Hi,
I am facing a problem with Sofia SIP external profile. Basically i have 10+
accounts, say a101 - a110, to a single service provider, say xyz.com. For
each account a have created a gateway in external profile's directory i.e.
/usr/local/freeswitch/conf/sip_profiles/external/a101.xml up to
Hi Michael,
The feature is already documented here:
http://wiki.freeswitch.org/wiki/Dialplan_XML#Clarification
http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
Perhaps the reason *why* it's the way it is can be expanded a bit ?
regards,
Leon
On Oct 21, 2009, at 7:02 PM, Michael
I'm using 'FreeSWITCH Version 1.0.trunk (15106)'.
Anthony Minessale-2 wrote:
which revision of FS are you using?
On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H. nagale...@gmail.com wrote:
I've tried with origination_uuid.
First, I tried with SIP and my program executes successfully as
cond would be helpful here? I updated the wiki on this one just now
with a bit more detail. It is a api call. so, you'd use it like:
${cond(eval ? trueval : falseval)}
so to get a value of ERR if the var my myvar is 15 you could:
${cond(${myvar} 15 ? ERR : OK)}
If both sides of the
TC
TCI have enabled crash-protection and when i do SIP = H323 call it
doesn't
TCgenerate coredumps... it is just this thread that is crashing ... pls
check
TCthe log downbelow:
core dump in case enabled crash-protection may be unusable, i have a case
then
my module crash silently,
Hi
I have set up a freeswitch with TLS and SRTP support. I'm sending encrypted
calls to Freswitch and the Freeswitch forwards the calls to an asterisk
unencrypted. I have issue by using G729 in this scenario. My UA supports
g279, asterisk supports g729 transcoding and I understood that
TCHi, here is the FS log without crash-protection:
TChttp://pastebin.freeswitch.org/10796 and here is the backtrace:
TChttp://pastebin.freeswitch.org/10797
i fix this crash already, please download latest version from same link
as previous, recompile and try again.
outgoing works, I can
On 2009-10-22 15:59 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC
TC
TC TCHi, here is the FS log without crash-protection:
TC TChttp://pastebin.freeswitch.org/10796 and here is the backtrace:
TC TChttp://pastebin.freeswitch.org/10797
TC
TC i fix this crash already, please
On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TCOn Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga tculj...@gmail.com wrote:
TC
TC
TC TCHi, here is the FS log without crash-protection:
TC TChttp://pastebin.freeswitch.org/10796 and here is the backtrace:
TC
crash protection has been completely removed from FreeSWITCH, I certianly
hope you are working on this against SVN trunk? Also you have been given an
svn area and a jira category for this so you should move all the info from
this thread to jira http://jira.freeswitch.org
It's much easier to
Have you started moving the code into our SVN and using our
ticketing / issue tracker to help you manage issues?
/b
On Oct 22, 2009, at 9:13 AM, Georgiewskiy Yuriy wrote:
On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre
...:
TCOn Thu, Oct 22, 2009 at 3:59 PM,
On 2009-10-22 09:30 -0500, Brian West wrote freeswitch-us...@lists.freeswit...:
hm, you not tell me what account created, and i don't try to do this.
BWHave you started moving the code into our SVN and using our
BWticketing / issue tracker to help you manage issues?
BW
BW/b
BW
BWOn Oct 22,
AS per the email you and I exchanged we created the account and the
mod_h323 folder in endpoints
/b
On Oct 22, 2009, at 9:34 AM, Georgiewskiy Yuriy wrote:
hm, you not tell me what account created, and i don't try to do this.
___
On 2009-10-22 09:27 -0500, Anthony Minessale wrote freeswitch-us...@lists.f...:
AMcrash protection has been completely removed from FreeSWITCH, I certianly
AMhope you are working on this against SVN trunk?
i don't have trunk at this time, my current work is based on 1.0.4 version.
AMAlso you
Thanks Mike for the link. I'll investigate more whether running FS on
ARM-based devices is a good idea.
For those interested, another thread on the subject:
http://www.nabble.com/Freeswitch-vs.-Asterisk-speed-on-ARM-td25086585.html
--
View this message in context:
TC
TCDo you need some logs ?
try disable medai-proxy, there is issue with rtp now then medai-proxy or
transcoding enabled.
Outbound calls:
disabled rtp proxy and it is still the same issue ... audio delay H323 =
SIP endpoint.
Inbound calls:
This is the extension i use to register
2009/10/22 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-22 09:27 -0500, Anthony Minessale wrote
freeswitch-us...@lists.f...:
AMcrash protection has been completely removed from FreeSWITCH, I
certianly
AMhope you are working on this against SVN trunk?
i don't have trunk at this time,
Hello Anyone,
Now, I'm attempting to load CDR Data on the database in Real Time.
by following the instruction on this link
http://wiki.freeswitch.org/wiki/CDR
However, when trying to send create_table.rb to the database,
I'm still struggling with connecting to mySQL database
Note:
my
I have a noobish question about setting up FS.
