lakshmanan ganapathy lakindi...@gmail.com wrote:
Thanks for your replay. I don't know what is latest trunk. Is it latest
version? I'm using freeswitch 1.0.4.
It's the latest version from the svn repository. Use svn checkout, then
compile it as documented on the wiki.
Nice, so I just rename the default to foo.org and bar.org and I put the
users I want inside them?
On Wed, Sep 23, 2009 at 5:42 AM, Brian West br...@freeswitch.org wrote:
You don't have to think about it with proper DNS it all just magically
happens.
/b
On Sep 23, 2009, at 12:22 AM, Diego
s/rename/copy/
On Wed, Sep 23, 2009 at 6:03 AM, Diego Viola diego.vi...@gmail.com wrote:
Nice, so I just rename the default to foo.org and bar.org and I put the
users I want inside them?
On Wed, Sep 23, 2009 at 5:42 AM, Brian West br...@freeswitch.org wrote:
You don't have to think about
Hi all,
I've written a C# module for FS that creates conference dialplans on the
fly. From my limited understanding the easiest way to do this is by writing
XML to the directory: conf/dialplan/default/ - so this is the approach I've
taken.
After some suggestions in IRC to remove errant
when i said inline ... i just meant to define some variables in your DP ...
this is not a solution for you ... it is rather a proof of concept instead.
you need to do a DB lookup (sqlite or mysql).
T.
On Wed, Sep 23, 2009 at 1:32 AM, Francis Vidal francisv.l...@gmail.comwrote:
Yes, this is
Hi All,
Can anybody please tell me what are the gateways in Freeswitch ?
Thanks,
--
Anil Kumar S. R.
http://sranil.googlepages.com/
The best way to succeed in this world is to act on the advice you give to
others.
___
FreeSWITCH-users mailing list
On Wed, Sep 23, 2009 at 12:30 AM, Anil Kumar S. R. sra...@gmail.com wrote:
Hi All,
Can anybody please tell me what are the gateways in Freeswitch ?
A gateway is simply a means of doing an outbound registration to another
server. Once the gateway is created you may send calls out on it. A
endpoints that you are sending/receiving calls to/from It is useful to
have a separate configuration (other than dialplan) when you need to specify
credentials for GW to register somewhere, to specify domain, etc, etc ...
T.
On Wed, Sep 23, 2009 at 9:30 AM, Anil Kumar S. R.
You have access to the full sdp in channel vars, so you can condition
on those with regex.
Mike
On Sep 17, 2009, at 6:25 PM, Tihomir Culjaga wrote:
Hi Michael, thanks for your response.
i think it will be enough to check the call capability... we always
know the call is fax. We just
This should now be resolved in svn trunk.
Mike
On Sep 16, 2009, at 11:39 AM, Christian Löschenkohl wrote:
as a good fs user - of course i am :-) - i made a jira on this
MODAPP-336 to be precise
i hope this helps to solve my problem
br
On 2009-09-16 17:05, Rupa Schomaker wrote:
Either:
What issues are there with libtool 2 under debian? Libtool 2 issues
that I am aware of were all sorted out quite some time ago.
Mike
On Sep 17, 2009, at 10:07 PM, Jason White wrote:
While trying to build FreeSWITCH rev. 14913, compilation failed with
the
following.
the operating
Try taking a list at the info here: http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Solaris
You need to be passing any necessary cflags in on configure
Mike
On Sep 18, 2009, at 2:26 PM, email lists wrote:
Forwarding the issue below to see if anyone is familiar with this
Michael Jerris m...@jerris.com wrote:
What issues are there with libtool 2 under debian? Libtool 2 issues
that I am aware of were all sorted out quite some time ago.
With libtool2, mod_portaudio fails to link to the Alsa sound library, hence
fails to load due to unresolved symbols.
I can't
if someone can contact we later in the day offlist with credentials
for a box I can try to fix these issues.
Mike
On Sep 23, 2009, at 4:36 AM, Jason White wrote:
Michael Jerris m...@jerris.com wrote:
What issues are there with libtool 2 under debian? Libtool 2 issues
that I am aware of
A couple people have taken on major work on packages for ubuntu. Most
of that work will translate directly back to debian, we should just
need people to do testing of debian pacakges once their work is done.
