Re: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000
In my case the 1001 resides in - /usr/local/freeswitch/conf/directory/default/1001.xml And you set the Caller Name and ID by adding - On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. q.edw...@gmail.com wrote: Edmar Eso esta en freeswitch/conf/vars.xml en ese archivo. If i am not mistaken and anyone welcome to correct me i just told Edmar this is set in freeswitch/conf/vars.xml ... file Ed On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz darklio...@yahoo.com wrote: When I calling an outbound extension it appears: name is FreeSWITCH and number is 0 How can i change it depends on the user who is calling? Sample 1001-64521223 I just want the name 1001 to appear not FreeSWITCH same as the number Thanks -- View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D0-tp24158823p24158823.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Compiling freeswitch for Dragonfly BSD
Hi.
I have been searching for an alternative PBX to asterisk (which has not
been all that stable) to run on Dragonfly BSD. I spent a fair amount of
time a couple months ago trying to compile freeswitch without success.
I have since tried Yate, but it consumes 85-95% of the cpu when idle
(not processing any calls).
I am considering revisiting freeswitch. I have included below my notes
of all the various problems I encountered before, which ones I resolved
and how, up to the point where I left off. I am in hopes of getting
feedback on whether any of the issues have been fixed or are planned to
be fixed and/or suggestions on getting it working. Also, perhaps these
notes on my experiences will be helpful for the developers to improve
freeswitch.
Keep in mind that Dragonfly is a branch from Freebsd and stays fairly
compatible. I am able to compile most software, that is ported to
FreeBSD, with few problems.
Here are my notes
=
To compile on dragonfly BSD 1.10.1-RELEASE
==
I had to add -D__FreeBSD__ to CPPFLAGS
ln sh to bash or zsh
because I got unexpected operator errors from test during configure
with the bsd shell.
ln make to gmake
Their scripts were calling make even though I ran the build
using gmake.
Must have apr-0.9.16.2.0.61 and apr-util-0.9.16.2.0.61 installed
apr-0.xxx has headers, which freeswitch is including, that apr-1.xxx
does not have.
I sym-linked the apr-util libs from apr-util-1.2.8nb1 because
apr-util-0.9.16.2.0.61 was not available as a binary package.
The freeswitch-1.0.3.tar.gz release did not have bootstrap.sh, which is
required for building from the svn repository.
Somebody on the #freeswitch IRC suggested I get it from the subversion
repository and run the bootstrap script. bootstrap.sh is not in the
release.
==
Tue Mar 31 00:12:37 CDT 2009
I posted on the freeswitch mailing list asking about the compilation errors
Got no responses.
This first set of apr_... warnings turned out to clearly be from not having the
correct apr-util package installed, I should have gotten a response on the list
about it, considering it is a clear dependency that they do not directly
specify on the web site or the source docs. apr-util is a dependency of
subversion. They list SVN as a dependency of freeswitch, which is the utility
in the subversion package, not the package name. svn should not be
a dependency to build from a release archive that is not retrieved from the svn
repository.
As it turns out, it had to be apr-util version 0.9.15. See notes below.
==
Tue Mar 31 22:22:58 CDT 2009
I tried the freeswitch-snapshot.tar.gz from the freeswitch site, which is
a 03/30/2009 snapshot from the svn trunk.
*concern*
It was nearly twice as big as the release for some reason.
27016871 Mar 28 13:08 freeswitch-1.0.3.tar.gz
52854882 Mar 31 18:25 freeswitch-snapshot.tar.gz
Running bootstrap.sh produced a bunch of these errors from automake:
Use of uninitialized value in exists at /usr/pkg/bin/automake line 4823, GEN0
line 1.
Use of uninitialized value in concatenation (.) or string at
/usr/pkg/bin/automake line 4823, GEN0 line 1.
automake:
automake: ## Internal Error ##
automake:
automake: unrequested trace `'
automake: Please contact bug-autom...@gnu.org.
at /usr/pkg/share/automake-1.10/Automake/Channels.pm line 570
Automake::Channels::msg('automake', '', 'unrequested trace `\'') called
at /usr/pkg/share/automake-1.10/Automake/ChannelDefs.pm line 191
Automake::ChannelDefs::prog_error('unrequested trace `\'') called at
/usr/pkg/bin/automake line 4823
Automake::scan_autoconf_traces('configure.ac') called at
/usr/pkg/bin/automake line 5046
Automake::scan_autoconf_files() called at /usr/pkg/bin/automake line 781
==
Thu Apr 2 20:51:44 CDT 2009
Going back to the 1.0.3 release.
I was getting a a bunch of apr_... warnings like
Compiling src/switch_apr.c ...
src/switch_apr.c: In function `switch_thread_self':
src/switch_apr.c:74: warning: implicit declaration of function
`apr_os_thread_current'
src/switch_apr.c:74: warning: return makes pointer from integer without a
cast
...
Then
gmake[3]: *** [libfreeswitch_la-switch_apr.lo] Error 1
I finally got past that by installing apr-0.9.16.2.0.61 and apr-util-0.9.15.
I compiled apr-util-0.9.15 myself because it was not available as a binary
package.
Source files are including headers from apr-util-0.9.15 that do not exist in
the
binary package, apr-util-1.2.8nb1.tgz.
I also got past the Cannot guess build type error during configure by adding
--build=i386 instead of having to use the uname wrapper to fake FreeBSD.
==
New error: (This one is not the fault of freeswitch)
Compiling
[Freeswitch-users] Newbee: Need Help Thin Client Environment
Hello All, I am an absolute newbee in the voip world but have a project where I believe freeswitch will work and need very, very basic guidance on how to setup a testbed in a thin client environment. I am using K12LTSP 5EL (Centos5) for my terminal server/host testbed and utilize fl_teachertool 0.07 to monitor the connected terminal clients (TCs). If you are not familiar with fl_teachertool, it allows a teacher to view thumbnail images of each TC logged in to the server. The teacher can click on any thumbnail and enlarge the view, monitor all applications running on a given TC, and take control of the keyboard and mouse of the TC. These are just a few of the capabilities of fl_teachertool. What I want to do is allow the teacher to establish voice communication using headsets and microphones with any one of the TCs by making a phone call via ethernet based upon ip of the TC through freeswitch using a softphone. Does this sound like something that is possible using freeswitch? If so, could someone please give me very basic instructions on how to setup this proof of concept? If I can just get a teacher stationed at my server talking to one student at a TC, I believe I can go from there. Currently I have a voiper softphone that functions, I believe, under gnome, but I have no idea how to configure the voiper to initiate calls through freeswitch or how to configure freeswitch to route the call to one of my TCs. I also need to keep this system fully self contained. That is, I can not have a requirement to use an outside sip service provider. Also, I would use any other linux sip softphones known to work with freeswitch that people feel would work better than a voiper. voiper seems to be more windows and mac based. I would really like to use an ekiga since they seem to be more linux based, but I do not believe that they have been thoroughly tested with freeswitch. Any help would be greatly appreciated! Regards, Murrah Boswell ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000
Sorry Edmar I missundertood you .. I thought you wanted to change the number showing once you were going out not the 1001.xml file. In this case Harmeet is right. There you have those values to to make the changes. My bad. Ed On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh harm...@litatel.com wrote: In my case the 1001 resides in - /usr/local/freeswitch/conf/directory/default/1001.xml And you set the Caller Name and ID by adding - On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. q.edw...@gmail.com wrote: Edmar Eso esta en freeswitch/conf/vars.xml en ese archivo. If i am not mistaken and anyone welcome to correct me i just told Edmar this is set in freeswitch/conf/vars.xml ... file Ed On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz darklio...@yahoo.comwrote: When I calling an outbound extension it appears: name is FreeSWITCH and number is 0 How can i change it depends on the user who is calling? Sample 1001-64521223 I just want the name 1001 to appear not FreeSWITCH same as the number Thanks -- View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D0-tp24158823p24158823.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000
Actually the extension_caller_id=Extension 1001 and extension_caller_number=1001 is set as Harmeet says but the same issue FreeSwitch the caller name and the number is 000 i just want 1001 the caller number and the id Edmar Edward Q. wrote: Sorry Edmar I missundertood you .. I thought you wanted to change the number showing once you were going out not the 1001.xml file. In this case Harmeet is right. There you have those values to to make the changes. My bad. Ed On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh harm...@litatel.com wrote: In my case the 1001 resides in - /usr/local/freeswitch/conf/directory/default/1001.xml And you set the Caller Name and ID by adding - On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. q.edw...@gmail.com wrote: Edmar Eso esta en freeswitch/conf/vars.xml en ese archivo. If i am not mistaken and anyone welcome to correct me i just told Edmar this is set in freeswitch/conf/vars.xml ... file Ed On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz darklio...@yahoo.comwrote: When I calling an outbound extension it appears: name is FreeSWITCH and number is 0 How can i change it depends on the user who is calling? Sample 1001-64521223 I just want the name 1001 to appear not FreeSWITCH same as the number Thanks -- View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D0-tp24158823p24158823.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D0-tp24158823p24160712.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000
depending how do you make out going call. On Jun 23, 2009, at 2:39 PM, Edmar Cruz wrote: Actually the extension_caller_id=Extension 1001 and extension_caller_number=1001 is set as Harmeet says but the same issue FreeSwitch the caller name and the number is 000 i just want 1001 the caller number and the id Edmar Edward Q. wrote: Sorry Edmar I missundertood you .. I thought you wanted to change the number showing once you were going out not the 1001.xml file. In this case Harmeet is right. There you have those values to to make the changes. My bad. Ed On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh harm...@litatel.com wrote: In my case the 1001 resides in - /usr/local/freeswitch/conf/directory/default/1001.xml And you set the Caller Name and ID by adding - On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. q.edw...@gmail.com wrote: Edmar Eso esta en freeswitch/conf/vars.xml en ese archivo. If i am not mistaken and anyone welcome to correct me i just told Edmar this is set in freeswitch/conf/vars.xml ... file Ed On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz darklio...@yahoo.comwrote: When I calling an outbound extension it appears: name is FreeSWITCH and number is 0 How can i change it depends on the user who is calling? Sample 1001-64521223 I just want the name 1001 to appear not FreeSWITCH same as the number Thanks -- View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D0-tp24158823p24158823.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D0-tp24158823p24160712.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] mod_dingaling picking wrong IP address / no audio?
