[Freeswitch-users] Question about bridging calls to a specific URI via a specific profile
Hello, I was wondering, I am bridging a call to a specific URI as follows: action application=bridge data=sofia/internal/3032223...@3.55.66.180;fs_nat=yes;fs_path=sip%3As%403.5 5.66.180%3A7812 EXECUTE sofia/internal/+1720946@ mailto:sofia/internal/+17209460...@2.3.4.5 2.3.4.5 bridge(sofia/internal/3032223...@3.55.66.180;fs_nat=yes;fs_path=sip%3As%403. 55.66.180%3A7812) 2009-06-22 00:16:15.722872 [WARNING] mod_sofia.c:2687 Cannot locate registered user 3032223...@3.55.66.180;fs_nat=yes;fs_path=...@3as%403.55.66.180%3a7812 The fs_nat and fs_path info and domain are coming from a previous dialplan app that looked up a user's registered info via sofia_contact. I am replacing the registered user's SIP username with the DID being called (3032223232 in this case) My understanding of bridging a call is that if I specify sofia/profile/u...@domain that FS will use the specified SIP profile to try and connect a call to the u...@domain specified. Since the full u...@domain was specified, there is no reason to lookup the registered user - the call will just be delivered as a sip call to sip:x...@domain . However, adding fs_nat=yes;fs_path=XXX seems to cause FS to look in the user directory for a matching user. Why is this? Maybe I could use a better understanding of how fs_nat and fs_path work, but I couldn't find much on the Wiki about them. Does appending them automatically cause FS to look for the user being contacted in the directory, as opposed to just using the fs_path variable? Is this behavior from fs_nat alone? Any explanation would be helpful. Thanks, Darren Schreiber ___ 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] Question about bridging calls to a specific URIvia a specific profile
Ignore this thread. Apparently I was stripping sip: from the prefix. I guess you have to specify sip: before utilizing fs_nat and fs_path variables. My bad. _ From: Darren Schreiber [mailto:d...@d-man.org] Sent: Monday, June 22, 2009 12:32 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Question about bridging calls to a specific URIvia a specific profile Hello, I was wondering, I am bridging a call to a specific URI as follows: action application=bridge data=sofia/internal/3032223...@3.55.66.180;fs_nat=yes;fs_path=sip%3As%403.5 5.66.180%3A7812 EXECUTE sofia/internal/+1720946@ mailto:sofia/internal/+17209460...@2.3.4.5 2.3.4.5 bridge(sofia/internal/3032223...@3.55.66.180;fs_nat=yes;fs_path=sip%3As%403. 55.66.180%3A7812) 2009-06-22 00:16:15.722872 [WARNING] mod_sofia.c:2687 Cannot locate registered user 3032223...@3.55.66.180;fs_nat=yes;fs_path=...@3as%403.55.66.180%3a7812 The fs_nat and fs_path info and domain are coming from a previous dialplan app that looked up a user's registered info via sofia_contact. I am replacing the registered user's SIP username with the DID being called (3032223232 in this case) My understanding of bridging a call is that if I specify sofia/profile/u...@domain that FS will use the specified SIP profile to try and connect a call to the u...@domain specified. Since the full u...@domain was specified, there is no reason to lookup the registered user - the call will just be delivered as a sip call to sip:x...@domain . However, adding fs_nat=yes;fs_path=XXX seems to cause FS to look in the user directory for a matching user. Why is this? Maybe I could use a better understanding of how fs_nat and fs_path work, but I couldn't find much on the Wiki about them. Does appending them automatically cause FS to look for the user being contacted in the directory, as opposed to just using the fs_path variable? Is this behavior from fs_nat alone? Any explanation would be helpful. Thanks, Darren Schreiber ___ 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 sound-path when switch language
default_language still don't work wirh say
but sound_prefix work fine.
example my javascript
---
session.execute(set, sound_prefix=/opt/freeswitch/sounds/th);
session.execute(say,th number pronounced 1346523);
session.execute(say,th number pronounced 21);
session.execute(say,th number pronounced 11);
session.execute(say,th number pronounced 101);
How to check in mod_say_th back to freeswotch ?
Dome C.
2009/6/3 Brian West br...@freeswitch.org:
You'll need to set the variable default_language
/b
On Jun 2, 2009, at 2:42 PM, Dome Charoenyost wrote:
Dear sir,
i create mod_say_th for Thai language. i found some problem
about sound-path.
