[Freeswitch-users] How to delay IVR answer during an outbound call
I have a dummy question. Say, you have an outbound call to the demo IVR as below: originate sofia/gateway/myvoip/19876543210 5000 How do I delay the IVR response until the recipient at 19876543210 picks up the call? I tried ignore_early_media=true, which had no effect. Many thanks in advance. ___ 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] files.freeswitch.org resets connection.
I would have to say its YOUR system and not ours. /b On Aug 11, 2009, at 11:56 PM, Diego Viola wrote: Resolving files.freeswitch.org... failed: Temporary failure in name resolution. Again... On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola diego.vi...@gmail.com wrote: Nope. ___ 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 delay IVR answer during an outbound call
Is your provider answering the call before its connected? If so then they should be shot. I can't imagine other way the call would be answered unless you're using ignore_early_media wrong can you show me who you're doing this? /b On Aug 12, 2009, at 12:58 AM, Paul Li wrote: I have a dummy question. Say, you have an outbound call to the demo IVR as below: originate sofia/gateway/myvoip/19876543210 5000 How do I delay the IVR response until the recipient at 19876543210 picks up the call? I tried ignore_early_media=true, which had no effect. Many thanks in advance. ___ 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 delay IVR answer during an outbound call
I am actually doing a lua script for IVR as follows -- answer the call session:answer(); while session:ready() == true do -- sleep a second session:sleep(1000); -- play a file session:streamFile(/path/to/blah.wav); -- hangup session:hangup(); end The problem lies in: when I picked up my phone, blah.wav was already played for a while, instead of from the beginning. I shall greatly appreciate any input. On Wed, Aug 12, 2009 at 12:58 AM, Paul Liplite2...@gmail.com wrote: I have a dummy question. Say, you have an outbound call to the demo IVR as below: originate sofia/gateway/myvoip/19876543210 5000 How do I delay the IVR response until the recipient at 19876543210 picks up the call? I tried ignore_early_media=true, which had no effect. Many thanks in advance. ___ 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 when adding more extension
Hi,
I wanted to add more extension to freeswitch.
to add extension 1050 with password 1234 I did the following:
$ cd /usr/local/freeswitch/conf/directory/default
created 1050.xml having all '1000' strings replaced by '1050' by typing
$ sed s/1000/1050/g 1000.xml 1050.xml
rescan and reload the xml by typing into the CLI
freeswi...@internal sofia profile internal rescan reloadxml
However, when I tried to login with these credentials I got the
following in the fs_cli:
2009-08-12 06:35:01 [DEBUG] sofia_reg.c:1469 sofia_reg_parse_auth()
SIP username 1050 does not match auth username
2009-08-12 06:35:01 [DEBUG] sofia_reg.c:869
sofia_reg_handle_register() Send challenge for
[1...@server_address.net]
below are the content of 1000 and 1050 xml files
please advise.
$ cat 1050.xml
include
user id=1050 mailbox=1050
params
param name=password value=1234/
param name=vm-password value=1050/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=1050/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 1050/
variable name=effective_caller_id_number value=1050/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
/variables
/user
/include
$ cat 1000.xml
include
user id=1000 mailbox=1000
params
param name=password value=1234/
param name=vm-password value=1000/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=1000/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 1000/
variable name=effective_caller_id_number value=1000/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
/variables
/user
/include
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Re: [Freeswitch-users] files.freeswitch.org resets connection.
Aww, ok. Bad luck to me :). On Wed, Aug 12, 2009 at 1:57 AM, Brian West br...@freeswitch.org wrote: I would have to say its YOUR system and not ours. /b On Aug 11, 2009, at 11:56 PM, Diego Viola wrote: Resolving files.freeswitch.org... failed: Temporary failure in name resolution. Again... On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola diego.vi...@gmail.com wrote: Nope. ___ 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] Spanish Prompts
I would say there are no changes in gender for dialects...but with so many languages around I can't assure it 100% ;) Samuel. 2009/8/11 Michael Collins m...@freeswitch.org 2009/8/11 João Mesquita jmesqu...@gmail.com Mike, the gender thing will eventually have to change code, I guess. I have not yet looked at the say code, so I am just imagining here. Are there gender differences between dialects of the same language? -MC ___ 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] hangup_after_bridge=false only works if bypass_media=false
Please post a bug for this on jira.freeswitch.org.
Mike
On Aug 11, 2009, at 2:29 PM, Jeremiah Johnson wrote:
This is an integral part of my application. I need to have
FreeSWITCH outside of the media path as well as be able to do
multiple bridges for the same A leg.
/*WORKS*/
action application=set data=hangup_after_bridge=false/
action application=bridge data=sofia/gateway/${mygateway}/1$
{client_one}/
action application=bridge data=sofia/gateway/${mygateway}/1$
{client_two}/
/*DOES NOT WORK*/
action application=set data=hangup_after_bridge=false/
action application=set data=bypass_media=true/
action application=bridge data=sofia/gateway/${mygateway}/1$
{client_one}/
action application=bridge data=sofia/gateway/${mygateway}/1$
{client_two}/
In the DOES NOT WORK example, the A leg hangs up as soon as the
leg for client_one hangs up.