Hi! I am trying to call from my corporate network (firewalled) using Gtalk to Freeswitch. I am not getting any audio. In the logs I see that mod_dingaling is using my internal corporate IP address which is not publically addressable. 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates 146.xx.xx.xx:50320 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable Candidate 146.xx.xx.xx:50320 Further on in the log, I can see GTalk sending a new candidate IP address to use: 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 name=rtp type=local protocol=udp username=e+JTkVHT1xEkqXGD password=fAxU6Pr1oF9Zq48U address=192.168.1.102 port=50322 pref=1.00 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 name=rtp type=stun protocol=udp username=RBqyF2XNMYLfJNoU password=DQMjon1fSVoJIRTp address=124.xxx.xxx.xxx port=50323 pref=0.90 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=relay protocol=udp username=62L5zs2FHbcUdeCJ password=KxmNgkUmZsLfuX6S address=209.xx.xxx.xxx port=19295 pref=0.50 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] Because of this, I never get audio. Any ideas how to fix this? Thanks! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Variable manipulation in the dialplan
Hello,
I once found in the wiki a page explaining how to substring a channel
variable,
something like
@[intra]lanman 12345 would be 345 if you do ${var:2}
I can't find that page on the wiki anymore, any hint on were it could
be? :-)
Also do you think it could be useful to extend this functionality with a
sort of Java indexOf() to extract a specific substring from a variable
(but without knowing its size like in the example above)?
Regards,
Claudio
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Re: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000
Check your dialplan where you call bridge to gateway to make outgoing
calls. Stick in the following lines before the bridge call -
action application=set
data=effective_caller_id_number=${effective_caller_id_number}/
action application=set
data=effective_caller_id_name=${effective_caller_id_name}/
On Tue, Jun 23, 2009 at 2:39 AM, Edmar Cruz darklio...@yahoo.com wrote:
Actually the extension_caller_id=Extension 1001 and
extension_caller_number=1001 is set as Harmeet says but the same issue
FreeSwitch the caller name and the number is 000 i just want 1001 the
caller number and the id
Edmar
Edward Q. wrote:
Sorry Edmar
I missundertood you .. I thought you wanted to change the number showing
once you were going out not the 1001.xml file.
In this case Harmeet is right. There you have those values to to make the
changes.
My bad.
Ed
On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh harm...@litatel.com
wrote:
In my case the 1001 resides in -
/usr/local/freeswitch/conf/directory/default/1001.xml
And you set the Caller Name and ID by adding -
On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. q.edw...@gmail.com wrote:
Edmar
Eso esta en freeswitch/conf/vars.xml en ese archivo.
If i am not mistaken and anyone welcome to correct me i just told Edmar
this is set in freeswitch/conf/vars.xml ... file
Ed
On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz
darklio...@yahoo.comwrote:
When I calling an outbound extension it appears:
name is FreeSWITCH and number is 0
How can i change it depends on the user who is calling?
Sample 1001-64521223
I just want the name 1001 to appear not FreeSWITCH same as the number
Thanks
--
View this message in context:
http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D0-tp24158823p24158823.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
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Re: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD
You are way off base in a few places, let me see if I can clarify a bit.
Here are at least 2 pointers:
1) The release tarballs do not come with bootstrap because they already are
bootstrapped.
2) FreeSWITCH does not depend on system libs so all the stuff about apr is
barking up the wrong tree.
we build our own apr and apr-utils
I suggest you try latest svn trunk of FS and follow the BSD build guidelines
on the WIKI since you say
it's closely compatible.
On Mon, Jun 22, 2009 at 11:08 PM, Vincent Stemen
vince.freeswi...@hightek.org wrote:
Hi.
I have been searching for an alternative PBX to asterisk (which has not
been all that stable) to run on Dragonfly BSD. I spent a fair amount of
time a couple months ago trying to compile freeswitch without success.
I have since tried Yate, but it consumes 85-95% of the cpu when idle
(not processing any calls).
I am considering revisiting freeswitch. I have included below my notes
of all the various problems I encountered before, which ones I resolved
and how, up to the point where I left off. I am in hopes of getting
feedback on whether any of the issues have been fixed or are planned to
be fixed and/or suggestions on getting it working. Also, perhaps these
notes on my experiences will be helpful for the developers to improve
freeswitch.
Keep in mind that Dragonfly is a branch from Freebsd and stays fairly
compatible. I am able to compile most software, that is ported to
FreeBSD, with few problems.
Here are my notes
=
To compile on dragonfly BSD 1.10.1-RELEASE
==
I had to add -D__FreeBSD__ to CPPFLAGS
ln sh to bash or zsh
because I got unexpected operator errors from test during configure
with the bsd shell.
ln make to gmake
Their scripts were calling make even though I ran the build
using gmake.
Must have apr-0.9.16.2.0.61 and apr-util-0.9.16.2.0.61 installed
apr-0.xxx has headers, which freeswitch is including, that apr-1.xxx
does not have.
I sym-linked the apr-util libs from apr-util-1.2.8nb1 because
apr-util-0.9.16.2.0.61 was not available as a binary package.
The freeswitch-1.0.3.tar.gz release did not have bootstrap.sh, which is
required for building from the svn repository.
Somebody on the #freeswitch IRC suggested I get it from the subversion
repository and run the bootstrap script. bootstrap.sh is not in the
release.
==
Tue Mar 31 00:12:37 CDT 2009
I posted on the freeswitch mailing list asking about the compilation errors
Got no responses.
This first set of apr_... warnings turned out to clearly be from not having
the
correct apr-util package installed, I should have gotten a response on the
list
about it, considering it is a clear dependency that they do not directly
specify on the web site or the source docs. apr-util is a dependency of
subversion. They list SVN as a dependency of freeswitch, which is the
utility
in the subversion package, not the package name. svn should not be
a dependency to build from a release archive that is not retrieved from the
svn
repository.
As it turns out, it had to be apr-util version 0.9.15. See notes below.
==
Tue Mar 31 22:22:58 CDT 2009
I tried the freeswitch-snapshot.tar.gz from the freeswitch site, which is
a 03/30/2009 snapshot from the svn trunk.
*concern*
It was nearly twice as big as the release for some reason.
27016871 Mar 28 13:08 freeswitch-1.0.3.tar.gz
52854882 Mar 31 18:25 freeswitch-snapshot.tar.gz
Running bootstrap.sh produced a bunch of these errors from automake:
Use of uninitialized value in exists at /usr/pkg/bin/automake line 4823,
GEN0 line 1.
Use of uninitialized value in concatenation (.) or string at
/usr/pkg/bin/automake line 4823, GEN0 line 1.
automake:
automake: ## Internal Error ##
automake:
automake: unrequested trace `'
automake: Please contact bug-autom...@gnu.org.
at /usr/pkg/share/automake-1.10/Automake/Channels.pm line 570
Automake::Channels::msg('automake', '', 'unrequested trace `\'')
called at /usr/pkg/share/automake-1.10/Automake/ChannelDefs.pm line 191
Automake::ChannelDefs::prog_error('unrequested trace `\'') called at
/usr/pkg/bin/automake line 4823
Automake::scan_autoconf_traces('configure.ac') called at
/usr/pkg/bin/automake line 5046
Automake::scan_autoconf_files() called at /usr/pkg/bin/automake line
781
==
Thu Apr 2 20:51:44 CDT 2009
Going back to the 1.0.3 release.