I have it installed and running.
I setup a soft client on the machine fs is on and point it to the ip address
of the FS instance and it registers with no issues.
I then setup an entry in my etc/hosts files mydomain.localhost and changed
the domain
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Mike,
here it is:
Dialplan:
extension name=Local_Extension
condition field=destination_number expression=^(10[01][0-9])$
action application=set data=dialed_extension=$1/
action application=export
An update for Tony, Brian, Mike, and everyone on the list...
I was able to get some phone time with the team yesterday. Tony
worked on my machine, found the issue, and had it committed within 30
minutes.
I've been testing T.38 all morning between the fax machines in the
office with few issues.
On Thu, Oct 22, 2009 at 5:44 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
An update for Tony, Brian, Mike, and everyone on the list...
I was able to get some phone time with the team yesterday. Tony
worked on my machine, found the issue, and had it committed within 30
Im going to guess its because mydomain.localhost doesn't resolve
outside the machine itself so the softphone never ends up knowing wtf
to do.
/b
On Oct 22, 2009, at 10:42 AM, freeswitch noob wrote:
I have a noobish question about setting up FS.
I have it installed and running.
I setup
I can't get what exactly you re talking about. Can you clarify ? Also
please include the packets of interest only not the full trace if its
not relevant to the bug.
/b
On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Mike,
here it is:
Hi Rupa,
Thanks again for your advice.
I have been searching for the method to record in the freeswitch documentation
but I'm still not sure which command and method to make the record for every
call automatically.
Which command or method do you use?
And to make the recording start and stop
I'm sure that problem is gone in svn trunk.
On Thu, Oct 22, 2009 at 11:25 AM, Brian West br...@freeswitch.org wrote:
I can't get what exactly you re talking about. Can you clarify ? Also
please include the packets of interest only not the full trace if its
not relevant to the bug.
/b
On
please update to svn trunk with make current and try again.
On Thu, Oct 22, 2009 at 4:08 AM, Nagalenoj nagale...@gmail.com wrote:
I'm using 'FreeSWITCH Version 1.0.trunk (15106)'.
Anthony Minessale-2 wrote:
which revision of FS are you using?
On Wed, Oct 21, 2009 at 1:06 AM,
What is heartbeat and what are the uses cases?
Sorry i didn't find much information on wiki.
Thanks.
On Sat, Oct 10, 2009 at 12:01 AM, Diego Viola diego.vi...@gmail.com wrote:
Here's my heartbeat script now.
#!/usr/bin/env ruby
require 'rubygems'
require 'fsr'
require
On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker r...@rupa.com wrote:
cond would be helpful here? I updated the wiki on this one just now
with a bit more detail. It is a api call. so, you'd use it like:
${cond(eval ? trueval : falseval)}
so to get a value of ERR if the var my myvar is 15
On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga tculj...@gmail.com wrote:
and what these few issues are? :P
One fax machine here in the office (still testing others) correctly
sends all fax pages. A minute or so after the fax is marked
successful on both sides it hangs up, redials, and
Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had
nothing but intermittent problems with Super G3 FAXes over T.38, unless
v.34 is strictly turned off on the machine.
Gabe
Kristian Kielhofner wrote:
On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga tculj...@gmail.com wrote:
On Thu, Oct 22, 2009 at 10:47 AM, Saeed Ahmad saeedahmad1...@gmail.comwrote:
What is heartbeat and what are the uses cases?
Sorry i didn't find much information on wiki.
Thanks.
A session heartbeat is just an event that is sent to your script and gives
you updated information about the
It just hangsand I CTRL-C out of it.
[r...@ss]# ./fs_cli -H 127.0.0.1
^C
[ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error]
Freeswitch process is running:
[r...@ss bin]# ps -ef|grep free
root 8889 31039 0 12:36 pts/200:00:00 ./freeswitch
root 8952
indeed, this looks like a dialect problem between your fax machine and
your T.38 device.
Anyhow, T.38 doesn't work well with SG3... I Always have to disable v.34 in
order to have a reliable fax service.
Also, cisco uses to suppress CM so that SG3 timeouts on ANSam the
communication fallbacks to
Hangs for how long? Are you sure you are not just waiting on a timeout?
JM
On Thu, Oct 22, 2009 at 4:42 PM, Ujjval Karihaloo
ujj...@simplesignal.comwrote:
It just hangs….and I CTRL-C out of it.
[r...@ss]# ./fs_cli -H 127.0.0.1
^C
[ERROR] libs/esl/fs_cli.c:652 main() Error
Gabe,
I don't think any of them are plus the T.38 SDP tells me the bitrate
is 14400, certainly not V.34 speed.
Are you saying the machine even trying to negotiate V.34 poses a problem?
On Thu, Oct 22, 2009 at 2:16 PM, Gabriel Kuri gk...@ieee.org wrote:
Out of curiosity, is it a Super G3
On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
finally i fix this rtp bug, check new wersion please.