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[Freeswitch-users] FS 1.0.4 official core dumping LUA streamFile
Greetings, I have the following LUA script (at end of email) in a fresh FS 1.0.4 install. I originally did an upgrade from one of the 1.0.4preX versions but when I came across this issue I went fresh just to make sure there wasn't an incompatibility with my previous config. What I'm seeing is a seg fault and a core dump after playing a sound file. I originally had a file I recorded but when I ran into this issue I figured I'd try an included sound file but that doesn't seem to make a bit of difference. 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Starting test.lua 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Caller [XX] connected 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Pre streamFile Segmentation fault (core dumped) Any ideas? Thanks, Charlie freeswitch.consoleLog(INFO, string.format(Starting test.lua\n)) session:answer(); session:setHangupHook(session_hangup_hook) calleridnumber = session:getVariable(caller_id_number) calleridname = session:getVariable(caller_id_name) if session:ready() then freeswitch.consoleLog(INFO, string.format(Caller [ .. calleridnumber .. ] connected\n)) freeswitch.consoleLog(INFO, string.format(Pre streamFile\n)) session:streamFile(conference/8000/conf-welcome.wav) freeswitch.consoleLog(INFO, string.format(Post streamFile.\n)) end function session_hangup_hook(status) freeswitch.consoleLog(INFO, Session hangup: \n) --[[ .. status .. \n) ]]-- error() end session:hangup() ___ 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 transcription
Not sure, but they do certainly have a reasonably large server farm for doing processing :) I note that sphinx4 I believe has a java example for doing dictation transcription from an audio file (saw something on a sphinx forum or mailing list while trawling the net). I'm still investigating modifications to use pocketsphinx. Regards Kirk 2009/8/11 João Mesquita jmesqu...@gmail.com I am sorry for the ignorance on the matter, but how does google voice does? Do they also have humans? jmesquita On Tue, Aug 11, 2009 at 11:17 AM, Kirk Bateman kirk.bate...@gmail.comwrote: I'm still interested in getting pocketsphinx to attempt speech recognition on an audio file. To be honest, most of the problem is that at 8Khz (mobile phone call rate), speech detection is NOT very accurate, at 16Khz it IS significantly better. I'm planning to have a play with the speechtools module and mod_pocketsphinx etc to try and get an audio file parsed, spare time permitting. Will let the list know if I get anywhere. Regards Kirk Bateman 2009/8/11 David Knell d...@3c.co.uk Hi Pete, I'm afraid that the answer's still the same: use a human. Here's an article describing the state of the art: http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/ - the links to previous stories at the bottom provide good background. --Dave I apologize, I should have been more clear. We will be using humans to scan the translated results. But we are looking for a system to perform the first pass on the audio to hopefully help the human type less. Although the question has been raised if it's faster to have a human just transcribe the whole thing, or fix up what the computer spit out. If you have any insights on this, that would be great. -pete Original Message Subject: Re: [Freeswitch-users] VoiceMail transcription From: David Knell d...@3c.co.uk Date: Mon, August 10, 2009 11:51 am To: freeswitch-users@lists.freeswitch.org Good evening Pete, The only way to do this is, I'm afraid, to use a human. We use Amazon's Mechanical Turk to good effect. Cheers -- Dave Good morning all, I realize this is slightly off the FS topic, but I am wondering if anyone out there has experience with software packages designed for the transcription of voicemails to text. I've used pocketsphinx with FS to handle IVR menus, but now have the task of figuring out how to convert recorded phone conversations (voicemails mostly) to text. This does not have to be a real-time process, I can store the audio files and process them over time. This would need to be a software (preferable open source) solution. ASPs like VoiceCloud would not work for this application. Thanks for any help -pete ___ 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 -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ 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 -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ 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
[Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 v1.0.4)
Hi All, I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and 1.0.4. I am running on Solaris 10 Update 5 on x86 hardware (32-bit). The build fails with: --- snip --- make: Fatal error: Command failed for target `all-recursive' Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4 *** Error code 1 make: Fatal error: Command failed for target `all' --- Looking back through the build I can see the following error: --- snip --- creating libfreeswitch.la (cd .libs rm -f libfreeswitch.la ln -s ../libfreeswitch.la libfreeswitch.la) /usr/bin/cc -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes -DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch freeswitch-switch.o ./.libs/libfreeswitch.so -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib/.libs/libexpat.a /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libapr-1.a -lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp -L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl -lcurses -lsocket -R/opt/freeswitch/lib -R/usr/sfw/lib Undefined first referenced symbol in file herror ./.libs/libfreeswitch.so ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch *** Error code 1 The following command caused the error: `if test -z ; then echo /bin/bash /export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` --tag=CC --mode=link /usr/bin/cc -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes -DPATH_MAX=2048 -g -v -Xc -xc99=all -lm -R/opt/freeswitch/lib -o freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib freeswitch-switch.o libfreeswitch.la libs/apr/libapr-1.la libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl -lcrypto -lsocket -lnsl -ldl -lcurses -lsocket --- snip --- Then a little above this error, there is the following warning that is displayed (I'm not sure if it is related): --- snip --- *** Warning: Linking the shared library libfreeswitch.la against the *** static library libs/libedit/src/.libs/libedit.a is not portable! --- snip --- My configure line is as follows: --- ./configure --prefix=/opt/freeswitch --- I have the complete configure and make output if anyone needs them. Any help/pointers would be greatly appreciated. Thanks Bruce ___ 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] problem when adding more extension
Hello,
I've just had the same problem. Solved it by adding the new extension to
the default group.
i.e. In the /usr/local/freeswitch/conf/directory/default.xml file you
need to add user id=1050 type=pointer with one of the group
blocks (e.g. after the line group name=default
Kevin
Tzury Bar Yochay wrote:
Hi,
I wanted to add more extension to freeswitch.
to add extension 1050 with password 1234 I did the following:
$ cd /usr/local/freeswitch/conf/directory/default
created 1050.xml having all '1000' strings replaced by '1050' by typing
$ sed s/1000/1050/g 1000.xml 1050.xml
rescan and reload the xml by typing into the CLI
freeswi...@internal sofia profile internal rescan reloadxml
However, when I tried to login with these credentials I got the
following in the fs_cli:
2009-08-12 06:35:01 [DEBUG] sofia_reg.c:1469 sofia_reg_parse_auth()
SIP username 1050 does not match auth username
2009-08-12 06:35:01 [DEBUG] sofia_reg.c:869
sofia_reg_handle_register() Send challenge for
[1...@server_address.net]
below are the content of 1000 and 1050 xml files
please advise.