I was getting a a bunch of apr_... warnings like
Compiling src/switch_apr.c ...
src/switch_apr.c: In function `switch_thread_self':
src/switch_apr.c:74: warning: implicit declaration of function
`apr_os_thread_current'
src/switch_apr.c:74: warning: return makes pointer from integer without
a cast
Re: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio?
try adding this to your jingle profile in client.xml param name=candidate-acl value=wan/ then edit acl.conf.xml and add this list list name=wan default=allow node type=deny cidr=10.0.0.0/8/ node type=deny cidr=172.16.0.0/12/ node type=deny cidr=192.168.0.0/16/ /list this tells mod_dingaling that it should only pick candidates that pass the acl list given the one we made called wan excludes all the private ranges. If you update to latest trunk this list is created internally as wan.auto so you can use that instead of making one in your config. On Tue, Jun 23, 2009 at 7:51 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! I am trying to call from my corporate network (firewalled) using Gtalk to Freeswitch. I am not getting any audio. In the logs I see that mod_dingaling is using my internal corporate IP address which is not publically addressable. 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates 146.xx.xx.xx:50320 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable Candidate 146.xx.xx.xx:50320 Further on in the log, I can see GTalk sending a new candidate IP address to use: 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 name=rtp type=local protocol=udp username=e+JTkVHT1xEkqXGD password=fAxU6Pr1oF9Zq48U address=192.168.1.102 port=50322 pref=1.00 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 name=rtp type=stun protocol=udp username=RBqyF2XNMYLfJNoU password=DQMjon1fSVoJIRTp address=124.xxx.xxx.xxx port=50323 pref=0.90 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=relay protocol=udp username=62L5zs2FHbcUdeCJ password=KxmNgkUmZsLfuX6S address=209.xx.xxx.xxx port=19295 pref=0.50 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] Because of this, I never get audio. Any ideas how to fix this? Thanks! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Variable manipulation in the dialplan
play with it from the cli
freeswitchglobal_setvar foo=12345
API CALL [global_setvar(foo=12345)] output:
+OK
freeswitch eval ${foo:2:1}
API CALL [eval(${foo:2:1})] output:
3
freeswitch eval ${foo:2:3}
API CALL [eval(${foo:2:3})] output:
345
freeswitch eval ${foo:3:2}
API CALL [eval(${foo:3:2})] output:
45
freeswitch eval ${foo:-4:4}
API CALL [eval(${foo:-4:4})] output:
2345
On Tue, Jun 23, 2009 at 8:33 AM, Cavalera Claudio Luigi
claudio.caval...@italtel.it wrote:
Hello,
I once found in the wiki a page explaining how to substring a channel
variable,
something like
@[intra]lanman 12345 would be 345 if you do ${var:2}
I can't find that page on the wiki anymore, any hint on were it could
be? :-)
Also do you think it could be useful to extend this functionality with a
sort of Java indexOf() to extract a specific substring from a variable
(but without knowing its size like in the example above)?
Regards,
Claudio
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ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
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Re: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio?
No it snot because of this.. you have to understand how Jingle works and if you notice it has three candidates 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] Its already picked this one, maybe a packet capture would clear this up. /b On Jun 23, 2009, at 7:51 AM, Mark Campbell-Smith wrote: Because of this, I never get audio. Any ideas how to fix this? Thanks! Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help with Socket event again
Hi Michael, Using loopback solves my problem. Thanks a lot. There is a strange thing i observed though. I need to paste my extension in the default.xml file. Having them in the default directory isn't enough. Is that normal? Max. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help with Socket event again
On Jun 23, 2009, at 7:04 AM, Max Bridgewater max.bridgewa...@gmail.com wrote: Hi Michael, Using loopback solves my problem. Thanks a lot. There is a strange thing i observed though. I need to paste my extension in the default.xml file. Having them in the default directory isn't enough. Is that normal? No it isn't. What is the name of the file that has your extension and what subdir is it in? Can you pb the contents? -MC Max. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help with Socket event again
The file is located under /usr/local/freeswitch/conf/dialplan/default/. The name is: mysocket.xml. The content is: include extension name=mysocket condition field=destination_number expression=^242.* break=on-true action application=socket data=192.168.50.66:1 full / /condition /extension /include Max. On Tue, Jun 23, 2009 at 10:44 AM, Michael S Collins m...@freeswitch.orgwrote: On Jun 23, 2009, at 7:04 AM, Max Bridgewater max.bridgewa...@gmail.com wrote: Hi Michael, Using loopback solves my problem. Thanks a lot. There is a strange thing i observed though. I need to paste my extension in the default.xml file. Having them in the default directory isn't enough. Is that normal? No it isn't. What is the name of the file that has your extension and what subdir is it in? Can you pb the contents? -MC Max. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to enable compact SIP headers in mod_sofia
If you can supply a patch to expose this as a config option for us it would be appreciated. Patches can be posted to http://jira.freeswitch.org . Mike On Jun 17, 2009, at 3:22 PM, Muhammad Shahzad wrote: Ok, thanks, i will take care of it in my code where necessary. Thank you. On Thu, Jun 18, 2009 at 12:54 AM, Brian West br...@freeswitch.org wrote: Its not possible right now but you could if you enable the config option and apply the tag... its something I have thought about adding but wasn't high on my list. NTATAG_SIPFLAGS(MSG_FLG_COMPACT) http://sofia-sip.sourceforge.net/refdocs/nta/nta__tag_8h.html#346ad7ff8e886335f8fec40c65983ca6 /b On Jun 17, 2009, at 1:43 PM, Muhammad Shahzad wrote: Hi, Is it possible to enable compact SIP headers in mod_sofia configuration? If yes, then how to do so? Kindly give an example. Thank you. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Polycom configuration problems?
How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb larc...@yahoo.com wrote: I’m sorry Chris, but I don’t know where the look for the “global sip.cfg and mac/phone specific cfg” settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Chris Burns *Sent:* Monday, June 22, 2009 2:57 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the dialplan/ tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ..digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb larc...@yahoo.com wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1...@192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ 1...@192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1...@192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1...@192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ 1...@192.168.10.29) State Change CS_NEW - CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1...@192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ 1...@192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ 1...@192.168.10.29) State Change CS_INIT - CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1...@192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1...@192.168.10.29) State ROUTING 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/ 1...@192.168.10.29 SOFIA ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1...@192.168.10.29 Standard ROUTING
Re: [Freeswitch-users] Newbee: Need Help Thin Client Environment
On Mon, Jun 22, 2009 at 9:31 PM, murrah boswellotrc...@isp-systems.net wrote: Hello All, I am an absolute newbee in the voip world but have a project where I believe freeswitch will work and need very, very basic guidance on how to setup a testbed in a thin client environment. I think this would be a fairly simple matter of installing freeswitch and your softphone of choice on your ltsp server and configuring your extensions to register on 127.0.0.1, or whatever interface your freeswitch internal profile will be active on. In other words, ltsp is designed such that you can install your telephony on the server and the ltsp infrastructure will proliferate that functionality to your thin client. Install freeswitch and ekiga on the server and get ekiga to register. If you have trouble with that then this would be the place to ask. Once you get your ekiga extension registered and you are able to call voice mail, moh, etc, then log into a thin client and try the same from there. I think it will just work, but if not, that would be a good problem for the ltsp-discuss mailing list. https://lists.sourceforge.net/lists/listinfo/ltsp-discuss db ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips?
Try turning up your logging level to debug to see why the call is hanging up. Mike On Jun 19, 2009, at 7:13 AM, Edmar Cruz wrote: My freeswitch has a mysql database consists of freeswitch tables, registrations and nibblebill on mysql configured it correctly and working... Issue is when I call external ip's sometimes it works sometimes not? 2009-06-19 19:02:01 [INFO] switch_core_session.c:1040 switch_core_session_enable_heartbeat() sofia/internal/ 1...@116.5.231.40 setting session heartbeat to 1 second(s). 2009-06-19 19:02:01 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup sofia/internal/1...@116.50.231.72 [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/internal/1...@116.5.231.40 ) Ended 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/1...@116.5.231.40 [CS_DESTROY] On my acl.conf.xml I allow ip 116.5.231.40 list name=globals default=deny node type=allow cidr=116.5.231.40/32/ !-- My PC ip-- node type=allow cidr=116.5.231.41/32/ /list I put this on my external and internal profile param name=apply-inbound-acl value=globals/ And put auth-calls to false... Please help me am really near to my success here in freeswitch... Thanks... ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] channel variable sip_to_tag
if you need to use the same tags, we should be using the whole same nh in the code. There is code to do this by call uuid but I can't recall if thats for NOTIFY or INFO. If its the wrong one, we should add teh same for what you need. Mike On Jun 21, 2009, at 6:05 AM, Christian Löschenkohl wrote: hello do someone know how to get the sip_to_tag from an active call? the sip_from_tag is available as a channel variable but sip_to_tag isn't. i don't know if it is available at call setup, the fist time i see the tag=... in the sip header is the challenge response answer from fs i need this to get my aoc (advice-of-charge) implementation running, this one is based on sip info messages and has to contain the same tag's as the active call. br -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Problem with handling unanswered calls for a managed redirect
Can anyone suggest a good way to do the following;
1 - I want to be alerted [via the event API] to a new incoming call.