Also we had one more person step up to help with spec file work. I
still need
Are you using freeswitch to detect the inband dtmf or are you getting
both inband and some other method (rfc 2833?) of dtmf as well?
Mike
On Sep 13, 2009, at 10:27 AM, Morten Henckel wrote:
Hi
I need to measure DTM digits duration and interdigit delay for
various phones in a two stage
Please catch up on irc to discuss this real time, this shouldn't be
happening and bkw or I likely will need remote access to your box to
figure out why it is doing this.
Mike
On Sep 21, 2009, at 1:06 PM, Luis Manuel Zuccolo wrote:
I' ve get the same error with a fresh tree
Thanks in
Michael Jerris m...@jerris.com wrote:
A couple people have taken on major work on packages for ubuntu.
Most of that work will translate directly back to debian, we should
just need people to do testing of debian pacakges once their work is
done.
I'm volunteering.
Hello Anthony,
I did further testing on a second machine and found out the following:
After
action application=set data=sip_ignore_remote_cause=true/
action application=hangup data=NO_ANSWER/
The called party receives a NO_ANSWER
and the calling party receives a NORMAL_CLEARING
So if you do this, how do you call between contexts? Say you have 100
tenants on one box each with their own domain and they are all 4 digit for
local dialing. If they call a 10 digit number like they are calling
outbound and it is another tenant on the same box, they don't want to go out
and
I upgraded to version 1.0.trunk.
And still with the problem.
I am using the soft phone X-Lite.
I set it (1000) and I connect. Once connected to create the User 1001. Ai in
another X-Lite I connect with the (1001) and create the User 1000.
Ai closing the two X-Lite and open the 1000 (it appears
Hi,
I know that FreeSWITCH uses libdingaling to talk to Jingle call parties.
Also I know that Jingle Protocol uses ICE protocol to traverse NAT devices.
So :
does libdingaling use an open library such as libnice for ICE?
Is it possible to use the ICE implementation in Sofia-SIP endpoint?
If not,
No mod_dingaling does not use LibNICE. However, i have plans to integrate
NICE with Sofia in mod_msn project, which is at the moment moving with very
slow pace due to some trouble in reverse engineering MSNP-18 protocol (used
in Windows Live Messenger 2009).
Thank you.
On Wed, Sep 23, 2009 at
Don't you think is better to integrate LibNICE to FreeSWITCH's RTP stack
than a module such as mod_msn / mod_dingaling ?
-- afshin
On Wed, Sep 23, 2009 at 4:21 PM, Muhammad Shahzad
shaherya...@googlemail.com wrote:
No mod_dingaling does not use LibNICE. However, i have plans to integrate
Yup, that's a good idea but not in my project list right now.
Thank you.
On Wed, Sep 23, 2009 at 7:17 PM, afshin afzali a.afzali2...@gmail.comwrote:
Don't you think is better to integrate LibNICE to FreeSWITCH's RTP stack
than a module such as mod_msn / mod_dingaling ?
-- afshin
On Wed,
Hello,
I finally solved it by using
action application=hangup data=${originate_disposition}/
Best regards
Peter
Peter P GMX schrieb:
Hello Anthony,
I did further testing on a second machine and found out the following:
After
action application=set
Use something like mod_easyroute to consult a database of DIDs. If
you host the DID it'll give you a route to dial out. If not, route
out your gateway.
Or load your lcr tables with your own DIDs. Consult mod_lcr and use
it's dialstring. It'll prefer longer prefix matches, so you will
always
That's a good idea. I thought about using a DB, but I was going to have to
use a lua script to look stuff up. I didn't think about easyroute or lcr.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
I'm not comfortable adding libnice into FreeSWITCH as it depends on
glib and that would add bloat in my opinion... is there no other
license compatible option?
/b
On Sep 23, 2009, at 8:42 AM, Muhammad Shahzad wrote:
Yup, that's a good idea but not in my project list right now.
Thank you.
Brian,
Thank yo very much for your reply.
I have tried to add transfer_ringback action, but it did not solve my
problem.
Destination phone is ringing, but the person who is calling does not hear
ringing tone in hte handset.
Is there anything in the logfile that can help you to identify the
We already have ice support in freeswitch, granted it is the slightly
twisted ice from the old jingle, but this should not be difficult to
fix. Knowing what I know about libnice architechture I can say almost
without doubt that it will never fit well into freeeswitch. Is the
basis of
I am exploring the possibility of building a Dialer that emulates the
logic of Call Files in asterisk.
