, but since call-capacity is the subject, what are
people using to analyze their CDR's to discover this? I'm handling
about 30k calls per day but have only a
On Tue, Oct 27, 2009 at 12:17 PM, Jerry Richards jerry.richa...@teotech.com
wrote:
Did the voting booth close? I was unable to vote. I'm not sure what link
to click and I have had some strange issues with my FS account today.
I would be interested in paid training. Do you have plans for
You probably should not be calling that function, what are you trying
to do?
Mike
On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote:
Hi,
when i am calling switch_xml_open_root(1,err) . i am getting this
warning message.
HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap
On Tue, Oct 27, 2009 at 2:31 PM, Michael Jerris m...@jerris.com wrote:
You probably should not be calling that function, what are you trying to
do?
Mike
On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote:
You might also want to move this conversation to the -dev list because it's
a
On 2009-10-23 15:14 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC
TC so the real solution is to implement a check for CallProceeding , Progress
TCand Facility message whether it has a faststart element included. It it is
TCtrue than you might start pre_answer.
TC
TC
TCalso, i
Thanks Eliot, It works.
2009/10/27 Eliot Gable
egable+freeswi...@gmail.comegable%2bfreeswi...@gmail.com
Try setting ext-rtp-ip and ext-sip-ip on both profiles.
On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang lei.tl...@gmail.com wrote:
Hi all, I run FS on a machine with two net interface, each
you should have just needed to set the rtp-ip and the sip-ip on both
profiles to their values and it would have worked fine... their would
have been no need to set the ext-*-ip equiv.
/b
On Oct 27, 2009, at 8:14 PM, Lei Tang wrote:
Thanks Eliot, It works.
Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when I set the
ext-rtp-ip, it works fine.
2009/10/28 Brian West br...@freeswitch.org
you should have just needed to set the rtp-ip and the sip-ip on both
profiles to their values and it would have worked fine... their would
have been
Ok then something is broken badly... and that makes NO sense. Because
it works on my box.
/b
On Oct 27, 2009, at 8:34 PM, Lei Tang wrote:
Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when I
set the ext-rtp-ip, it works fine.
2009/10/28 Brian West br...@freeswitch.org
you
Hi mike,
thank for your reply.
i am trying to call that function from swig.cs. its working fine first
time with the warning information, then onwards it is not working.
is there other way can i call reload function from freeswitch.managed code.
any help
srinivas
On Wed, Oct 28, 2009 at 3:01
Just like you can call conference list for conferences, is there a way to
retrieve the profile state of a conference member using the cli or xml/rpc?
Lon
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just api execute reloadxml
Mike
On Oct 28, 2009, at 12:37 AM, srinivasula reddy wrote:
Hi mike,
thank for your reply.
i am trying to call that function from swig.cs. its working fine
first time with the warning information, then onwards it is not
working.
is there other way can i
i tried switch_api_execute its working fine, thanks rupa
On Tue, Oct 27, 2009 at 8:48 PM, Rupa Schomaker r...@rupa.com wrote:
Is it necessary for you to call the reload function directly? Why not
make an api call using switch_api_execute and the command reloadxml.
On Tue, Oct 27, 2009 at
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