TC
TC TC
TC TCDo you need some logs ?
TC
TC try disable medai-proxy, there is issue with rtp now then medai-proxy or
TC transcoding enabled.
TC
TC
TCOutbound
I use the dialplan app session_record:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
I call that in the appropriate parts before bridging to call. For
incoming it is just before the bridge to user/username. For outgoing
it is before I bridge to the outgoing provider.
On
AFAIK T.38 v2 supports a max speed of 14.4 anyhow, so that's the max
speed you'll ever see in the SDP. T.38 v3 supposedly supports v.34
speeds, however no one that I've seen has implemented it yet - not sure
it's even an official standard?
Yes, in my experience, v.34 capable FAXes do not properly
Hello gents,
I know that it is possible to make a 3-way conferencing using att_xfer and
by pressing 0.
But I'm more interested in the way of doing 3-way conferencing using mod
conference and bind_app. Saying in other words, I want to do the same thing
what 3-way conferencing using att_xfer but
2009/10/22 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
finally i fix this rtp bug, check new wersion please.
if course i can do that, but tomorrow morning ... i'm not in the office
anymore.
BTW: can we please move
freeswi...@ss_freeswitch sofia_gateway_data
Segmentation fault (core dumped)
Just ran the gateway command above w/o any parameters,,, and it core dumped..
I am sure mistakes like that happen...but I not sure if it should core dump
and shutdown.
What SVN rev of FS? What operating system? If you're not on the latest then
do a make current and get to the latest SVN and see if you can replicate
the issue.
-MC
On Thu, Oct 22, 2009 at 12:45 PM, Ujjval Karihaloo
ujj...@simplesignal.comwrote:
freeswi...@ss_freeswitch sofia_gateway_data
If I run .freeswitch , get back to the ROOT prompt and then from same window
type in fs_cli...it fails...hangs forever
However, if I open another new ssh session and fs_cli from the new session, it
works.
From: freeswitch-users-boun...@lists.freeswitch.org
Hi All,
I have FS registered to an ITSP. The contact is showing as follows..
Contact: sip:gw+i...@1.1.1.1:5080;transport=udp
Itsp is the name of the SIP gatway and its IP is changed to 1.1.1.1
I want the Phone number (FromUser)to show in the contact header in the REGISTER
msg going to the
On 2009-10-22 21:44 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC2009/10/22 Georgiewskiy Yuriy bottle...@icf.org.ru
TC
TC On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote
TC freeswitch-us...@lists.fre...:
TC
TC finally i fix this rtp bug, check new wersion please.
TC
TC
TCif
If it's not windows probably be safer to just do freeswitch -nc for
no console. Give it a little to start up then fs_cli should be fine.
netstat -anp can also be used to see that the ports have binded
correctly.
-- W
___
FreeSWITCH-users mailing list
Ok I got this one...just put ext-in-contact setting and then define the
extension to be same as FromUser in my provider.xml in
/conf/sip-profile/external/
!--/// extension for inbound calls: *optional* same as username, if blank
///--
param name=extension value=xx/
!--extra sip
I did test this on trunk and it seems to work right:
freeswi...@default sofia_gateway_data
-ERR Parameter missing
Mike
On Oct 22, 2009, at 3:58 PM, Michael Collins wrote:
What SVN rev of FS? What operating system? If you're not on the
latest then do a make current and get to the latest SVN
We have probably 30-40 fax machines running T.38 between Linksys SPA-3102
and Cisco gateways. We have a pretty good success rate and we have been
doing this for probably 2-3 years. We have a couple of them going thru an *
1.4 box and it seems to work ok. One of my projects is to try and get
I do have the core dump, should I open a ticket.
I am running latest Freeswitch 1.0.4 and had done a make current just before it
happened.
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
Yes, if this is latest SVN (after a make current) then open a jira.
-MC
On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo
ujj...@simplesignal.comwrote:
I do have the core dump, should I open a ticket.
I am running latest Freeswitch 1.0.4 and had done a make current just
before it happened.
How do I tell if it's the latest...I downloaded is yesterday..and installed it
from freeswitch.org
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
Hi
I have the Basic Conferencing working. Here is my Dial Plan.
I want to be able to setup a Moderator PIN different from other participants,
when I add the moderator flag it logs me in directly w/o asking for a PIN..
action application=conference
Type version on the CLI.
On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo
ujj...@simplesignal.comwrote:
How do I tell if it’s the latest…I downloaded is yesterday..and installed
it from freeswitch.org
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
I think the (exported) means you don't have the latest svn, but probably the
officially released build 1.0.4 that can be downloaded from the FS page.
I think you should see something like (the latest trunk is 15203):
freeswi...@internal version
FreeSWITCH Version 1.0.trunk (15126)
I guess you
He's running 1.0.4, he needs to checkout from SVN trunk and build FS from
scratch.
Diego
On Fri, Oct 23, 2009 at 4:00 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
I think the (exported) means you don't have the latest svn, but probably
the officially released build 1.0.4 that can
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