I have config th.xml in conf/lang/th/
language name=en sound-path=$${base_dir}/sounds/en
tts-engine=cepstral tts-voice=callie
...
when i try
action application=say data=th number pronounced 20230021/
Freeswitch still looking sounf file in /sounds/en/us/callie (en
sound-path)
Someone help me please
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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[Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED
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] Thanks -- View this message in context: http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-tp24143545p24143545.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] Nibblebill heartbeat on B-leg
Hello,
I am using Nibblebill to bill bridged calls initiated via API.
The problem is that the billing on the B-leg is only done when the call is
terminated and not at heartbeat.
For the a-leg it is working. I added the global_heartbeat variable at the
b-leg but without success.
I use now:
originate
{ignore_early_media=true,nibble_account=100,nibble_rate=0.02}sofia/external/12...@serverip
bridge({global_heartbeat=50,nibble_account=100,nibble_rate=0.02}sofia/external/45...@serverip)
thx,
MdM
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Sent from the Freeswitch-users mailing list archive at Nabble.com.
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Re: [Freeswitch-users] How to change sound-path when switch language
Please open a jira about this. /b On Jun 22, 2009, at 4:10 AM, Dome Charoenyost wrote: default_language still don't work wirh say but sound_prefix work fine. example my javascript --- session.execute(set, sound_prefix=/opt/freeswitch/sounds/th); session.execute(say,th number pronounced 1346523); session.execute(say,th number pronounced 21); session.execute(say,th number pronounced 11); session.execute(say,th number pronounced 101); How to check in mod_say_th back to freeswotch ? Dome C. 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] rpm build issues on CentOS 5 with trunk
It's failing to build the core library. There should be some warning before it tried to build the modules in the log. You'll have to bear with me, I am not sure exactly what part of the build that is, I see this in the log: + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /usr/bin/make install + +--+ followed by this: + FreeSWITCH install Complete --+ + FreeSWITCH has been successfully installed. + + + + Install sounds: + + (uhd-sounds includes hd-sounds, sounds) + + (hd-sounds includes sounds) + + + and finally it ends with: Checking for unpackaged file(s): /usr/lib/rpm/check-files /var/tmp/freeswitch-1.0.4-1-root-rpmbuilder RPM build errors: Which doesn't help:) Thanks! jlc ___ 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] rpm build issues on CentOS 5 with trunk
Scroll up and look for the real error. /b On Jun 22, 2009, at 10:36 AM, Joseph L. Casale wrote: You'll have to bear with me, I am not sure exactly what part of the build that is, I see this in the log: + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /usr/bin/make install + +--+ followed by this: + FreeSWITCH install Complete --+ + FreeSWITCH has been successfully installed. + + + + Install sounds: + + (uhd-sounds includes hd-sounds, sounds) + + (hd-sounds includes sounds) + + + and finally it ends with: Checking for unpackaged file(s): /usr/lib/rpm/check-files /var/tmp/ freeswitch-1.0.4-1-root-rpmbuilder RPM build errors: Which doesn't help:) Thanks! jlc 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
originate {origination_caller_id_number=120003}sofia/internal/242...@192.168.50.67
socket(192.168.50.67:1 full)
/b
On Jun 22, 2009, at 11:57 AM, Max Bridgewater wrote:
Hi Mike,
Unfortunately this doesn't seem to solve my problem. Here is my
extension again:
extension name=mysocket
condition field=destination_number expression=^242.*
action application=socket data=192.168.50.67:1
full /
/condition
/extension
I've copied it now under:
/user/local/freeswitch/conf/dialplan/default
/user/local/freeswitch/conf/dialplan/public
The different dial strings i tried:
{origination_caller_id_number=120003}sofia/internal/242...@192.168.50.67
park()
{origination_caller_id_number=120003}sofia/internal/242...@192.168.50.67
{origination_caller_id_number=120003}sofia/internal/242424
{origination_caller_id_number=120003}sofia/internal/
242424%192.168.50.62
My goal: have the call captured by the above extension and
redirected to a server socket running at 192.168.50.67:1.
Any thought?
Max.
On Fri, Jun 19, 2009 at 5:22 PM, Max Bridgewater max.bridgewa...@gmail.com
wrote:
I don't have my settings to try it right now. Still i have a
question. If it's the way you describe it, why wouldn't sofia/
extenal/f...@bar solve the problem? I think i even copied the
extension both to the default directory. But i will confirm and let
you know.
Max.