2 - I do not want that incoming call to be answered but just stay
ringing.
3 - Then via the API I want to send a redirect command to push the call
off to a new destination of my choice, I do not want to use the
answer/deflect sequence.
So far I've managed
1 - I see on the incoming call on the event API
2 - I used sleep 18 (3 mins) see rule below.
3 - failed - because the rule is executing a sleep command and I cannot
break in with my redirect.
extension name=Trunk_Line1
condition field=destination_number expression=^012345$
action application=set data=domain_name=x.x.x.x/
action application=ring_ready /
action application=sleep data=18/
/condition
/extension
I have tested the following works as single DP rule.
Using the fixed dial plan rule below I do get the SIP signalling I want
but of course it's a redirect immediately and to a fixed destination.
The redirect causes FS to send a 302 moved temporarily, and the move
works.
extension name=Trunk_Line1
condition field=destination_number expression=^012345$
action application=set data=domain_name=x.x.x.x/
action application=ring_ready /
action application=set data=effective_caller_id_number=
00123456789/
action application=set data=effective_caller_id_name= fred/
action application=redirect data=sip:012345678...@${domain_name}/
/condition
/extension
=
Any suggestions would be gratefully received
Richard Lamkin
richard.lam...@mettoni.com
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Re: [Freeswitch-users] Problem with handling unanswered calls for a managed redirect
On Jun 23, 2009, at 11:31 AM, Richard Lamkin wrote: Can anyone suggest a good way to do the following; 1 - I want to be alerted [via the event API] to a new incoming call. See below.. ie park. You should get an event via event socket you can decide what to do. 2 - I do not want that incoming call to be answered but just stay ringing. Can't really do it that way.. you can answer it but then you're responsible for generating ringback. And billing starts when you answer it. 3 – Then via the API I want to send a redirect command to push the call off to a new destination of my choice, I do not want to use the answer/deflect sequence. Try using park ... this way you put the call in limbo and you can send the call commands at your leisure. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_park So far I’ve managed 1 - I see on the incoming call on the event API 2 – I used sleep 18 (3 mins) see rule below. 3 – failed - because the rule is executing a sleep command and I cannot break in with my redirect. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED
if you turn up the debug logs it should tell you why. On Jun 22, 2009, at 11:38 PM, Edmar Cruz wrote: Nope. I just want to call a mobile number with no register number. Brian West-3 wrote: I'm going to guess you're calling a registered user? If so replace the @ with % /b On Jun 22, 2009, at 4:38 AM, Edmar Cruz wrote: Hi, API CALL [originate sofia/external/1...@116.50.456.212] -ERR SERVICE_NOT_IMPLEMENTED I receiving this error i dont know y? Can u help mo on this? I dialing a mobile number on this sometimes it works... Sometimes it destroys the call [CALL_DESTROY] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Sound file or lua script not played under load
Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I've changed ulimit -n to 9 but no diff, and ideas where else I might look? Regards, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Polycom configuration problems?
Basically read the polycom manual ... it is the polycom producing the dialtone and deciding when to dial the number you are entering, using its own dialplan and interdigit timers. On Tue, Jun 23, 2009 at 10:38 AM, Rupa Schomaker r...@rupa.com wrote: How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb larc...@yahoo.com wrote: I’m sorry Chris, but I don’t know where the look for the “global sip.cfg and mac/phone specific cfg” settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Chris Burns *Sent:* Monday, June 22, 2009 2:57 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the dialplan/ tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ..digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb larc...@yahoo.com wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1...@192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ 1...@192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1...@192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1...@192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ 1...@192.168.10.29) State Change CS_NEW - CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1...@192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ 1...@192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ 1...@192.168.10.29) State Change CS_INIT - CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1...@192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483
Re: [Freeswitch-users] Sound file or lua script not played under load
Does the log show anything? if the lua script fails to execute it should appear in freeswitch.log On Tue, Jun 23, 2009 at 9:45 AM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I’ve changed ulimit –n to 9 but no diff, and ideas where else I might look? Regards, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sound file or lua script not played under load
Hmm, Looking at console I'm seeing this, does this offer any additional clues to anyone? 2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 23 June 2009 17:46 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Sound file or lua script not played under load Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I've changed ulimit -n to 9 but no diff, and ideas where else I might look? Regards, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Polycom configuration problems?
Via a web browser. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 23, 2009 8:39 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb larc...@yahoo.com wrote: I'm sorry Chris, but I don't know where the look for the global sip.cfg and mac/phone specific cfg settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Chris Burns Sent: Monday, June 22, 2009 2:57 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the dialplan/ tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ..digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb larc...@yahoo.com wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1...@192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1...@192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1...@192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1...@192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1...@192.168.10.29) State Change CS_NEW - CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1...@192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/1...@192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/1...@192.168.10.29) State Change CS_INIT - CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1...@192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1...@192.168.10.29) State ROUTING 2009-06-22
Re: [Freeswitch-users] Polycom configuration problems?
Ok, most of us configure the polycoms via a provisioning interface. usually ftp or http. Anyway, when using the web interface, you want to look at: Goto the web interface, Click on SIP. Scroll down to the Local Settings section and you need to modify digitmap and digitmap timeout. the syntax is in the polycom manuals which you can donwload from polycom. On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb larc...@yahoo.com wrote: Via a web browser. *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Rupa Schomaker *Sent:* Tuesday, June 23, 2009 8:39 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Polycom configuration problems? How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb larc...@yahoo.com wrote: I’m sorry Chris, but I don’t know where the look for the “global sip.cfg and mac/phone specific cfg” settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Chris Burns *Sent:* Monday, June 22, 2009 2:57 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the dialplan/ tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ...digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb larc...@yahoo.com wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1...@192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ 1...@192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1...@192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1...@192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ 1...@192.168.10.29) State Change CS_NEW - CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1...@192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ 1...@192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ 1...@192.168.10.29) State Change CS_INIT
Re: [Freeswitch-users] Sound file or lua script not played under load
Are you making many calls share a single local_stream? This error usually means a handle open to a local_stream is not reading from that stream source, such as if you paused during playback of a local_stream. They are only a real issue if you are getting them with no calls up. On Tue, Jun 23, 2009 at 11:54 AM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hmm, Looking at console I’m seeing this, does this offer any additional clues to anyone? 2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] -- *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Nik Middleton *Sent:* 23 June 2009 17:46 *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Sound file or lua script not played under load Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I’ve changed ulimit –n to 9 but no diff, and ideas where else I might look? Regards, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] channel variable sip_to_tag
hi thank you for your reply how can we procced? br On 2009-06-23 18:20, Michael Jerris wrote: if you need to use the same tags, we should be using the whole same nh in the code. There is code to do this by call uuid but I can't recall if thats for NOTIFY or INFO. If its the wrong one, we should add teh same for what you need. Mike On Jun 21, 2009, at 6:05 AM, Christian Löschenkohl wrote: hello do someone know how to get the sip_to_tag from an active call? the sip_from_tag is available as a channel variable but sip_to_tag isn't. i don't know if it is available at call setup, the fist time i see the tag=... in the sip header is the challenge response answer from fs i need this to get my aoc (advice-of-charge) implementation running, this one is based on sip info messages and has to contain the same tag's as the active call. br -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sound file or lua script not played underload
They're reading an audio file from a ram disk. Wouldn't have thought that this would cause a problem or am I wrong. Running at around 400 concurrent calls Regards, From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 23 June 2009 19:21 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Sound file or lua script not played underload Are you making many calls share a single local_stream? This error usually means a handle open to a local_stream is not reading from that stream source, such as if you paused during playback of a local_stream. They are only a real issue if you are getting them with no calls up. On Tue, Jun 23, 2009 at 11:54 AM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hmm, Looking at console I'm seeing this, does this offer any additional clues to anyone? 2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 23 June 2009 17:46 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Sound file or lua script not played under load Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I've changed ulimit -n to 9 but no diff, and ideas where else I might look? Regards, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Sound file or lua script not played underload
the lines you pasted indicate something stuck playing local_stream (hold music) and not actually reading it. playing a file from a ram disk with 400 is for sure fine. I have done many thousand before. if you turn up your debugging do you see anything else about the box going wrong? On Tue, Jun 23, 2009 at 1:35 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: They’re reading an audio file from a ram disk. Wouldn’t have thought that this would cause a problem or am I wrong. Running at around 400 concurrent calls Regards, -- *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* 23 June 2009 19:21 *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Sound file or lua script not played underload Are you making many calls share a single local_stream? This error usually means a handle open to a local_stream is not reading from that stream source, such as if you paused during playback of a local_stream. They are only a real issue if you are getting them with no calls up. On Tue, Jun 23, 2009 at 11:54 AM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hmm, Looking at console I’m seeing this, does this offer any additional clues to anyone? 2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:43.747518 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:45.452340 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:55.582892 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:44:57.285124 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:05.217018 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:07.136734 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:16.934136 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:17.751749 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:29.489415 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:30.877783 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:42.711411 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:45:53.924364 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:07.826509 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] 2009-06-23 17:46:17.623225 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015] -- *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Nik Middleton *Sent:* 23 June 2009 17:46 *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Sound file or lua script not played under load Hi Guys, Scratching my head on this one, under load FS is not playing an audio file, OR and lua script is not getting executed. Not all the time, just some. I’ve changed ulimit –n to 9 but no diff, and ideas where else I might look? Regards, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888
Re: [Freeswitch-users] Help with Socket event again
Hi, I've got some news on this. When i move my extension to a different directory (/usr/local/freeswitch/conf/dialplan/sockets/) and add an include element at the very sample place where the default is included, things work just as expected. That is, my default.xml now include following: X-PRE-PROCESS cmd=include data=sockets/*.xml/ X-PRE-PROCESS cmd=include data=default/*.xml/ Cheers, Max. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help with Socket event again
I love it when users figure it out AND report back what they did to solve the issue! Nice work. -MC On Tue, Jun 23, 2009 at 12:11 PM, Max Bridgewater max.bridgewa...@gmail.com wrote: Hi, I've got some news on this. When i move my extension to a different directory (/usr/local/freeswitch/conf/dialplan/sockets/) and add an include element at the very sample place where the default is included, things work just as expected. That is, my default.xml now include following: X-PRE-PROCESS cmd=include data=sockets/*.xml/ X-PRE-PROCESS cmd=include data=default/*.xml/ Cheers, Max. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] ClueCon 2009 - Important Information
I know you are all eagerly anticipating the arrival of the coolest conference around! We want to make sure that everyone is aware of the following information: * The last day to get the early-bird registration is Wednesday, July 1. Early birds get into the conference for only $499. After July 1 the price is $699 per person. Please call 877.742.CLUE and get registered today! * The last day to book a hotel room at the Wyndham is Tuesday, July 21. Be sure to use expedia.com to get the best deal available. The ClueCon team is working hard to make this a very special event and we hope to have more announcements soon. You don't want to miss ClueCon 2009 - it will be the best conference you attend this year, bar none! -The ClueCon Team http://www.cluecon.com 877.742.CLUE ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio?