$ cat 1050.xml
include
user id=1050 mailbox=1050
params
param name=password value=1234/
param name=vm-password value=1050/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=1050/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 1050/
variable name=effective_caller_id_number value=1050/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
/variables
/user
/include
$ cat 1000.xml
include
user id=1000 mailbox=1000
params
param name=password value=1234/
param name=vm-password value=1000/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=1000/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 1000/
variable name=effective_caller_id_number value=1000/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
/variables
/user
/include
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Re: [Freeswitch-users] answer command
I've tried to use answer command from outbound event socket and it's working, but the problem is that FS answering the call, but SIP Client (we tried this with EyeBeam and CISCO 7960) doesn't know that call was answered. So, as long as FS doesn't know what to do with this number it then disconnects the call. 2009-08-12 11:25:07.599250 [NOTICE] mod_dptools.c:649 Channel [sofia/internal/sip:1...@10.107.181.160:42840] has been answered 2009-08-12 11:25:07.599250 [NOTICE] switch_ivr_originate.c:2015 Channel [sofia/internal/1...@10.107.249.12] has been answered 2009-08-12 11:25:07.614875 [ERR] switch_core_io.c:118 sofia/internal/sip:1...@10.107.181.160:42840 has no read codec. 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:503 Hangup sofia/internal/sip:1...@10.107.181.160:42840 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:1016 Hangup sofia/internal/1...@10.107.249.12 [CS_EXECUTE] [NORMAL_CLEARING] 2009-08-12 11:25:07.630500 [NOTICE] switch_core_session.c:1086 Session 133 (sofia/internal/sip:1...@10.107.181.160:42840) Ended Maybe there is the way to acknowledge SIP client that call was answered? Regards, Maxim Tsvetov Diego Viola wrote: I suggest that you learn the differences between mod_commands commands and mod_dptools applications, and also the interfaces where you can access and use them. As said before, mod_dptools is accessible from dialplan, event socket outbound, etc. and mod_commands is accessible from the CLI, event socket (inbound/outbound), XML RPC, etc. That's all described in the wiki I think. Let us know if you have any questions =D. On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola diego.vi...@gmail.com wrote: Michael, you're welcome :). Milena, answer is a mod_dptools command, you can use it from the XML dialplan or from the event socket outbound. mod_commands API are APIs that you execute from the socket, event socket inbound, etc. But you can also execute them from event socket outbound using the api command. I hope that makes sense, correct me if I'm wrong =D. On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins m...@freeswitch.orgwrote: On Tue, Aug 11, 2009 at 9:05 AM, Milena testeado...@gmail.com wrote: Hello Brian, I wanna fix the wiki, but to make sure i got it right, does it only work on outbound event socket? or is there any other scenario where it would work. FYI, Diego Viola fixed the wiki. (Thanks Diego!) -MC ___ 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/answer-command-tp24912812p24931876.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] FS 1.0.4 official core dumping LUA streamFile
If your seeing a segfault, please report it to jira.freeswitch.org with a backtrace and details of how to reproduce. Mike On Aug 12, 2009, at 2:37 AM, Charles Boening wrote: Greetings, I have the following LUA script (at end of email) in a fresh FS 1.0.4 install. I originally did an upgrade from one of the 1.0.4preX versions but when I came across this issue I went fresh just to make sure there wasn’t an incompatibility with my previous config. What I’m seeing is a seg fault and a core dump after playing a sound file. I originally had a file I recorded but when I ran into this issue I figured I’d try an included sound file but that doesn’t seem to make a bit of difference. 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Starting test.lua 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Caller [XX] connected 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Pre streamFile Segmentation fault (core dumped) Any ideas? Thanks, Charlie freeswitch.consoleLog(INFO, string.format(Starting test.lua\n)) session:answer(); session:setHangupHook(session_hangup_hook) calleridnumber = session:getVariable(caller_id_number) calleridname = session:getVariable(caller_id_name) if session:ready() then freeswitch.consoleLog(INFO, string.format(Caller [ .. calleridnumber .. ] connected\n)) freeswitch.consoleLog(INFO, string.format(Pre streamFile\n)) session:streamFile(conference/8000/conf-welcome.wav) freeswitch.consoleLog(INFO, string.format(Post streamFile.\n)) end function session_hangup_hook(status) freeswitch.consoleLog(INFO, Session hangup: \n) --[[ .. status .. \n) ]]-- error() end session:hangup() ___ 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] problem when adding more extension
still not working, I mean, I can initiate a call from 1060 to 1000 but
not from 1000 to 1060.
1060 is just an example. This applies to all new extension I have
added (beyond to the default 1000-1019).
as you can see below I added them all to group name=support
This is how the confs look like
/usr/local/freeswitch/conf/directory# cat default.xml
include
!--the domain or ip (the right hand side of the @ in the addr--
domain name=$${domain}
params
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}/
/params
variables
variable name=record_stereo value=true/
variable name=default_gateway value=$${default_provider}/
variable name=default_areacode value=$${default_areacode}/
variable name=transfer_fallback_extension value=operator/
/variables
groups
group name=default
users
X-PRE-PROCESS cmd=include data=default/*.xml/
/users
/group
group name=sales
users
!--
type=pointer is a pointer so you can have the
same user in multiple groups. It basically means
to keep searching for the user in the directory.