On Fri, Jun 19, 2009 at 4:55 PM, Michael Collins
m...@freeswitch.org wrote:
Now I feel stupid because I didn't read your original post closely
enough.
You've defined your mysocket extension in the public context but
when you do an origination with sofia/internal/f...@bar it will use
the default context. I think the quickest way to handle this is to
create a copy of your mysocket.xml file and put it in conf/dialplan/
default/ and be done with it.
-MC
On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater max.bridgewa...@gmail.com
wrote:
Mike,
Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very
XML to me though.
Max.
On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins
m...@freeswitch.org wrote:
Max,
that pastebin failed miserably as none of the xml shows up. can you
try again or use our pastebin.freeswitch.org site?
-MC
On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater max.bridgewa...@gmail.com
wrote:
Hi Mike,
It's pasted here: http://pastebin.ca/1466521
Thanks,
Max.
On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins
m...@freeswitch.org wrote:
Can you turn on debugging (F8) and capture all the output after your
originate? Put it into a pastebin. (pastebin.freeswitch.org)
-MC
On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater max.bridgewa...@gmail.com
wrote:
Any help our there?
I'm still trying to get this piece working. Essentially what i wan
to do is, when a call comes in (from registered devices as well as
unregistered devices), notify the my server socket. Somehow it's not
working. The change i made compared to the standard Freeswitch
settings are the following:
1) Added following extension that in /usr/local/freeswitch/conf/
dialplan/public/mysocket.xml:
include
extension name=mysocket
condition field=destination_number expression=^242.*$
action application=socket data=192.168.50.67:1
full /
/condition
/extension
/include
2) Changed file: /usr/local/freeswitch/conf/autoload_configs/
event_socket.conf to:
configuration name=event_socket.conf description=Socket Client
settings
param name=nat-map value=false/
param name=listen-ip value=0.0.0.0/
param name=listen-port value=8021/
param name=password value=1234/
!--param name=apply-inbound-acl value=lan/--
/settings
/configuration
I noticed that with this extension, all calls received from external
providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my
socket. But calls from registered devices and initiated using the
socket interface are not forwarded. Is there something that need to
be changed in the profiles?
or is something wrong with my dial string? {origination_caller_id_number=12000}sofia/internal/242...@192.168.1.62
.
In the logs, i cannot see that that my extension is being matched.
Any idea,
Max.
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Re: [Freeswitch-users] Help with Socket event again
Hi Mike,
Unfortunately this doesn't seem to solve my problem. Here is my extension
again:
extension name=mysocket
condition field=destination_number expression=^242.*
action application=socket data=192.168.50.67:1 full /
/condition
/extension
I've copied it now under:
/user/local/freeswitch/conf/dialplan/default
/user/local/freeswitch/conf/dialplan/public
The different dial strings i tried:
{origination_caller_id_number=120003}sofia/internal/242...@192.168.50.67
park()
{origination_caller_id_number=120003}sofia/internal/242...@192.168.50.67
{origination_caller_id_number=120003}sofia/internal/242424
{origination_caller_id_number=120003}sofia/internal/242424%192.168.50.62
My goal: have the call captured by the above extension and redirected to a
server socket running at 192.168.50.67:1.
Any thought?
Max.
On Fri, Jun 19, 2009 at 5:22 PM, Max Bridgewater
max.bridgewa...@gmail.comwrote:
I don't have my settings to try it right now. Still i have a question. If
it's the way you describe it, why wouldn't sofia/extenal/f...@bar solve the
problem? I think i even copied the extension both to the default directory.
But i will confirm and let you know.
Max.
On Fri, Jun 19, 2009 at 4:55 PM, Michael Collins m...@freeswitch.orgwrote:
Now I feel stupid because I didn't read your original post closely
enough.
You've defined your mysocket extension in the public context but when
you do an origination with sofia/internal/f...@bar it will use the
default context. I think the quickest way to handle this is to create a
copy of your mysocket.xml file and put it in conf/dialplan/default/ and be
done with it.
-MC
On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater
max.bridgewa...@gmail.com wrote:
Mike,
Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very XML
to me though.
Max.
On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins m...@freeswitch.orgwrote:
Max,
that pastebin failed miserably as none of the xml shows up. can you try
again or use our pastebin.freeswitch.org site?
-MC
On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater
max.bridgewa...@gmail.com wrote:
Hi Mike,
It's pasted here: http://pastebin.ca/1466521
Thanks,
Max.