Also, if and when you get this working please send a message to the list. I'd like to make sure that your setup gets documented on the wiki. -MC On Tue, Jun 23, 2009 at 6:44 AM, Anthony Minessale anthony.miness...@gmail.com wrote: try adding this to your jingle profile in client.xml param name=candidate-acl value=wan/ then edit acl.conf.xml and add this list list name=wan default=allow node type=deny cidr=10.0.0.0/8/ node type=deny cidr=172.16.0.0/12/ node type=deny cidr=192.168.0.0/16/ /list this tells mod_dingaling that it should only pick candidates that pass the acl list given the one we made called wan excludes all the private ranges. If you update to latest trunk this list is created internally as wan.auto so you can use that instead of making one in your config. On Tue, Jun 23, 2009 at 7:51 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! I am trying to call from my corporate network (firewalled) using Gtalk to Freeswitch. I am not getting any audio. In the logs I see that mod_dingaling is using my internal corporate IP address which is not publically addressable. 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates 146.xx.xx.xx:50320 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable Candidate 146.xx.xx.xx:50320 Further on in the log, I can see GTalk sending a new candidate IP address to use: 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 name=rtp type=local protocol=udp username=e+JTkVHT1xEkqXGD password=fAxU6Pr1oF9Zq48U address=192.168.1.102 port=50322 pref=1.00 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 name=rtp type=stun protocol=udp username=RBqyF2XNMYLfJNoU password=DQMjon1fSVoJIRTp address=124.xxx.xxx.xxx port=50323 pref=0.90 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=relay protocol=udp username=62L5zs2FHbcUdeCJ password=KxmNgkUmZsLfuX6S address=209.xx.xxx.xxx port=19295 pref=0.50 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] Because of this, I never get audio. Any ideas how to fix this? Thanks! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Newbee: Need Help Thin Client Environment
Curious - what kinds of SIP phones do the clients support? Have you decided what you'd be using? -MC On Mon, Jun 22, 2009 at 8:31 PM, murrah boswell otrc...@isp-systems.netwrote: Hello All, I am an absolute newbee in the voip world but have a project where I believe freeswitch will work and need very, very basic guidance on how to setup a testbed in a thin client environment. I am using K12LTSP 5EL (Centos5) for my terminal server/host testbed and utilize fl_teachertool 0.07 to monitor the connected terminal clients (TCs). If you are not familiar with fl_teachertool, it allows a teacher to view thumbnail images of each TC logged in to the server. The teacher can click on any thumbnail and enlarge the view, monitor all applications running on a given TC, and take control of the keyboard and mouse of the TC. These are just a few of the capabilities of fl_teachertool. What I want to do is allow the teacher to establish voice communication using headsets and microphones with any one of the TCs by making a phone call via ethernet based upon ip of the TC through freeswitch using a softphone. Does this sound like something that is possible using freeswitch? If so, could someone please give me very basic instructions on how to setup this proof of concept? If I can just get a teacher stationed at my server talking to one student at a TC, I believe I can go from there. Currently I have a voiper softphone that functions, I believe, under gnome, but I have no idea how to configure the voiper to initiate calls through freeswitch or how to configure freeswitch to route the call to one of my TCs. I also need to keep this system fully self contained. That is, I can not have a requirement to use an outside sip service provider. Also, I would use any other linux sip softphones known to work with freeswitch that people feel would work better than a voiper. voiper seems to be more windows and mac based. I would really like to use an ekiga since they seem to be more linux based, but I do not believe that they have been thoroughly tested with freeswitch. Any help would be greatly appreciated! Regards, Murrah Boswell ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] voicemail problem
Did you ever get resolution on this? If not, join us on IRC and we'll discuss it. -MC On Sat, Jun 20, 2009 at 6:28 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! I have a problem with voicemail in that freeswitch fails to let users leave their message. Something wrong in the config I guess. I see this in the logs: 2009-06-21 11:28:32.202912 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-06-21 11:28:32.238744 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-record_message] (en:en) 2009-06-21 11:28:32.290756 [ERR] mod_native_file.c:68 Error opening /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.PCMU 2009-06-21 11:28:32.402826 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-06-21 11:28:32.434733 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-goodbye.wav] (en:en) I assume the vm-record_message.PCMU is the file that will be created to record the voicemail. Is that correct and how can I fix this? Thanks! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] voicemail problem
You're using native files and you have no native files in PCMU... /b On Jun 23, 2009, at 3:15 PM, Michael Collins wrote: Did you ever get resolution on this? If not, join us on IRC and we'll discuss it. -MC On Sat, Jun 20, 2009 at 6:28 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! I have a problem with voicemail in that freeswitch fails to let users leave their message. Something wrong in the config I guess. I see this in the logs: 2009-06-21 11:28:32.202912 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-06-21 11:28:32.238744 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-record_message] (en:en) 2009-06-21 11:28:32.290756 [ERR] mod_native_file.c:68 Error opening /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm- record_message.PCMU 2009-06-21 11:28:32.402826 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-06-21 11:28:32.434733 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-goodbye.wav] (en:en) I assume the vm-record_message.PCMU is the file that will be created to record the voicemail. Is that correct and how can I fix this? Thanks! Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Polycom configuration problems?
Thanks to Rupa and Chris for this help. I didn't know enough to understand Chris was pointing me to the Polycom phone rather than FS. I would never have figured this out. Are Polycoms the only SIP phones which have this feature? Lars From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 23, 2009 10:46 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? Ok, most of us configure the polycoms via a provisioning interface. usually ftp or http. Anyway, when using the web interface, you want to look at: Goto the web interface, Click on SIP. Scroll down to the Local Settings section and you need to modify digitmap and digitmap timeout. the syntax is in the polycom manuals which you can donwload from polycom. On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb larc...@yahoo.com wrote: Via a web browser. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Tuesday, June 23, 2009 8:39 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb larc...@yahoo.com wrote: I'm sorry Chris, but I don't know where the look for the global sip.cfg and mac/phone specific cfg settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Chris Burns Sent: Monday, June 22, 2009 2:57 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the dialplan/ tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ...digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb larc...@yahoo.com wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1...@192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/1...@192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1...@192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1...@192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/1...@192.168.10.29) State Change CS_NEW - CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.10.29 [BREAK] 2009-06-22
Re: [Freeswitch-users] Polycom configuration problems?
Nope other phones have this also. /b On Jun 23, 2009, at 4:57 PM, Lars Zeb wrote: Thanks to Rupa and Chris for this help. I didn’t know enough to understand Chris was pointing me to the Polycom phone rather than FS. I would never have figured this out. Are Polycoms the only SIP phones which have this feature? Lars Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Polycom configuration problems?