A CallerID catcher is creating CUSTOM events that I can store in a
database. I can trigger callbacks using ESL but I wonder what is the best
way/nicer/geekier to do something like outgoing calls
Personally i am not a fan of GLib as well and always prefer STL over it due
to so many good reasons. But on the other hand libnice is the only library
that has Microsoft extensions to ICE protocol, which are required for
mod_msn to work.
So far on mod_msn, i am able to send and receive voice call
On Wed, Sep 23, 2009 at 10:19 AM, digilord g...@digilord.net wrote:
Hello all,
I know this is done and I think I figured out how to do it but I don't
want to reinvent the wheel so here goes. I am looking for a program
that will sit on the PBX. This program will intercept DHCP reply
What you want is NOT possible the way you describe it. Snom does a
multicast PNP which lets you reply with a notify. Polycom does a
DHCPINFORM which lets you respond with a DHCPACK with additional
options. Aastra does MDNS which dictates where to go get the configs.
/b
On Sep 23, 2009,
I don't think it's trivially possible, unless you can stick the PBX
between the DHCP server and the rest of the network. The reason is that
DHCP reply packets are not broadcast, but sent back to the MAC address
of the originator, so your Ethernet switch won't even let your PBX see
the replies.
make an esl script that monitors a dir for new files, and push the contents
into your same db?
On Wed, Sep 23, 2009 at 10:32 AM, Alberto Escudero aep.li...@it46.sewrote:
I am exploring the possibility of building a Dialer that emulates the
logic of Call Files in asterisk.
A CallerID catcher
libdingaling does not do the gtalk ice stuff, mod_dingaling does by using
utils in the FS core.
We have stun client code, random string generators in switch_utils.c and
settings in the rtp stack to send and recv authed stun packets using the
aforementioned functions mixed in with the audio.
ESL is probably the way to go tho...if you want to build a dialer.
The Dial Plans can get pretty advanced in FreeSWITCH...and if that is not
enough you might consider using mod_perl or something of that sort.
--matt
Voice Broadcasting http://www.hellohunter.com/voice_broadcast.php
Predictive
Just to give everyone an update. There are working Ubuntu packages in a
launchpad ppa. Debian users can add the ppa to their apt sources and
build the package on your box. I'm currently using the packages on my
home box and it is working great.
Alright. I'm looking for people who want to use the
I thought I had it figured out and was just missing one piece. Brian
provided me with the missing info. My assumptions were wrong. Now I
need to find software that will do what he described.
On Wed, 2009-09-23 at 18:51 +0300, David Knell wrote:
I don't think it's trivially possible, unless
Brian,
Is there code someplace that I can get that will help with the Polycom
DHCPINFORM way?
Thanks
On Wed, 2009-09-23 at 10:46 -0500, Brian West wrote:
What you want is NOT possible the way you describe it. Snom does a
multicast PNP which lets you reply with a notify. Polycom
Hi William,
I will be very happy to test them, can you share the source and procedure
to create the .debs?
It will be also very good to find ways to have a cepstral package included
*pending the licence* of course :)
/aep
--
Stopping junk mailers is good for the environment
Just to give
I would be more than happy to share the code I use.
Here is the git repo:
http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/
If anyone wants git commit access just send me your ssh public key and
I'll add it to the repo. As you can see Frank and I have been busy for
the last week.
Polycom responds to SIP server dhcp option (I think it's 120). I haven't
still seen any other phone doing that.
Ognjen
On Wed, Sep 23, 2009 at 6:16 PM, digilord g...@digilord.net wrote:
Brian,
Is there code someplace that I can get that will help with the
Polycom
DHCPINFORM way?
On Tue, Sep 22, 2009 at 11:05 PM, Jason White ja...@jasonjgw.net wrote:
lakshmanan ganapathy lakindi...@gmail.com wrote:
Thanks for your replay. I don't know what is latest trunk. Is it latest
version? I'm using freeswitch 1.0.4.
It's the latest version from the svn repository. Use svn
There are several ways to accomplish this, with enough effort.