On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins
m...@freeswitch.orgwrote:
Can you turn on debugging (F8) and capture all the output after your
originate? Put it into a pastebin. (pastebin.freeswitch.org)
-MC
On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater
max.bridgewa...@gmail.com wrote:
Any help our there?
I'm still trying to get this piece working. Essentially what i wan to
do is, when a call comes in (from registered devices as well as
unregistered
devices), notify the my server socket. Somehow it's not working. The
change
i made compared to the standard Freeswitch settings are the following:
1) Added following extension that in
/usr/local/freeswitch/conf/dialplan/public/mysocket.xml:
include
extension name=mysocket
condition field=destination_number expression=^242.*$
action application=socket data=192.168.50.67:1full
/
/condition
/extension
/include
2) Changed file:
/usr/local/freeswitch/conf/autoload_configs/event_socket.conf to:
configuration name=event_socket.conf description=Socket Client
settings
param name=nat-map value=false/
param name=listen-ip value=0.0.0.0/
param name=listen-port value=8021/
param name=password value=1234/
!--param name=apply-inbound-acl value=lan/--
/settings
/configuration
I noticed that with this extension, all calls received from external
providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my
socket.
But calls from registered devices and initiated using the socket
interface
are not forwarded. Is there something that need to be changed in the
profiles?
or is something wrong with my dial string?
{origination_caller_id_number=12000}sofia/internal/
242...@192.168.1.62.
In the logs, i cannot see that that my extension is being matched.
Any idea,
Max.
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Re: [Freeswitch-users] Help with Socket event again
Hmm thamks. I tried it and it doesn't work out of the box. Here are my logs:
http://pastebin.freeswitch.org/9454
Thanks,
Max.
On Mon, Jun 22, 2009 at 12:06 PM, Brian West br...@freeswitch.org wrote:
originate
{origination_caller_id_number=120003}sofia/internal/242...@192.168.50.67%7Borigination_caller_id_number=120003%7Dsofia/internal/242...@192.168.50.67socket(
192.168.50.67:1 full)
/b
On Jun 22, 2009, at 11:57 AM, Max Bridgewater wrote:
Hi Mike,
Unfortunately this doesn't seem to solve my problem. Here is my extension
again:
extension name=mysocket
condition field=destination_number expression=^242.*
action application=socket data=192.168.50.67:1 full
/
/condition
/extension
I've copied it now under:
/user/local/freeswitch/conf/dialplan/default
/user/local/freeswitch/conf/dialplan/public
The different dial strings i tried:
{origination_caller_id_number=120003}sofia/internal/242...@192.168.50.67
park()
{origination_caller_id_number=120003}sofia/internal/242...@192.168.50.67
{origination_caller_id_number=120003}sofia/internal/242424
{origination_caller_id_number=120003}sofia/internal/242424%192.168.50.62
My goal: have the call captured by the above extension and redirected to a
server socket running at 192.168.50.67:1.
Any thought?
Max.
On Fri, Jun 19, 2009 at 5:22 PM, Max Bridgewater
max.bridgewa...@gmail.com wrote:
I don't have my settings to try it right now. Still i have a question. If
it's the way you describe it, why wouldn't sofia/extenal/f...@bar solve
the problem? I think i even copied the extension both to the default
directory. But i will confirm and let you know.
Max.
On Fri, Jun 19, 2009 at 4:55 PM, Michael Collins m...@freeswitch.orgwrote:
Now I feel stupid because I didn't read your original post closely
enough.
You've defined your mysocket extension in the public context but when
you do an origination with sofia/internal/f...@bar it will use the
default context. I think the quickest way to handle this is to create a
copy of your mysocket.xml file and put it in conf/dialplan/default/ and be
done with it.
-MC
On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater
max.bridgewa...@gmail.com wrote:
Mike,
Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very XML
to me though.
Max.
On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins
m...@freeswitch.orgwrote:
Max,
that pastebin failed miserably as none of the xml shows up. can you try
again or use our pastebin.freeswitch.org site?
-MC
On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater
max.bridgewa...@gmail.com wrote:
Hi Mike,
It's pasted here: http://pastebin.ca/1466521
Thanks,
Max.
On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins m...@freeswitch.org
wrote:
Can you turn on debugging (F8) and capture all the output after your
originate? Put it into a pastebin. (pastebin.freeswitch.org)
-MC
On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater
max.bridgewa...@gmail.com wrote:
Any help our there?