Every sip phone I've used has this feature. Even ATAs -- though they tend to ship with more forgiving defaults. On Tue, Jun 23, 2009 at 4:57 PM, Lars Zeb larc...@yahoo.com wrote: Thanks to Rupa and Chris for this help. I didn’t know enough to understand Chris was pointing me to the Polycom phone rather than FS. I would never have figured this out. Are Polycoms the only SIP phones which have this feature? Lars *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Rupa Schomaker *Sent:* Tuesday, June 23, 2009 10:46 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Polycom configuration problems? Ok, most of us configure the polycoms via a provisioning interface. usually ftp or http. Anyway, when using the web interface, you want to look at: Goto the web interface, Click on SIP. Scroll down to the Local Settings section and you need to modify digitmap and digitmap timeout. the syntax is in the polycom manuals which you can donwload from polycom. On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb larc...@yahoo.com wrote: Via a web browser. *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Rupa Schomaker *Sent:* Tuesday, June 23, 2009 8:39 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Polycom configuration problems? How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb larc...@yahoo.com wrote: I’m sorry Chris, but I don’t know where the look for the “global sip.cfg and mac/phone specific cfg” settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Chris Burns *Sent:* Monday, June 22, 2009 2:57 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the dialplan/ tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb larc...@yahoo.com wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1...@192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ 1...@192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1...@192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1...@192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG]
Re: [Freeswitch-users] Transmit fax locally for test
Did anyone have any suggestions on this? Just to reiterate...
- 8000 is a local extension defined in the default dialplan... see
http://pastebin.freeswitch.org/9450 for definition
- didn't work: originate sofia/default/8...@192.168.10.35
txfax(storage/fax/test.tif) ... see http://pastebin.freeswitch.org/9440 for log
- had to add the FS ip (192.168.10.35) to the domains acl... now it to works
list name=domains default=deny
node type=allow cidr=192.168.10.35/32/
node type=allow domain=$${domain}/
/list
Is this the proper way to configure?
Tim
From: timb0...@hotmail.com
To: freeswitch-users@lists.freeswitch.org
Subject: RE: Transmit fax locally for test
Date: Mon, 22 Jun 2009 18:37:47 -0400
8000 is a local extension defined in the default dialplan.
Tim
--
Message: 2
Date: Mon, 22 Jun 2009 15:05:20 -0400
From: Brian West br...@freeswitch.org
Subject: Re: [Freeswitch-users] Transmit fax locally for test
To: freeswitch-users@lists.freeswitch.org
Message-ID: 8618988e-bb27-4400-bddf-99c87a26f...@freeswitch.org
Content-Type: text/plain; charset=us-ascii
what is 8000? is it local or is it a remote endpoint?
/b
On Jun 22, 2009, at 3:01 PM, Tim B wrote:
originate sofia/default/8...@192.168.10.35 txfax(storage/fax/
test.tif)
Brian West
br...@freeswitch.org
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Re: [Freeswitch-users] Problem with handling unanswered calls for a managed redirect
Brian, Thank you for suggesting I try PARK. I tried PARK but unfortunately it sends out a 183 with SDP which stops the originator hearing ringing (ring back). If you know of a way to park without sending a 183 that would solve my problem. Regards Richard Lamkin richard.lam...@mettoni.com From: Brian West [mailto:br...@freeswitch.org] Sent: 23 June 2009 17:37 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem with handling unanswered calls for amanaged redirect On Jun 23, 2009, at 11:31 AM, Richard Lamkin wrote: Can anyone suggest a good way to do the following; 1 - I want to be alerted [via the event API] to a new incoming call. See below.. ie park. You should get an event via event socket you can decide what to do. 2 - I do not want that incoming call to be answered but just stay ringing. Can't really do it that way.. you can answer it but then you're responsible for generating ringback. And billing starts when you answer it. 3 - Then via the API I want to send a redirect command to push the call off to a new destination of my choice, I do not want to use the answer/deflect sequence. Try using park ... this way you put the call in limbo and you can send the call commands at your leisure. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_park So far I've managed 1 - I see on the incoming call on the event API 2 - I used sleep 18 (3 mins) see rule below. 3 - failed - because the rule is executing a sleep command and I cannot break in with my redirect. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com http://www.cluecon.com/ * Please consider the environment before printing this e-mail * This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN * ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Transmit fax locally for test
Is 8000 just a dialplan extension? I'm curious about the whole
8...@192.168.10.35 thing. I doubt that's necessary. For kicks try something
like this:
originate loopback/8000 txfax(storage/fax/test.tif)
That will drop the A leg right into extension 8000.
-MC
On Tue, Jun 23, 2009 at 3:12 PM, Tim B timb0...@hotmail.com wrote:
Did anyone have any suggestions on this? Just to reiterate...
- 8000 is a local extension defined in the default dialplan... see
http://pastebin.freeswitch.org/9450 for definition
- didn't work: originate
sofia/default/8...@192.168.10.35txfax(storage/fax/test.tif) ... see
http://pastebin.freeswitch.org/9440 for log
- had to add the FS ip (192.168.10.35) to the domains acl... now it to
works
list name=domains default=deny
node type=allow cidr=192.168.10.35/32/
node type=allow domain=$${domain}/
/list
Is this the proper way to configure?
Tim
--
From: timb0...@hotmail.com
To: freeswitch-users@lists.freeswitch.org
Subject: RE: Transmit fax locally for test
Date: Mon, 22 Jun 2009 18:37:47 -0400
8000 is a local extension defined in the default dialplan.
Tim
--
Message: 2
Date: Mon, 22 Jun 2009 15:05:20 -0400
From: Brian West br...@freeswitch.org
Subject: Re: [Freeswitch-users] Transmit fax locally for test
To: freeswitch-users@lists.freeswitch.org
Message-ID: 8618988e-bb27-4400-bddf-99c87a26f...@freeswitch.org
Content-Type: text/plain; charset=us-ascii
what is 8000? is it local or is it a remote endpoint?
/b
On Jun 22, 2009, at 3:01 PM, Tim B wrote:
originate sofia/default/8...@192.168.10.35 txfax(storage/fax/
test.tif)
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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Re: [Freeswitch-users] Newbee: Need Help Thin Client Environment
Curious - what kinds of SIP phones do the clients support? Have you decided what you'd be using? -MC I am still experimenting! I have a zoiper 2.0 installed on one of my test clients. zoiper seems to work fine, so now I am attempting to get the freeswitch/zoiper interface working. I will also try to get an ekiga working, but first the zoiper. First I have to figure out how to get freeswitch operational and will be working on that tonight! Regards, Murrah Boswell On Mon, Jun 22, 2009 at 8:31 PM, murrah boswell otrc...@isp-systems.netwrote: Hello All, I am an absolute newbee in the voip world but have a project where I believe freeswitch will work and need very, very basic guidance on how to setup a testbed in a thin client environment. I am using K12LTSP 5EL (Centos5) for my terminal server/host testbed and utilize fl_teachertool 0.07 to monitor the connected terminal clients (TCs). If you are not familiar with fl_teachertool, it allows a teacher to view thumbnail images of each TC logged in to the server. The teacher can click on any thumbnail and enlarge the view, monitor all applications running on a given TC, and take control of the keyboard and mouse of the TC. These are just a few of the capabilities of fl_teachertool. What I want to do is allow the teacher to establish voice communication using headsets and microphones with any one of the TCs by making a phone call via ethernet based upon ip of the TC through freeswitch using a softphone. Does this sound like something that is possible using freeswitch? If so, could someone please give me very basic instructions on how to setup this proof of concept? If I can just get a teacher stationed at my server talking to one student at a TC, I believe I can go from there. Currently I have a voiper softphone that functions, I believe, under gnome, but I have no idea how to configure the voiper to initiate calls through freeswitch or how to configure freeswitch to route the call to one of my TCs. I also need to keep this system fully self contained. That is, I can not have a requirement to use an outside sip service provider. Also, I would use any other linux sip softphones known to work with freeswitch that people feel would work better than a voiper. voiper seems to be more windows and mac based. I would really like to use an ekiga since they seem to be more linux based, but I do not believe that they have been thoroughly tested with freeswitch. Any help would be greatly appreciated! Regards, Murrah Boswell ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Transmit fax locally for test
Yeah 8000 is just a dialplan extension. That worked...
originate loopback/8000 txfax(storage/fax/test.tif)
Thanks MC. I guess the loopback bypasses all the security stuff and jumps
right into the dialplan looking for a matching # condition?
Tim
--
Message: 3
Date: Tue, 23 Jun 2009 16:14:11 -0700
From: Michael Collins m...@freeswitch.org
Subject: Re: [Freeswitch-users] Transmit fax locally for test
To: freeswitch-users@lists.freeswitch.org
Message-ID:
87f2f3b90906231614t6223f65cr64e3dc492564a...@mail.gmail.com
Content-Type: text/plain; charset=windows-1252
Is 8000 just a dialplan extension? I'm curious about the whole
8...@192.168.10.35 thing. I doubt that's necessary. For kicks try something
like this:
originate loopback/8000 txfax(storage/fax/test.tif)
That will drop the A leg right into extension 8000.
-MC
On Tue, Jun 23, 2009 at 3:12 PM, Tim B timb0...@hotmail.com wrote:
Did anyone have any suggestions on this? Just to reiterate...