--
user id=1000 type=pointer/
user id=1001 type=pointer/
user id=1002 type=pointer/
user id=1003 type=pointer/
user id=1004 type=pointer/
/users
/group
group name=billing
users
user id=1005 type=pointer/
user id=1006 type=pointer/
user id=1007 type=pointer/
user id=1008 type=pointer/
user id=1009 type=pointer/
/users
/group
group name=support
users
user id=1010 type=pointer/
user id=1011 type=pointer/
user id=1012 type=pointer/
user id=1013 type=pointer/
user id=1014 type=pointer/
user id=1015 type=pointer/
user id=1016 type=pointer/
user id=1017 type=pointer/
user id=1018 type=pointer/
user id=1019 type=pointer/
user id=1020 type=pointer/
user id=1050 type=pointer/
user id=1051 type=pointer/
user id=1052 type=pointer/
user id=1053 type=pointer/
user id=1054 type=pointer/
user id=1055 type=pointer/
user id=1056 type=pointer/
user id=1057 type=pointer/
user id=1058 type=pointer/
user id=1059 type=pointer/
user id=1060 type=pointer/
user id=1061 type=pointer/
user id=1062 type=pointer/
user id=1063 type=pointer/
/users
/group
/groups
/domain
/include
and the xml files under directory/default
r...@snoip-srv-001:/usr/local/freeswitch/conf/directory/default# ls -l
total 164
-rw-r--r-- 1 root root 750 2009-07-21 19:44 1000.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1001.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1002.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1003.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1004.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1005.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1006.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1007.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1008.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1009.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1010.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1011.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1012.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1013.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1014.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1015.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1016.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1017.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1018.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1019.xml
-rw-r--r-- 1 root root 750 2009-08-12 06:49 1020.xml
-rw-r--r-- 1 root root 750 2009-07-27 13:14 1050.xml
-rw-r--r-- 1 root root 750 2009-07-27 13:14 1051.xml
-rw-r--r-- 1 root root 750 2009-07-27 13:14 1052.xml
-rw-r--r-- 1 root root 750 2009-08-06 08:59 1053.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:56 1054.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:56 1055.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:56 1056.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:56 1057.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:57 1058.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:57 1059.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:57 1060.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:57 1061.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:57 1062.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:58 1063.xml
-rw-r--r-- 1 root root 750 2009-08-11 10:10 1064.xml
-rw-r--r-- 1 root root 750 2009-08-11 10:12 1065.xml
-rw-r--r-- 1 root root 5029 2009-07-20 23:47 brian.xml
-rw-r--r-- 1 root root 526 2009-07-20 23:47 default.xml
-rw-r--r-- 1
Re: [Freeswitch-users] problem when adding more extension
Edit the line below as shown (in the dialplan/default.xml file)
Original line (about line 206)
condition field=desination_number expression=^(10[01][0-9])$
Replacement line
condition field=desination_number expression=^(10[0-9][0-9])$
This will allow extensions number 1000 to 1099.
Kevin
Tzury Bar Yochay wrote:
still not working, I mean, I can initiate a call from 1060 to 1000 but
not from 1000 to 1060.
1060 is just an example. This applies to all new extension I have
added (beyond to the default 1000-1019).
as you can see below I added them all to group name=support
This is how the confs look like
/usr/local/freeswitch/conf/directory# cat default.xml
include
!--the domain or ip (the right hand side of the @ in the addr--
domain name=$${domain}
params
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}/
/params
variables
variable name=record_stereo value=true/
variable name=default_gateway value=$${default_provider}/
variable name=default_areacode value=$${default_areacode}/
variable name=transfer_fallback_extension value=operator/
/variables
groups
group name=default
users
X-PRE-PROCESS cmd=include data=default/*.xml/
/users
/group
group name=sales
users
!--
type=pointer is a pointer so you can have the
same user in multiple groups. It basically means
to keep searching for the user in the directory.
--
user id=1000 type=pointer/
user id=1001 type=pointer/
user id=1002 type=pointer/
user id=1003 type=pointer/
user id=1004 type=pointer/
/users
/group
group name=billing
users
user id=1005 type=pointer/
user id=1006 type=pointer/
user id=1007 type=pointer/
user id=1008 type=pointer/
user id=1009 type=pointer/
/users
/group
group name=support
users
user id=1010 type=pointer/
user id=1011 type=pointer/
user id=1012 type=pointer/
user id=1013 type=pointer/
user id=1014 type=pointer/
user id=1015 type=pointer/
user id=1016 type=pointer/
user id=1017 type=pointer/
user id=1018 type=pointer/
user id=1019 type=pointer/
user id=1020 type=pointer/
user id=1050 type=pointer/
user id=1051 type=pointer/
user id=1052 type=pointer/
user id=1053 type=pointer/
user id=1054 type=pointer/
user id=1055 type=pointer/
user id=1056 type=pointer/
user id=1057 type=pointer/
user id=1058 type=pointer/
user id=1059 type=pointer/
user id=1060 type=pointer/
user id=1061 type=pointer/
user id=1062 type=pointer/
user id=1063 type=pointer/
/users
/group
/groups
/domain
/include
and the xml files under directory/default
r...@snoip-srv-001:/usr/local/freeswitch/conf/directory/default# ls -l
total 164
-rw-r--r-- 1 root root 750 2009-07-21 19:44 1000.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1001.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1002.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1003.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1004.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1005.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1006.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1007.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1008.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1009.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1010.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1011.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1012.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1013.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1014.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1015.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1016.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1017.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1018.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1019.xml
-rw-r--r-- 1 root root 750 2009-08-12 06:49 1020.xml
-rw-r--r-- 1 root root 750 2009-07-27 13:14 1050.xml
-rw-r--r-- 1 root root 750 2009-07-27 13:14 1051.xml
-rw-r--r-- 1 root root 750 2009-07-27 13:14 1052.xml
-rw-r--r-- 1 root root 750 2009-08-06 08:59 1053.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:56 1054.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:56 1055.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:56 1056.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:56 1057.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:57 1058.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:57 1059.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:57 1060.xml
-rw-r--r--
Re: [Freeswitch-users] problem when adding more extension
Thanks allot Kevin.