I'm still trying to get this piece working. Essentially what i wan
to do is, when a call comes in (from registered devices as well as
unregistered devices), notify the my server socket. Somehow it's not
working. The change i made compared to the standard Freeswitch
settings are
the following:
1) Added following extension that in
/usr/local/freeswitch/conf/dialplan/public/mysocket.xml:
include
extension name=mysocket
condition field=destination_number expression=^242.*$
action application=socket
data=192.168.50.67:1full /
/condition
/extension
/include
2) Changed file:
/usr/local/freeswitch/conf/autoload_configs/event_socket.conf to:
configuration name=event_socket.conf description=Socket Client
settings
param name=nat-map value=false/
param name=listen-ip value=0.0.0.0/
param name=listen-port value=8021/
param name=password value=1234/
!--param name=apply-inbound-acl value=lan/--
/settings
/configuration
I noticed that with this extension, all calls received from external
providers (e.g. Les.net, Gafachi, etc.) are indeed forwarded to my
socket.
But calls from registered devices and initiated using the socket
interface
are not forwarded. Is there something that need to be changed in the
profiles?
or is something wrong with my dial string?
{origination_caller_id_number=12000}sofia/internal/
242...@192.168.1.62.
In the logs, i cannot see that that my extension is being matched.
Any idea,
Max.
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[Freeswitch-users] Limit length of call with mod_limit?
Hi there, Can mod_limit be used to restrict the length of a single call? I checked the wiki, dug into the code of mod_limit this weekend and couldn't find an answer. Lon ___ 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
what is 242424? If its a locally registered user you should be using a % instead of an @ /b On Jun 22, 2009, at 1:08 PM, Max Bridgewater wrote: Hmm thamks. I tried it and it doesn't work out of the box. Here are my logs: http://pastebin.freeswitch.org/9454 Thanks, Max. 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] Limit length of call with mod_limit?
http://wiki.freeswitch.org/wiki/Channel_Variables /b On Jun 22, 2009, at 1:18 PM, Lon Baker wrote: Hi there, Can mod_limit be used to restrict the length of a single call? I checked the wiki, dug into the code of mod_limit this weekend and couldn't find an answer. Lon ___ 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 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] rpm build issues on CentOS 5 with trunk
I have recently updated the RPM spec for FreeSWITCH to use the latest
SVN release, including some unspecified files for the newest mods (with
nibblebil, unimrcp, etc, just like your build output is bitching about).
Follow these steps to build it:
1 - Get the latest SVN release and make a tar-ball for it:
$ cd /usr/src/redhat/SOURCES/
$ svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk
freeswitch-1.0.4trunk
$ tar -cjvf freeswitch-1.0.4trunk.tar.bz2 freeswitch-1.0.4trunk/
2 - Grab the libraries required to build FS:
$ wget -q -O - http://www.etellicom.com/~raul/freeswitch_deps.txt | bash
3 - Download the RPM spec:
$ cd /usr/src/redhat/SPECS/
$ wget http://www.etellicom.com/~raul/freeswitch.spec
4 - Build it :)
$ rpmbuild -ba freeswitch.spec
It works fine with a standard CentOS 5.3 installation.
Regards,
Raul
On Mon, 2009-06-22 at 01:25 +, Joseph L. Casale wrote:
I attempted to build rpm's from the included spec file using a non-root user
build environment. The steps I used are as follows:
1. Check build deps @
http://wiki.freeswitch.org/wiki/Installation_Guide#RHEL.2FCentOS
2. Pulled a copy of trunk in the SOURCES directory tar/bzip2 it as expected
by the spec:
svn co http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch-1.0.4
tar cjf freeswitch-1.0.4.tar.bz2 freeswitch-1.0.4/
3. Copy spec to SPECS directory:
4. Pull in Source (Source0 doesn't exist yet, I make it above):
for SOURCE in $(awk '/^Source[1-9][0-9]*:/ {print $2}'
freeswitch.spec); do wget -P ../../SOURCES/ $SOURCE; done
5. Check spec in svn copy for deps:
yum install $(awk -v ORS= '/^BuildRequires:/ {print $2}'
freeswitch.spec)
6.Build rpm:
rpmbuild -ba freeswitch.spec
After some time, near the end I see various issues like the following:
making install mod_speex
installing mod_speex.so
quiet_libtool: install: warning:
`/home/builder/rpmbuild/BUILD/freeswitch-1.0.4/libfreeswitch.la'
has not been installed in `/opt/freeswitch/lib'
It also seems to download everything it would normally again, then fails with
several errors like the following:
RPM build errors:
File not found by glob:
/var/tmp/freeswitch-1.0.4-1-root-builder/opt/freeswitch/mod/ozmod_wanpipe.so*
Installed (but unpackaged) file(s) found:
/opt/freeswitch/conf/autoload_configs/cidlookup.conf.xml
/opt/freeswitch/conf/autoload_configs/nibblebill.conf.xml
/opt/freeswitch/conf/autoload_configs/unimrcp.conf.xml
/opt/freeswitch/conf/lang/ru/demo/demo-ivr.xml
.