- 8000 is a local extension defined in the default dialplan... see
http://pastebin.freeswitch.org/9450 for definition
- didn't work: originate
sofia/default/8...@192.168.10.35txfax(storage/fax/test.tif) ... see
http://pastebin.freeswitch.org/9440 for log
- had to add the FS ip (192.168.10.35) to the domains acl... now it to
works
list name=domains default=deny
node type=allow cidr=192.168.10.35/32/
node type=allow domain=$${domain}/
/list
Is this the proper way to configure?
Tim
--
From: timb0...@hotmail.com
To: freeswitch-users@lists.freeswitch.org
Subject: RE: Transmit fax locally for test
Date: Mon, 22 Jun 2009 18:37:47 -0400
8000 is a local extension defined in the default dialplan.
Tim
--
Message: 2
Date: Mon, 22 Jun 2009 15:05:20 -0400
From: Brian West br...@freeswitch.org
Subject: Re: [Freeswitch-users] Transmit fax locally for test
To: freeswitch-users@lists.freeswitch.org
Message-ID: 8618988e-bb27-4400-bddf-99c87a26f...@freeswitch.org
Content-Type: text/plain; charset=us-ascii
what is 8000? is it local or is it a remote endpoint?
/b
On Jun 22, 2009, at 3:01 PM, Tim B wrote:
originate sofia/default/8...@192.168.10.35 txfax(storage/fax/
test.tif)
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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Re: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips?
Where can i find this logs? Michael Jerris wrote: Try turning up your logging level to debug to see why the call is hanging up. Mike On Jun 19, 2009, at 7:13 AM, Edmar Cruz wrote: My freeswitch has a mysql database consists of freeswitch tables, registrations and nibblebill on mysql configured it correctly and working... Issue is when I call external ip's sometimes it works sometimes not? 2009-06-19 19:02:01 [INFO] switch_core_session.c:1040 switch_core_session_enable_heartbeat() sofia/internal/ 1...@116.5.231.40 setting session heartbeat to 1 second(s). 2009-06-19 19:02:01 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup sofia/internal/1...@116.50.231.72 [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/internal/1...@116.5.231.40 ) Ended 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/1...@116.5.231.40 [CS_DESTROY] On my acl.conf.xml I allow ip 116.5.231.40 list name=globals default=deny node type=allow cidr=116.5.231.40/32/ !-- My PC ip-- node type=allow cidr=116.5.231.41/32/ /list I put this on my external and internal profile param name=apply-inbound-acl value=globals/ And put auth-calls to false... Please help me am really near to my success here in freeswitch... Thanks... ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/Freeswitch-Warning-Cannot-Call-External-Ips--tp24109532p24177512.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD
Thanks for the response Anthony. On Tue, Jun 23, 2009 at 08:29:13AM -0500, Anthony Minessale wrote: You are way off base in a few places, let me see if I can clarify a bit. Here are at least 2 pointers: 1) The release tarballs do not come with bootstrap because they already are bootstrapped. 2) FreeSWITCH does not depend on system libs so all the stuff about apr is barking up the wrong tree. we build our own apr and apr-utils Interesting. I do not know why I got the errors I mentioned before then until I installed the exact versions of those packages it seemed to need. I suggest you try latest svn trunk of FS and follow the BSD build guidelines on the WIKI since you say it's closely compatible. Ok. I did this. Compilation still failed but there are significant improvements since the last time. Here is what I did and the results: Checked out the current trunk with svn. Patched /usr/include/sys/resource.h Since Dragonfly has fixed or will be fixing this future releases I patched the system header to add RLIMIT_AS rather than patching freeswitch to use RLIMIT_VMEM. Compilation still failed but there are significant improvements. bootstrap.sh seems to have been successful this time. I seems to have worked with the bsd shell this time. I also did not have to link make to gmake. It appears to have properly called gmake when building in sub-directories when gmake was run from the top. Configure completed successfully but there were these warnings: checking dlfcn.h usability... no checking dlfcn.h presence... yes configure: WARNING: dlfcn.h: present but cannot be compiled configure: WARNING: dlfcn.h: check for missing prerequisite headers? configure: WARNING: dlfcn.h: see the Autoconf documentation configure: WARNING: dlfcn.h: section Present But Cannot Be Compiled configure: WARNING: dlfcn.h: proceeding with the preprocessor's result configure: WARNING: dlfcn.h: in the future, the compiler will take precedence checking for dlfcn.h... yes I do not know if this is going to cause a problem. I did not have to use the --build=i386 option to configure this time. Compiling = Still lots of warnings of: warning: return makes pointer from integer without a cast Errors: It is apparently not checking return codes from make. It continues even when there are errors. Is this intentional?? su_alloc.c: In function `su_salloc': su_alloc.c:1518: warning: return makes pointer from integer without a cast gmake[9]: *** [su_alloc.lo] Error 1 gmake[8]: *** [all] Error 2 Making all in features LTCOMPILE features.lo ... Making all in sresolv LTCOMPILE sres.lo LTCOMPILE sres_cache.lo LTCOMPILE sres_blocking.lo LTCOMPILE sresolv.lo LTCOMPILE sres_sip.lo sres_sip.c: In function `sres_sip_new': sres_sip.c:267: warning: return makes pointer from integer without a cast gmake[8]: *** [sres_sip.lo] Error 1 Making all in ipt LTCOMPILE base64.lo LTCOMPILE token64.lo LINK libipt.la ... There are about 12 errors of this nature before ending with Making all in nua LTCOMPILE nua.lo nua.c: In function `nua_create': nua.c:141: warning: return makes pointer from integer without a cast nua.c:144: warning: return makes pointer from integer without a cast gmake[9]: *** [nua.lo] Error 1 gmake[8]: *** [all] Error 2 gmake[8]: *** No rule to make target `iptsec/libiptsec.la', needed by `libsofia-sip-ua.la'. Stop. gmake[7]: *** [all-recursive] Error 1 Making all in packages gmake[6]: *** [all-recursive] Error 1 gmake[5]: *** [all] Error 2 gmake[4]: *** [/u1/falcon/ports/freeswitch-20090623/work/freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 gmake[3]: *** [mod_sofia-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + gmake install + +--+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 It says it has been successfully built. Apparently part of the same problem of not checking the return codes. It does not say what most of the errors are except for near the last when it says No rule to make target `iptsec/libiptsec.la' It just says Error 1 or Error 2 which does not tell me what the problem is. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_dingaling picking wrong IP address / no audio?
Thanks Anthony.
I am getting closer. I had to put in the 146 address, which is the
firewalled address I get at work. The problem now is that when the
call is bridged, I do not hear audio.
2 scenarios:
1 - the local extension is not registered. There is two way audio -
I hear the voicemail in Gtalk and I can leave a message which can then
be played back.
2 - the local extension is registered. There is no audio
In my incoming dialplan I am doing this bridge: action
application=bridge data=user/1...@${domain}/
It bridges okay, the phone rings, but there is no audio.
On a side note: Isn't putting the candidate-acl list a temporary
measure? When I travel, I will most likely get a different internal
company IP address that does not start with 146. Isn't there a
smarter way for dingaling to know that there is no RTP packets being
received and then modify which candidate should be used?
Thanks!
On Tue, Jun 23, 2009 at 6:44 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
try adding this to your jingle profile in client.xml
param name=candidate-acl value=wan/
then edit acl.conf.xml and add this list
list name=wan default=allow
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
this tells mod_dingaling that it should only pick candidates that pass the
acl list given
the one we made called wan excludes all the private ranges.
If you update to latest trunk this list is created internally as wan.auto
so you can use that
instead of making one in your config.
On Tue, Jun 23, 2009 at 7:51 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
I am trying to call from my corporate network (firewalled) using Gtalk
to Freeswitch. I am not getting any audio.
In the logs I see that mod_dingaling is using my internal corporate IP
address which is not publically addressable.
2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing
session for 4085152502
2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates
2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates
146.xx.xx.xx:50320
2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable
Candidate 146.xx.xx.xx:50320
Further on in the log, I can see GTalk sending a new candidate IP
address to use:
2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1
name=rtp
type=local
protocol=udp
username=e+JTkVHT1xEkqXGD
password=fAxU6Pr1oF9Zq48U
address=192.168.1.102
port=50322
pref=1.00
2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing
session for 4085152502
2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked
an IP [146.xx.xx.xx]
and
2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2
name=rtp
type=stun
protocol=udp
username=RBqyF2XNMYLfJNoU
password=DQMjon1fSVoJIRTp
address=124.xxx.xxx.xxx
port=50323
pref=0.90
2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing
session for 4085152502
2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked
an IP [146.xx.xx.xx]
and
2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3
name=rtp
type=relay
protocol=udp
username=62L5zs2FHbcUdeCJ
password=KxmNgkUmZsLfuX6S
address=209.xx.xxx.xxx
port=19295
pref=0.50
2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing
session for 4085152502
2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked
an IP [146.xx.xx.xx]
Because of this, I never get audio. Any ideas how to fix this?
Thanks!