I felt it is about a missing configuration parameter
On Wed, Aug 12, 2009 at 1:31 PM, Kevin Goldingke...@kgolding.co.uk wrote:
Edit the line below as shown (in the dialplan/default.xml file)
Original line (about line 206)
condition field=desination_number expression=^(10[01][0-9])$
Replacement line
condition field=desination_number expression=^(10[0-9][0-9])$
This will allow extensions number 1000 to 1099.
Kevin
Tzury Bar Yochay wrote:
still not working, I mean, I can initiate a call from 1060 to 1000 but
not from 1000 to 1060.
1060 is just an example. This applies to all new extension I have
added (beyond to the default 1000-1019).
as you can see below I added them all to group name=support
This is how the confs look like
/usr/local/freeswitch/conf/directory# cat default.xml
include
!--the domain or ip (the right hand side of the @ in the addr--
domain name=$${domain}
params
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}/
/params
variables
variable name=record_stereo value=true/
variable name=default_gateway value=$${default_provider}/
variable name=default_areacode value=$${default_areacode}/
variable name=transfer_fallback_extension value=operator/
/variables
groups
group name=default
users
X-PRE-PROCESS cmd=include data=default/*.xml/
/users
/group
group name=sales
users
!--
type=pointer is a pointer so you can have the
same user in multiple groups. It basically means
to keep searching for the user in the directory.
--
user id=1000 type=pointer/
user id=1001 type=pointer/
user id=1002 type=pointer/
user id=1003 type=pointer/
user id=1004 type=pointer/
/users
/group
group name=billing
users
user id=1005 type=pointer/
user id=1006 type=pointer/
user id=1007 type=pointer/
user id=1008 type=pointer/
user id=1009 type=pointer/
/users
/group
group name=support
users
user id=1010 type=pointer/
user id=1011 type=pointer/
user id=1012 type=pointer/
user id=1013 type=pointer/
user id=1014 type=pointer/
user id=1015 type=pointer/
user id=1016 type=pointer/
user id=1017 type=pointer/
user id=1018 type=pointer/
user id=1019 type=pointer/
user id=1020 type=pointer/
user id=1050 type=pointer/
user id=1051 type=pointer/
user id=1052 type=pointer/
user id=1053 type=pointer/
user id=1054 type=pointer/
user id=1055 type=pointer/
user id=1056 type=pointer/
user id=1057 type=pointer/
user id=1058 type=pointer/
user id=1059 type=pointer/
user id=1060 type=pointer/
user id=1061 type=pointer/
user id=1062 type=pointer/
user id=1063 type=pointer/
/users
/group
/groups
/domain
/include
and the xml files under directory/default
r...@snoip-srv-001:/usr/local/freeswitch/conf/directory/default# ls -l
total 164
-rw-r--r-- 1 root root 750 2009-07-21 19:44 1000.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1001.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1002.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1003.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1004.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1005.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1006.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1007.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1008.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1009.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1010.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1011.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1012.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1013.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1014.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1015.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1016.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1017.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1018.xml
-rw-r--r-- 1 root root 750 2009-07-20 23:47 1019.xml
-rw-r--r-- 1 root root 750 2009-08-12 06:49 1020.xml
-rw-r--r-- 1 root root 750 2009-07-27 13:14 1050.xml
-rw-r--r-- 1 root root 750 2009-07-27 13:14 1051.xml
-rw-r--r-- 1 root root 750 2009-07-27 13:14 1052.xml
-rw-r--r-- 1 root root 750 2009-08-06 08:59 1053.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:56 1054.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:56 1055.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:56 1056.xml
-rw-r--r-- 1 root root 750 2009-08-11 09:56 1057.xml
-rw-r--r-- 1 root root 750
Re: [Freeswitch-users] problem when adding more extension
On Aug 12, 2009, at 2:44 PM, Tzury Bar Yochay wrote:
Hi,
I wanted to add more extension to freeswitch.
to add extension 1050 with password 1234 I did the following:
$ cd /usr/local/freeswitch/conf/directory/default
created 1050.xml having all '1000' strings replaced by '1050' by
typing
$ sed s/1000/1050/g 1000.xml 1050.xml
just run reloadxml should be ok no need to rescan the profile
rescan and reload the xml by typing into the CLI
freeswi...@internal sofia profile internal rescan reloadxml
However, when I tried to login with these credentials I got the
following in the fs_cli:
2009-08-12 06:35:01 [DEBUG] sofia_reg.c:1469 sofia_reg_parse_auth()
SIP username 1050 does not match auth username
2009-08-12 06:35:01 [DEBUG] sofia_reg.c:869
sofia_reg_handle_register() Send challenge for
[1...@server_address.net]
below are the content of 1000 and 1050 xml files
please advise.