.
.
After which no rpm's are built? Anyone know what tricks are still needed with
the spec from svn?
Thanks!
jlc
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Re: [Freeswitch-users] rpm build issues on CentOS 5 with trunk
Scroll up and look for the real error. All I see are these: *** Warning: Linking the shared library libfreeswitch.la against the *** static library libs/libedit/src/.libs/libedit.a is not portable! ** Integer sample type enabled ** *** The gtk-config script installed by GTK could not be found *** If GTK was installed in PREFIX, make sure PREFIX/bin is in *** your path, or set the GTK_CONFIG environment variable to the *** full path to gtk-config. I am pulling down the newer libraries and updated spec now to try that. Thanks guys! jlc ___ 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
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 ___ 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] rpm build issues on CentOS 5 with trunk
Ok, thats not the issue... look lower or post the full log. /b On Jun 22, 2009, at 2:50 PM, Joseph L. Casale wrote: All I see are these: *** Warning: Linking the shared library libfreeswitch.la against the *** static library libs/libedit/src/.libs/libedit.a is not portable! ** Integer sample type enabled ** *** The gtk-config script installed by GTK could not be found *** If GTK was installed in PREFIX, make sure PREFIX/bin is in *** your path, or set the GTK_CONFIG environment variable to the *** full path to gtk-config. I am pulling down the newer libraries and updated spec now to try that. Thanks guys! jlc 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] Transmit fax locally for test
Just wanted to follow-up again. Is this the proper or best way to configure
this? See below...
Tim
From: timb0...@hotmail.com
To: freeswitch-users@lists.freeswitch.org
Subject: (Found Fix) Transmit fax locally for test
Date: Fri, 19 Jun 2009 23:06:10 -0400
Ok so after many attempts of trial and error I narrowed it down to acls. So
when trying to orginate a call to the local FS extension it was getting blocked.
Adding the following allow with my freeswitch IP to the domains list allowed
the originate to take place.
acl.conf.xml:
list name=domains default=deny
node type=allow cidr=192.168.10.35/32/
node type=allow domain=$${domain}/
/list
So now this statement works for local fax testing:
originate sofia/default/8...@192.168.10.35 txfax(storage/fax/test.tif)
Now my question is, is this the proper or best way to configure this?
Tim
--
Message: 1
Date: Fri, 19 Jun 2009 10:00:35 -0500
From: Michael Collins m...@freeswitch.org
Subject: [Freeswitch-users] Update - Transmit fax locally for test
To: freeswitch-users@lists.freeswitch.org
Message-ID:
87f2f3b90906190800u5d9436cbu2bd594bc8d09...@mail.gmail.com
Content-Type: text/plain; charset=windows-1252
Tim,
Look at lines 47 and 48 of the pastebin. I think something goofy is
happening there. What is 8...@x.x.x.x in your system? Is that the receive
fax extension?
-MC
-- Forwarded message --
From: Tim B timb0...@hotmail.com
Date: Fri, Jun 19, 2009 at 7:39 AM
Subject: Re: [Freeswitch-users] Freeswitch-users Digest, Vol 36, Issue 188
To: freeswitch-users@lists.freeswitch.org
here is the log...
http://pastebin.freeswitch.org/9440
haha, yeah i see it now... duh. pulled an all nighter, too many things
going on. must have overlooked it.
When I connect to pastebin.freeswitch.org I get a helpful notice saying
the login and password is pastebin/freeswitch
been trying to break myself into freeswitch on top of my original workload.
thanks for the help.
Bing™ brings you maps, menus, and reviews organized in one place. Try it now.