The most straight forward approach is to run two DHCP servers on that
network, with each of the DHCP servers ignoring the OUIs of devices they do
not wish to manage.
You'll need to configure two separate pools on the two
servers (and
Yes, sounds the best way to go.
I assume that Unique-ID is the unique key to track the call via ESL
Unique-ID: a984afd4-a865-11de-a5b4-fb5a867b002c
and Answer-State: the variable to determine if the call is successful?
Or should wait for the reason of CS_DESTROY message. I want to avoid to
keep
I have perl code for both snom-pnp and polycom dhcp inform but my wish
list looks attractive ;)
/b
On Sep 23, 2009, at 12:09 PM, Dan White wrote:
There are several ways to accomplish this, with enough effort.
The most straight forward approach is to run two DHCP servers on that
network,
Hello,
I have a number of Polycom phones that appear to forget that they have
a message waiting then a little while later they seem to remember again.
This was not the case with the same phones under Asterisk (Sorry to say that
here but it's the only comparison I have).
I have looked at a
Should I delete the directory default and default.xml when I copy default
to foo.org and bar.org etc?
Diego
On Wed, Sep 23, 2009 at 2:11 PM, Peder pe...@networkoblivion.com wrote:
That's a good idea. I thought about using a DB, but I was going to have to
use a lua script to look stuff up. I
I have never seen this behavior. What do you have your MWI callback
method setup as?
/b
On Sep 23, 2009, at 12:44 PM, Daniel Morrigan wrote:
Hello,
I have a number of Polycom phones that appear to forget that
they have a message waiting then a little while later they seem to
Thats up to you :P
/b
On Sep 23, 2009, at 1:10 PM, Diego Viola wrote:
Should I delete the directory default and default.xml when I copy
default to foo.org and bar.org etc?
Diego
___
FreeSWITCH-users mailing list
Hello
I don't have the technical expertise to tell, so here goes: Unless mistaken,
Freeswitch is written in C and/or C++, so I guess it's not linked to the x86
instruction set.
Is so, would it be a lot of work of porting it so it runs on ARM processors?
It'd be cool because it could run on tiny
You can use Answer-State, CS_DESTROY won't happen until the call is
over.
On Sep 23, 2009, at 1:26 PM, Alberto Escudero wrote:
Yes, sounds the best way to go.
I assume that Unique-ID is the unique key to track the call via ESL
Unique-ID: a984afd4-a865-11de-a5b4-fb5a867b002c
and
I didn't get much help for my problem with XML CURL. What I meant to say is,
suppose I want to have some 1 users on freeswitch. Do we have to create
some many xml files in the directory or is there some way in which the users
can be put in the db ?
Also. my another question is, what is the
http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Cross_Compiling_for_ARM_on_Linux
Mike
On Sep 23, 2009, at 2:25 PM, Fred-145 wrote:
Hello
I don't have the technical expertise to tell, so here goes: Unless
mistaken,
Freeswitch is written in C and/or C++, so I guess it's
Hey Brian,
On 23-Sep-2009, at 11:16 PM, Brian West br...@freeswitch.org wrote:
I have perl code for both snom-pnp and polycom dhcp inform but my wish
list looks attractive ;)
Aahan so is it (wish list) documented anywhere? Lol
Mitul Limbani
Enterux Solutions Pvt. Ltd.
www. Enterux.com
There are tons of details on this at
http://wiki.freeswitch.org/wiki/Mod_xml_curl
Are you having an issue?
Mike
On Sep 23, 2009, at 2:37 PM, Anil Kumar S. R. wrote:
I didn't get much help for my problem with XML CURL. What I meant to
say is, suppose I want to have some 1 users on
Ok I have configured the two domains with their own directory and I can
register fine with them now.
But I need to configure two different dialplans with their own profiles.
How do I tell a specific domain to use a specific profile/dialplan?
Thanks,
Diego
On Wed, Sep 23, 2009 at 6:52 PM,
s/directory/directories/
Should I use context for that?
On Wed, Sep 23, 2009 at 7:40 PM, Diego Viola diego.vi...@gmail.com wrote:
Ok I have configured the two domains with their own directory and I can
register fine with them now.
But I need to configure two different dialplans with their
I prefer to specify the context as a per-domain so it affects all the users
on the domain directly...