$ cat 1050.xml
include
user id=1050 mailbox=1050
params
param name=password value=1234/
param name=vm-password value=1050/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=1050/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 1050/
variable name=effective_caller_id_number value=1050/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
/variables
/user
/include
$ cat 1000.xml
include
user id=1000 mailbox=1000
params
param name=password value=1234/
param name=vm-password value=1000/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=1000/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 1000/
variable name=effective_caller_id_number value=1000/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
/variables
/user
/include
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Re: [Freeswitch-users] answer command
It's not Eyebeam but FS hung up the call because it have nothing to do after answer. You should either playback a sound, do the echo command, record, hold the call, bridge to another channel or transfer somewhere else. On Aug 12, 2009, at 4:54 PM, Maxim Tsvetov wrote: I've tried to use answer command from outbound event socket and it's working, but the problem is that FS answering the call, but SIP Client (we tried this with EyeBeam and CISCO 7960) doesn't know that call was answered. So, as long as FS doesn't know what to do with this number it then disconnects the call. 2009-08-12 11:25:07.599250 [NOTICE] mod_dptools.c:649 Channel [sofia/internal/sip:1...@10.107.181.160:42840] has been answered 2009-08-12 11:25:07.599250 [NOTICE] switch_ivr_originate.c:2015 Channel [sofia/internal/1...@10.107.249.12] has been answered 2009-08-12 11:25:07.614875 [ERR] switch_core_io.c:118 sofia/internal/sip:1...@10.107.181.160:42840 has no read codec. 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:503 Hangup sofia/internal/sip:1...@10.107.181.160:42840 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:1016 Hangup sofia/internal/1...@10.107.249.12 [CS_EXECUTE] [NORMAL_CLEARING] 2009-08-12 11:25:07.630500 [NOTICE] switch_core_session.c:1086 Session 133 (sofia/internal/sip:1...@10.107.181.160:42840) Ended Maybe there is the way to acknowledge SIP client that call was answered? Regards, Maxim Tsvetov Diego Viola wrote: I suggest that you learn the differences between mod_commands commands and mod_dptools applications, and also the interfaces where you can access and use them. As said before, mod_dptools is accessible from dialplan, event socket outbound, etc. and mod_commands is accessible from the CLI, event socket (inbound/outbound), XML RPC, etc. That's all described in the wiki I think. Let us know if you have any questions =D. On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola diego.vi...@gmail.com wrote: Michael, you're welcome :). Milena, answer is a mod_dptools command, you can use it from the XML dialplan or from the event socket outbound. mod_commands API are APIs that you execute from the socket, event socket inbound, etc. But you can also execute them from event socket outbound using the api command. I hope that makes sense, correct me if I'm wrong =D. On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins m...@freeswitch.orgwrote: On Tue, Aug 11, 2009 at 9:05 AM, Milena testeado...@gmail.com wrote: Hello Brian, I wanna fix the wiki, but to make sure i got it right, does it only work on outbound event socket? or is there any other scenario where it would work. FYI, Diego Viola fixed the wiki. (Thanks Diego!) -MC ___ 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/answer-command-tp24912812p24931876.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] Fwd: Fwd: Scheduler in module
Hi,
In my LOAD_FUNCTION, I am trying to have freeswitch to flush out some data
every 10 s. The following lines of code does not show any effect at all.
switch_scheduler_task_thread_start();
switch_scheduler_add_task(switch_epoch_time_now(NULL),
data_flush_callback, data_flush,core,0,NULL,SSHF_NONE|SSHF_NO_DEL);
SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, starting to flush
data buffer...\n);
task-runtime = switch_time_now() + 10;
}
Does anyone know how to get it to work?
Thanks,
Mark
-- Forwarded message --
From: Brian West br...@freeswitch.org
Date: Mon, Aug 10, 2009 at 8:53 PM
Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
To: freeswitch-users@lists.freeswitch.org
switch_rtp.c has a simple one for the zrtp cache storing.
/b
On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
Re schedule is done in your callback, take a look at places that use
these apis in the code for details.
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[Freeswitch-users] freeswitch time conversion
Does anyone know how to take the epoch time in switch_event_t and convert it into a format such as Sat Jul 5 02:44:33 2009? Is there any existing facility that I can use for this purpose? br, JB ___ 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] hangup_after_bridge=false only works if bypass_media=false
I posted it yesterday evening: http://jira.freeswitch.org/browse/FSCORE-417
On Tue, Aug 11, 2009 at 9:43 PM, Michael Jerris m...@jerris.com wrote:
Please post a bug for this on jira.freeswitch.org.
Mike
On Aug 11, 2009, at 2:29 PM, Jeremiah Johnson wrote:
This is an integral part of my application. I need to have
FreeSWITCH outside of the media path as well as be able to do
multiple bridges for the same A leg.
/*WORKS*/
action application=set data=hangup_after_bridge=false/
action application=bridge data=sofia/gateway/${mygateway}/1$
{client_one}/
action application=bridge data=sofia/gateway/${mygateway}/1$
{client_two}/
/*DOES NOT WORK*/
action application=set data=hangup_after_bridge=false/
action application=set data=bypass_media=true/
action application=bridge data=sofia/gateway/${mygateway}/1$
{client_one}/
action application=bridge data=sofia/gateway/${mygateway}/1$
{client_two}/
In the DOES NOT WORK example, the A leg hangs up as soon as the
leg for client_one hangs up.