_
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Re: [Freeswitch-users] Help with Socket event again
On Mon, Jun 22, 2009 at 1:01 PM, Max Bridgewater
max.bridgewa...@gmail.comwrote:
It's nothing. There is no extension like that. Shouldn't this nonetheless
be caught by a regex such as the following?
field=destination_number expression=^242.*
The issue i have here is that it seems that the extensions aren't even
processed. Usually, the log would show the list of processed extensions,
each prefixed with the result PASS, FAIL.
Max, if your originate line already has the sofia dialstring then there's
really no reason to send the call through the dialplan - it already knows
where to go. If you want to force the call through the dialplan then use
loopback. However, you need some sort of endpoint for that to work. In your
example you have this originate line:
originate {origination_caller_id_number=120003}sofia/internal/
242...@192.168.50.67 socket(192.168.50.67:1 full)
Is 242...@192.168.50.67 a locally registered user? If so you could just do
this:
originate {origination_caller_id_number=120003} loopback/242424 socket(
192.168.50.67:1 full)
This would run the A leg through the dialplan to look for destination number
242424 and then handle appropriately.
If I understand your scenario I believe you are trying to get one leg of the
call established and then the other leg handled by the event socket. What is
the endpoint you want handled? A SIP phone that is registered locally? Or
something else? In any case, you can CAN loop it through the dialplan but
you aren't forced to do so. Assuming 1000 is locally registered:
originate {origination_caller_id_number=120003}
sofia/internal/1000%192.168.50.67 socket(192.168.50.67:1 full)
originate {origination_caller_id_number=120003} user/1000 socket(
192.168.50.67:1 full)
originate {origination_caller_id_number=120003} loopback/1000 socket(
192.168.50.67:1 full)
NOTE: the first two do not use the dialplan but the third example does. This
means you MUST handle destination_number=1000 in your dialplan (which the
default config does).
Hope this helps.
-MC
Max.
On Mon, Jun 22, 2009 at 1:18 PM, Brian West br...@freeswitch.org wrote:
what is 242424? If its a locally registered user you should be using a %
instead of an @
/b
On Jun 22, 2009, at 1:08 PM, Max Bridgewater wrote:
Hmm thamks. I tried it and it doesn't work out of the box. Here are my
logs: http://pastebin.freeswitch.org/9454
Thanks,
Max.
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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[Freeswitch-users] Polycom configuration problems?
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
2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing
1001-323 in context default
Dialplan: sofia/internal/1...@192.168.10.29 parsing [default-unloop]
continue=false
Dialplan: sofia/internal/1...@192.168.10.29 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1...@192.168.10.29 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
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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
2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing
1001-323 in context default
Dialplan: sofia/internal/1...@192.168.10.29 parsing [default-unloop]
continue=false
Dialplan: sofia/internal/1...@192.168.10.29 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1...@192.168.10.29 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
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Re: [Freeswitch-users] Transmit fax locally for test
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 _ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd_062009___ 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] rpm build issues on CentOS 5 with trunk
I have recently updated the RPM spec for FreeSWITCH to use the latest SVN release, including some unspecified files for the newest mods (with nibblebil, unimrcp, etc, just like your build output is bitching about). Follow these steps to build it: Raul, Appreciate this, it worked. I am re-running it as the first time I was only logging stdout and it appears there are some errors in my build worth exploring. Nothing prevented the build, so I created a local repo with all the rpm's and executed a `yum install freeswitch` and I see that it never pulled in anything else. Where abouts in the docs could I find info on knowing what I need for an initial install to test? Thanks for the help! jlc ___ 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] Limit length of call with mod_limit?