On Wed, Sep 23, 2009 at 7:48 PM, Diego Viola diego.vi...@gmail.com wrote:
Do I specific the context as a per-user thing, can I specific the context
as a per-domain way?
Diego
On Wed, Sep
On Wed, Sep 23, 2009 at 8:08 PM, William King quentus...@gmail.com wrote:
Just to give everyone an update. There are working Ubuntu packages in a
launchpad ppa. Debian users can add the ppa to their apt sources and
build the package on your box. I'm currently using the packages on my
home box
Do I specific the context as a per-user thing, can I specific the context as
a per-domain way?
Diego
On Wed, Sep 23, 2009 at 7:42 PM, Diego Viola diego.vi...@gmail.com wrote:
s/directory/directories/
Should I use context for that?
On Wed, Sep 23, 2009 at 7:40 PM, Diego Viola
You really don't have too but you can set a user_context on each user
in the domain or on the domain level to specify the domain's or user's
context.
/b
On Sep 23, 2009, at 2:40 PM, Diego Viola wrote:
Ok I have configured the two domains with their own directory and I
can register fine
Can you next time pause a few moments... Think about what you're
sending and send ONE email with your questions? This 10 emails from
you replying to yourself things looks like you're a crazy man! :P
/b
PS: ask on IRC or mailing list NOT BOTH please.
On Sep 23, 2009, at 2:48 PM, Diego
On Wed, Sep 23, 2009 at 8:08 PM, William King quentus...@gmail.com wrote:
(..)
Any questions?
Another problem (on Ubuntu jaunty):
Setting up freeswitch (1.0.4+repack6-0ubuntu14925.1) ...
adduser: The --group, --ingroup, and --gid options are mutually exclusive.
addgroup: The user `freeswitch'
Ok, sorry for that and thanks for the help :).
Diego
On Wed, Sep 23, 2009 at 8:09 PM, Brian West br...@freeswitch.org wrote:
Can you next time pause a few moments... Think about what you're sending
and send ONE email with your questions? This 10 emails from you replying to
yourself things
On Wed, Sep 23, Dmitry Bely wrote:
Can you enable mod_skypiax in your debian package?
We will be enabling as much as we can cleanly build on debian/ubuntu. There
will be a lot more to come. We will be breaking the mods and end points in to
different packages so that you can install what you
On Thu, Sep 24, Dmitry Bely wrote:
On Wed, Sep 23, 2009 at 8:08 PM, William King quentus...@gmail.com wrote:
(..)
Any questions?
Another problem (on Ubuntu jaunty):
Setting up freeswitch (1.0.4+repack6-0ubuntu14925.1) ...
adduser: The --group, --ingroup, and --gid options are mutually
For those that are not aware, I've made some changes to the Windows
installer over the past month. You can find a summary on this link:
http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Precompiled_Binaries
The latest change is the bundling of 32 and 64 bits builds in one
file. The
On Thu, 24 Sep 2009 04:08:27 William King wrote:
Any questions?
What file structure did you create the package with i.e. /opt/, /usr/local/,
/usr/ - just wondering about inclusion into the archive.
Also you mentioned multiverse - what parts have licensing that requires going
into multiverse
Hi Rich
On Thu, Sep 24, Hadley Rich wrote:
What file structure did you create the package with i.e. /opt/, /usr/local/,
/usr/ - just wondering about inclusion into the archive.
Currently it's /opt/freeswitch. I would like to see it move to FHS correct
locations for inclusion in to
On Thu, 24 Sep 2009 09:18:23 Frank Carmickle wrote:
Currently it's /opt/freeswitch. I would like to see it move to FHS correct
locations for inclusion in to debian/ubuntu. This is the next bit that I
will be working on.
Yeah, the FHS stuff was the bit that I got a little stuck on a while
I would be willing to package it. It would go faster with some help, or
a patch. One of my goals is to have all of the possible mods for
freeswitch as built packages.
-William King
Dmitry Bely wrote:
On Wed, Sep 23, 2009 at 8:08 PM, William King quentus...@gmail.com wrote:
Just to give
I would like to get the package into which ever would work best. I
haven't taken enough time to find out which repo would be the most
appropriate.
-William King
Hadley Rich wrote:
On Thu, 24 Sep 2009 04:08:27 William King wrote:
Any questions?