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Re: [Freeswitch-users] Loopback and bypass_media
David / Michael - thanks for your your replies. The SoftIVR example is particularly useful. Must admit though - I was hoping not to have to do any custom stuff at this stage. It does appear there is no method to do this by staking bridge lines so I will put an issue in jira to try and get loopback working with bypass_media. In the meantime I will also start looking to build a custom bridging app. As I said though - not a road I wanted to go down. Thanks for your help! Phillip Jones On Tue, Aug 11, 2009 at 6:13 PM, Michael Giagnocavom...@giagnocavo.net wrote: It's also simple enough to write a plugin in one of the scripting languages to add an app to do exactly what you want... -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of David Knell Sent: Tuesday, August 11, 2009 1:55 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Loopback and bypass_media Just to add my $0.02-worth (if you're feeling generous..) - I don't think that the dialplan is expressive enough to do what's needed here, and that's where the trouble's coming from. It's not enormously tricky to build a generic dial this set of numbers according to these rules service using something hanging off the event socket - there's a writeup here: http://www.softivr.com/wiki/index.php/Find_me showing how it could be done on SoftIVR. To roll something similar yourself using the event socket, you'd need to map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge', and have some way of passing messages around between the threads handling the different call legs, assuming that you're using one thread per leg. --Dave ___ 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] Loopback and bypass_media
perhaps we need to add some syntax + logic to originate: application=originate data=(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum) This would acomplish the equiv of loopback/bar,loopback/yum where bar and yum are then further expanded in the dialplan as sofia/foo/bar|sofia/baz/bar and sofia/foo/yum|sofia/baz/yum except that the threads of execution are handled directly by originate. I'm not sure that is really the solution since each () group would still have to be a separate thread to run independently. To me, loopback is the way to accomplish this issue (how I've done it with the same requirements that you have) since all the hard work is layered and works. The problem is that you require bypass_media which doesn't play nice with loopback. Perhaps have an on answer hook that tries to enable bypass media (re-invite) after the call is setup? On Wed, Aug 12, 2009 at 8:29 AM, Phillip Jonespjinthe...@gmail.com wrote: David / Michael - thanks for your your replies. The SoftIVR example is particularly useful. Must admit though - I was hoping not to have to do any custom stuff at this stage. It does appear there is no method to do this by staking bridge lines so I will put an issue in jira to try and get loopback working with bypass_media. In the meantime I will also start looking to build a custom bridging app. As I said though - not a road I wanted to go down. Thanks for your help! Phillip Jones On Tue, Aug 11, 2009 at 6:13 PM, Michael Giagnocavom...@giagnocavo.net wrote: It's also simple enough to write a plugin in one of the scripting languages to add an app to do exactly what you want... -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of David Knell Sent: Tuesday, August 11, 2009 1:55 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Loopback and bypass_media Just to add my $0.02-worth (if you're feeling generous..) - I don't think that the dialplan is expressive enough to do what's needed here, and that's where the trouble's coming from. It's not enormously tricky to build a generic dial this set of numbers according to these rules service using something hanging off the event socket - there's a writeup here: http://www.softivr.com/wiki/index.php/Find_me showing how it could be done on SoftIVR. To roll something similar yourself using the event socket, you'd need to map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge', and have some way of passing messages around between the threads handling the different call legs, assuming that you're using one thread per leg. --Dave ___ 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 -- -Rupa ___ 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] FW: Sangoma/FS...
Hello Peter, I'd appreciate if you can keep the discussion going in the freeswitch-users mailing list, there are other people there that will benefit of the discussion or even can help. Read my comments below. On Wed, Aug 12, 2009 at 5:59 AM, Peter Olsson peter.ols...@visionutveckling.se wrote: Sorry for spamming you :) But I have some more results now. I’ve tried using another lab PBX with Q.SIG enabled, and when using that one I’m able to connect calls as I should. At least incoming to FS, outgoing seem to have some problems still.. So the problem for the PBX I used yesterday seems to be both related to Q.SIG (maybe) and the PBX itself (it does connect to other providers though, so I know the trunk works). Should I take some dumps from the PRI card to try to find out why it didn’t work with the first one, or is this “as expected”, since they have Q.SIG enabled? I have no experience with Q.SIG, so I won't be able to help much. One thing though, is that if I were you, I'd be using openzap with libpri support, is that what you are using, or are you using the ISDN openzap stack? As of the dumps, they may help, or not, but pastebin them anyways so I can make an un-educated guess. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Build Issue on Solaris 10 (FS v1.0.4pre9 v1.0.4)