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Re: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD
On Tue, Jun 23, 2009 at 09:15:30PM -0500, Vincent Stemen wrote: Ok. I did this. Compilation still failed but there are significant improvements since the last time. Here is what I did and the results: It looks like some the games that sofia plays with errno makes Dragonfly unhappy. I also noticed that where the code checks for BSD-like systems (*BSD and OSX) in libsofia-sip-ua/su/sofia-sip/su_errno.h, DragonFly is omitted, so obviously one of the first steps would be to fix that (if applicable). If you disable mod_sofia in modules conf, do the rest of the default modules build OK? For the record, DragonFly and FreeBSD have rather seriously diverged at this point, DragonFly forked from FreeBSD back in the 4.10 days or so and has changed a *lot* of things since, so I don't think it's gonna be quite as easy as you expected (but it's far from impossible either). Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] email core dump
Thanks Brian, but still no luck with the email.. I have configured exim4 so that I can send messages from the command line using 'mail' command and these are sent successfully. I still get a core dump in the log when freeswitch is trying to send the mail: /bin/cat: write error: Broken pipe sh: line 1: 4492 Done(1) /bin/cat /tmp/mail.1245811149abdc 4493 Segmentation fault (core dumped) | /usr/local/bin/eximcompat.sh -t x...@xx.com 2009-06-24 12:39:09.285351 [DEBUG] switch_utils.c:554 Emailed file [/tmp/mail.1245811149abdc] to [...@xx.com] 2009-06-24 12:39:09.285351 [DEBUG] mod_voicemail.c:2491 Sending message to x...@xx.com eximcompat.sh is as described on the wiki: freeswitch:/# cat /usr/local/bin/eximcompat.sh #!/bin/bash exec exim4 -t Any other thoughts? From: Brian West br...@freeswitch.org Subject: Re: [Freeswitch-users] email core dump To: freeswitch-users@lists.freeswitch.org Message-ID: 7c7a8ed9-eced-4100-87f6-0875c054e...@freeswitch.org Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings /b On Jun 21, 2009, at 6:27 AM, Mark Campbell-Smith wrote: Hi! I am trying to email from 2009-06-21 20:43:24.273895 [DEBUG] switch_core_codec.c:122 Restore original codec. 2009-06-21 20:43:24.273895 [DEBUG] mod_voicemail.c:2321 Deliver VM to 1...@192.168.0.20 /bin/cat: write error: Broken pipe sh: line 1: 11975 Done(1) /bin/cat /tmp/mail. 124558382500b1 11976 Segmentation fault (core dumped) | exim4 -t myem...@xx.com 2009-06-21 20:43:24.670899 [DEBUG] switch_utils.c:554 Emailed file [/tmp/mail.12455810042c7f] to [myem...@xx.com] I can manually send an email to myself with exim4, but freeswitch fails. Any ideas what I have configured incorrectly? Thanks ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Freeswitch Warning Cannot Call External Ips?
Please see the debugging pages on the wiki On Jun 23, 2009, at 10:10 PM, Edmar Cruz darklio...@yahoo.com wrote: Where can i find this logs? Michael Jerris wrote: Try turning up your logging level to debug to see why the call is hanging up. Mike On Jun 19, 2009, at 7:13 AM, Edmar Cruz wrote: My freeswitch has a mysql database consists of freeswitch tables, registrations and nibblebill on mysql configured it correctly and working... Issue is when I call external ip's sometimes it works sometimes not? 2009-06-19 19:02:01 [INFO] switch_core_session.c:1040 switch_core_session_enable_heartbeat() sofia/internal/ 1...@116.5.231.40 setting session heartbeat to 1 second(s). 2009-06-19 19:02:01 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup sofia/internal/1...@116.50.231.72 [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 7 (sofia/internal/1...@116.5.231.40 ) Ended 2009-06-19 19:02:01 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/1...@116.5.231.40 [CS_DESTROY] On my acl.conf.xml I allow ip 116.5.231.40 list name=globals default=deny node type=allow cidr=116.5.231.40/32/ !-- My PC ip-- node type=allow cidr=116.5.231.41/32/ /list I put this on my external and internal profile param name=apply-inbound-acl value=globals/ And put auth-calls to false... Please help me am really near to my success here in freeswitch... Thanks... ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/Freeswitch-Warning-Cannot-Call-External-Ips--tp24109532p24177512.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD
On Jun 23, 2009, at 10:15 PM, Vincent Stemen vince.freeswi...@hightek.org wrote: Thanks for the response Anthony. On Tue, Jun 23, 2009 at 08:29:13AM -0500, Anthony Minessale wrote: You are way off base in a few places, let me see if I can clarify a bit. Here are at least 2 pointers: 1) The release tarballs do not come with bootstrap because they already are bootstrapped. 2) FreeSWITCH does not depend on system libs so all the stuff about apr is barking up the wrong tree. we build our own apr and apr-utils Interesting. I do not know why I got the errors I mentioned before then until I installed the exact versions of those packages it seemed to need. I suggest you try latest svn trunk of FS and follow the BSD build guidelines on the WIKI since you say it's closely compatible. Ok. I did this. Compilation still failed but there are significant improvements since the last time. Here is what I did and the results: Checked out the current trunk with svn. Patched /usr/include/sys/resource.h Since Dragonfly has fixed or will be fixing this future releases I patched the system header to add RLIMIT_AS rather than patching freeswitch to use RLIMIT_VMEM. Can we make a patch ifdefing on RLIMIT_AS to make this always work without patches to system header files? Compilation still failed but there are significant improvements. bootstrap.sh seems to have been successful this time. I seems to have worked with the bsd shell this time. I also did not have to link make to gmake. It appears to have properly called gmake when building in sub-directories when gmake was run from the top. Configure completed successfully but there were these warnings: checking dlfcn.h usability... no checking dlfcn.h presence... yes configure: WARNING: dlfcn.h: present but cannot be compiled configure: WARNING: dlfcn.h: check for missing prerequisite headers? configure: WARNING: dlfcn.h: see the Autoconf documentation configure: WARNING: dlfcn.h: section Present But Cannot Be Compiled configure: WARNING: dlfcn.h: proceeding with the preprocessor's result configure: WARNING: dlfcn.h: in the future, the compiler will take precedence checking for dlfcn.h... yes This is probably fine, it means what it says, it won't try to compile with them bit the issue should probably be reported to distro maintainers I do not know if this is going to cause a problem. I did not have to use the --build=i386 option to configure this time. Compiling = Still lots of warnings of: warning: return makes pointer from integer without a cast Errors: It is apparently not checking return codes from make. It continues even when there are errors. Is this intentional?? su_alloc.c: In function `su_salloc': su_alloc.c:1518: warning: return makes pointer from integer without a cast gmake[9]: *** [su_alloc.lo] Error 1 gmake[8]: *** [all] Error 2 Making all in features LTCOMPILE features.lo ... Making all in sresolv LTCOMPILE sres.lo LTCOMPILE sres_cache.lo LTCOMPILE sres_blocking.lo LTCOMPILE sresolv.lo LTCOMPILE sres_sip.lo sres_sip.c: In function `sres_sip_new': sres_sip.c:267: warning: return makes pointer from integer without a cast gmake[8]: *** [sres_sip.lo] Error 1 Making all in ipt LTCOMPILE base64.lo LTCOMPILE token64.lo LINK libipt.la ... There are about 12 errors of this nature before ending with Making all in nua LTCOMPILE nua.lo nua.c: In function `nua_create': nua.c:141: warning: return makes pointer from integer without a cast nua.c:144: warning: return makes pointer from integer without a cast gmake[9]: *** [nua.lo] Error 1 gmake[8]: *** [all] Error 2 gmake[8]: *** No rule to make target `iptsec/libiptsec.la', needed by `libsofia-sip-ua.la'. Stop. gmake[7]: *** [all-recursive] Error 1 Making all in packages gmake[6]: *** [all-recursive] Error 1 gmake[5]: *** [all] Error 2 gmake[4]: *** [/u1/falcon/ports/freeswitch-20090623/work/ freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip- ua.la] Error 2 gmake[3]: *** [mod_sofia-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + gmake install + +--+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 Can you post a bug to Jira.freeswitch.org with all these warnings, even better with patches to fix it. It says it has been successfully built. Apparently part of the same problem
[Freeswitch-users] Nibblebill and multiple gateway
Dear All,
Look like nibblebill does't work with multiple gatreway.
I try
action application=set
data=nibble_account=0838833133/
action application=bridge
data={absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626734...@203.xxx.xxx.xxx
|[nibble_rate=0.5]sofia/external/6626734...@202.xxx.xxx.xxx
nibblebill not found nibble_rate
But
action application=set data=nibble_rate=0.05/
action application=set
data=nibble_account=0838833133/
action application=bridge
data={absolute_codec_string='GSM,G729'}sofia/external/6626734...@203.xxx.xxx.xxx
|sofia/external/6626734...@202.xxx.xxx.xxx
Work fine
What's difference from set application and [] ?
Best Regards.
Dome C.
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