This line in your diaplan will set a timer to hangup the calls x secs after answer. action application=set data=execute_on_answer=sched_hangup +time in secs ALLOTED_TIMEOUT/ On Tue, Jun 23, 2009 at 3:18 AM, Lon Baker l...@kickasspixels.com wrote: Hi there, Can mod_limit be used to restrict the length of a single call? I checked the wiki, dug into the code of mod_limit this weekend and couldn't find an answer. Lon ___ 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 -- Jim Burke Director Evolutiontel. http://www.evolutiontel.net ___ 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] How to originate gtalk calls
Hi Guys, I've configured a gtalk client based on the steps in this url: http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/. But i'm not sure how to originate calls to different gtalk users dynamically. I've tried this: freeswitch *originate dingaling/gmail.com/user...@gmail.com echo* but got CHAN_NOT_IMPLEMENTED error. *2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot create outgoing channel of type [dingaling] cause: [CHAN_NOT_IMPLEMENTED]* Please kindly let me know what the correct originate string is. 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] video playback on FS
Dear All, on the default.xml in dialplan directory of FS, content video extension dialplan with file extension fsv, 562 extension name=video_record 563 condition field=destination_number expression=^9993$ 564 action application=answer/ 565 action application=record_fsv data=/tmp/testrecord.fsv/ 566 /condition 567 /extension 568 569 extension name=video_playback 570 condition field=destination_number expression=^9994$ 571 action application=answer/ 572 action application=play_fsv data=/tmp/testrecord.fsv/ 573 /condition 574 /extension what is the fsv video format from? as we know flv for flash video, how to convert from mp4 or avi to fsv file extension? thank you in advanced, best regard, mashudi Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? Ikuti Dahsyatnya FLEXI KOMUNITAS. Ketik CREATEspasi[NAMA GRUP], sms ke 345. Contoh: CREATE SMU2, sms ke 345. Informasi selanjutnya: - hubungi 147 - http://www.telkomflexi.com - ketik INFO, sms ke 345. ___ 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 originate gtalk calls
Might need to compile and load mod_dingaling first. /b On Jun 22, 2009, at 9:33 PM, Jingwei Yang wrote: Hi Guys, I've configured a gtalk client based on the steps in this url: http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/ . But i'm not sure how to originate calls to different gtalk users dynamically. I've tried this: freeswitch originate dingaling/gmail.com/user...@gmail.com echo but got CHAN_NOT_IMPLEMENTED error. 2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot create outgoing channel of type [dingaling] cause: [CHAN_NOT_IMPLEMENTED] Please kindly let me know what the correct originate string is. 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 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] How to originate gtalk calls
try load mod_dingaling. If that does not work, get to the source dir, edit modules.conf, uncomment mod_dingaling, make make install Dont forget to load the mod once FS is up again.. jmesquita On Mon, Jun 22, 2009 at 11:33 PM, Jingwei Yang jingwei.y...@gmail.comwrote: Hi Guys, I've configured a gtalk client based on the steps in this url: http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/. But i'm not sure how to originate calls to different gtalk users dynamically. I've tried this: freeswitch *originate dingaling/gmail.com/user...@gmail.com echo* but got CHAN_NOT_IMPLEMENTED error. *2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot create outgoing channel of type [dingaling] cause: [CHAN_NOT_IMPLEMENTED]* Please kindly let me know what the correct originate string is. 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] How to originate gtalk calls
Hi Brian and João, you're right, I forgot to load mod_dingaling. Thanks for the help. 2009/6/23 João Mesquita jmesqu...@gmail.com try load mod_dingaling. If that does not work, get to the source dir, edit modules.conf, uncomment mod_dingaling, make make install Dont forget to load the mod once FS is up again.. jmesquita On Mon, Jun 22, 2009 at 11:33 PM, Jingwei Yang jingwei.y...@gmail.comwrote: Hi Guys, I've configured a gtalk client based on the steps in this url: http://chesterton.id.au/blog/2007/12/31/freeswitch-and-google-talk/. But i'm not sure how to originate calls to different gtalk users dynamically. I've tried this: freeswitch *originate dingaling/gmail.com/user...@gmail.com echo* but got CHAN_NOT_IMPLEMENTED error. *2009-06-23 10:11:01.25414 [ERR] switch_ivr_originate.c:1495 Cannot create outgoing channel of type [dingaling] cause: [CHAN_NOT_IMPLEMENTED] * Please kindly let me know what the correct originate string is. 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 ___ 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] video playback on FS
On Jun 22, 2009, at 10:28 PM, mashudi wrote: Dear All, on the default.xml in dialplan directory of FS, content video extension dialplan with file extension fsv, 562 extension name=video_record 563 condition field=destination_number expression=^9993$ 564 action application=answer/ 565 action application=record_fsv data=/tmp/testrecord.fsv/ 566 /condition 567 /extension 568 569 extension name=video_playback 570 condition field=destination_number expression=^9994$ 571 action application=answer/ 572 action application=play_fsv data=/tmp/ testrecord.fsv/ 573 /condition 574 /extension what is the fsv video format from? as we know flv for flash video, how to convert from mp4 or avi to fsv file extension? We just save the raw RTP and stream it back out... btw don't hijack threads please. thank you in advanced, best regard, mashudi 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
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] 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 -- View this message in context: http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-tp24143545p24158819.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] How to change outbound_caller_name=FreeSWITCH and number=000000000
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
Re: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000
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