What file structure did you create the
It seems I had a port forwarded incorrectly for the external access to
the git web interface. here it is again:
http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/
I've tested it to work now.
-William King
Hadley Rich wrote:
On Thu, 24 Sep 2009 09:18:23 Frank Carmickle wrote:
On 24/09/09 3:49 AM, Anthony Minessale wrote:
make an esl script that monitors a dir for new files, and push the
contents into your same db?
An easy way to do this is to use incron.
--
Cheers,
Matt Riddell
Director
___
On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle fr...@carmickle.com wrote:
On Wed, Sep 23, Dmitry Bely wrote:
Can you enable mod_skypiax in your debian package?
We will be enabling as much as we can cleanly build on debian/ubuntu. There
will be a lot more to come. We will be breaking the
Thanks Jason, I'll look into your suggestions. Does this imply that creating
a conference which requires a pin via my current approach will not work?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Sure, post it here and I'll add it in the next build in a few hours.
-William King
Dmitry Bely wrote:
On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle fr...@carmickle.com wrote:
On Wed, Sep 23, Dmitry Bely wrote:
Can you enable mod_skypiax in your debian package?
We will
I've been having the same idea, except completely different. I'd probably
start with StructureMap or MEF. I'm attracted to the idea of creating an
alternative to mod_event_socket except using WCF as the transport, enabling
both WS-* and Rest access into the FreeSWITCH core.
I think an effort to
Brian,
It was set for contact. Would that cause this behavior?
Daniel
On Wed, Sep 23, 2009 at 11:15 AM, Brian West br...@freeswitch.org wrote:
I have never seen this behavior. What do you have your MWI callback
method setup as?
/b
On Sep 23, 2009, at 12:44 PM, Daniel Morrigan
Brian,
Just for kicks I changed it to registration. Same thing is happening.
Daniel
On Wed, Sep 23, 2009 at 4:38 PM, Daniel Morrigan g...@digilord.net wrote:
Brian,
It was set for contact. Would that cause this behavior?
Daniel
On Wed, Sep 23, 2009 at 11:15 AM, Brian West
NO I have never seen it happen what firmware version are you running?
/b
On Sep 23, 2009, at 6:38 PM, Daniel Morrigan wrote:
Brian,
It was set for contact. Would that cause this behavior?
Daniel
___
FreeSWITCH-users mailing list
Hey nice to see someone interested in this - that's a lot of files you have
there, and looks like you put a lot of effort into it! I haven't had time to
look at it much, but here are a few initial impressions.
Right off the bat: there can be tons of cleanup and refactoring, no doubt about
Hello,
I am trying to run FreeSwitch on a machine which has more than one
interface, all of them should be used for SIP. The FreeSwitch binds only to
the first one. I tried setting bind_server_ip to either auto or 0.0.0.0
but it doesn't help.
Any idea what to do?
sorry when I said on profile I want to say one profile
2009/9/24 Seven Du dujinf...@gmail.com
It not possible to use 0.0.0.0 for on profile. however, you can create more
sip profiles for each of your interfaces. Search freeswitch-users archievs
then you will find similar topics.
2009/9/24
It not possible to use 0.0.0.0 for on profile. however, you can create more
sip profiles for each of your interfaces. Search freeswitch-users archievs
then you will find similar topics.
2009/9/24 Yehavi Bourvine yehavi.bourv...@gmail.com
Hello,
I am trying to run FreeSwitch on a machine
Thanks!
__Yehavi:
2009/9/24 Seven Du dujinf...@gmail.com
It not possible to use 0.0.0.0 for on profile. however, you can create more
sip profiles for each of your interfaces. Search freeswitch-users archievs
then you will find similar topics.
2009/9/24 Yehavi Bourvine
On Wed, Sep 23, 2009 at 10:44 PM, Seven Du dujinf...@gmail.com wrote:
It not possible to use 0.0.0.0 for on profile. however, you can create more
sip profiles for each of your interfaces. Search freeswitch-users archievs
then you will find similar topics.
It sure would be nice to be able to
On Wed, Sep 23, 2009 at 12:37 PM, Anil Kumar S. R. sra...@gmail.com wrote:
I didn't get much help for my problem with XML CURL. What I meant to say is,
suppose I want to have some 1 users on freeswitch. Do we have to create
some many xml files in the directory or is there some way in which
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