Hi Bruce, I am having similar issues trying build freeswitch 1.0.4 on Solaris x86 as well. I sent some information over the mailing list, and I received a response from Michal Bielicki (attached), stating he'd test this and direct me to the steps to successfully build freeswitch. Just an FYI in case you see his response. Vladimir -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Bruce McAlister Sent: Wednesday, August 12, 2009 4:44 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 v1.0.4) Hi All, I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and 1.0.4. I am running on Solaris 10 Update 5 on x86 hardware (32-bit). The build fails with: --- snip --- make: Fatal error: Command failed for target `all-recursive' Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4 *** Error code 1 make: Fatal error: Command failed for target `all' --- Looking back through the build I can see the following error: --- snip --- creating libfreeswitch.la (cd .libs rm -f libfreeswitch.la ln -s ../libfreeswitch.la libfreeswitch.la) /usr/bin/cc -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/s rc -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes -DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch freeswitch-switch.o ./.libs/libfreeswitch.so -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/ expat/lib /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/ex pat/lib/.libs/libexpat.a /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libap r-1.a -lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp -L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl -lcurses -lsocket -R/opt/freeswitch/lib -R/usr/sfw/lib Undefined first referenced symbol in file herror ./.libs/libfreeswitch.so ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch *** Error code 1 The following command caused the error: `if test -z ; then echo /bin/bash /export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` --tag=CC --mode=link /usr/bin/cc -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/s rc -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes -DPATH_MAX=2048 -g -v -Xc -xc99=all -lm -R/opt/freeswitch/lib -o freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib freeswitch-switch.o libfreeswitch.la libs/apr/libapr-1.la libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl -lcrypto -lsocket -lnsl -ldl -lcurses -lsocket --- snip --- Then a little above this error, there is the following warning that is displayed (I'm not sure if it is related): --- snip --- *** Warning: Linking the shared library libfreeswitch.la against the *** static library libs/libedit/src/.libs/libedit.a is not portable! --- snip --- My configure line is as follows: --- ./configure --prefix=/opt/freeswitch --- I have the complete configure and make output if anyone needs them. Any help/pointers would be greatly appreciated. Thanks Bruce ___ 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-use rs http://www.freeswitch.org ---BeginMessage--- I'll retst it later today and give you a link with instructions Am 10.08.2009 um 20:14 schrieb vmorales: By ./compile I was referring to ./configure Vladimir -Original Message- From: vmorales [mailto:email.list.subscri...@gmail.com] Sent: Monday, August 10, 2009 11:49 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86 Thanks for the response(s): I ran the ./compile script with a set PREFIX. This took a few attempts with errors before it was able to complete error-free, as I had to install libtool. Since then, I have tried running 'make', 'gmake', and '/opt/gnu/bin/make', but each results with an error. This is the error when running 'make' or 'gmake': snip make: Fatal error: Command failed for target `all-recursive' Current working directory /home/vmorales/freeswitch-1.0.4 *** Error code 1 make: Fatal error: Command failed for target `all' /snip This is the error when running '/opt/gnu/bin/make': snip make[5]: *** [mod_amr.so] Error 1 make[4]: *** [all] Error 1
Re: [Freeswitch-users] Sangoma/FS...
Of course – no problem! I’m not using libpri support now, I don’t think it’s ported for Windows (yet)? I’ll try it out some more, and try to detect what’s going wrong... /Peter Från: Moises Silva [mailto:moises.si...@gmail.com] Skickat: den 12 augusti 2009 16:17 Till: freeswitch-users@lists.freeswitch.org Kopia: Peter Olsson Ämne: Re: FW: Sangoma/FS... Hello Peter, I'd appreciate if you can keep the discussion going in the freeswitch-users mailing list, there are other people there that will benefit of the discussion or even can help. Read my comments below. On Wed, Aug 12, 2009 at 5:59 AM, Peter Olsson peter.ols...@visionutveckling.semailto:peter.ols...@visionutveckling.se wrote: Sorry for spamming you :) But I have some more results now. I’ve tried using another lab PBX with Q.SIG enabled, and when using that one I’m able to connect calls as I should. At least incoming to FS, outgoing seem to have some problems still.. So the problem for the PBX I used yesterday seems to be both related to Q.SIG (maybe) and the PBX itself (it does connect to other providers though, so I know the trunk works). Should I take some dumps from the PRI card to try to find out why it didn’t work with the first one, or is this “as expected”, since they have Q.SIG enabled? I have no experience with Q.SIG, so I won't be able to help much. One thing though, is that if I were you, I'd be using openzap with libpri support, is that what you are using, or are you using the ISDN openzap stack? As of the dumps, they may help, or not, but pastebin them anyways so I can make an un-educated guess. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.commailto:m...@sangoma.com !DSPAM:4a82cefe32931477278362! ___ 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 time conversion
Hi, The standard C function is strftime. FreeSWITCH has some wrapped ones: switch_apr.h:SWITCH_DECLARE(switch_status_t) switch_strftime(char *s, switch_size_t *retsize, switch_size_t max, const char *format, switch_time_exp_t *tm); switch_apr.h:SWITCH_DECLARE(switch_status_t) switch_strftime_nocheck(char *s, switch_size_t *retsize, switch_size_t max, const char *format, switch_time_exp_t *tm); switch_core.h:SWITCH_DECLARE(switch_status_t) switch_strftime_tz(const char *tz, const char *format, char *date, size_t len, switch_time_t thetime); Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 12-Aug-09, at 8:01 AM, Juan Backson wrote: Does anyone know how to take the epoch time in switch_event_t and convert it into a format such as Sat Jul 5 02:44:33 2009? Is there any existing facility that I can use for this purpose? br, JB ___ 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] Fwd: Fwd: Scheduler in module
Hi,
I did the same thing on my side
API CALL [load(mod_skel)] output:
+OK
2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task 2
data_flush (core) to run at 1250089698
2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889
Successfully Loaded [mod_skel]
2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270 Adding
API Function 'skel'
freeswi...@maths-mac.local 2009-08-12 11:08:18.207113 [ERR]
mod_skel.c:120 starting to flush data buffer...
Note that you don't need to start the thread manually, the core
already has threads running for the scheduler.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 12-Aug-09, at 7:26 AM, mark morreny wrote:
Hi,
In my LOAD_FUNCTION, I am trying to have freeswitch to flush out
some data every 10 s. The following lines of code does not show any
effect at all.
switch_scheduler_task_thread_start();
switch_scheduler_add_task(switch_epoch_time_now(NULL),
data_flush_callback, data_flush,core,0,NULL,SSHF_NONE|
SSHF_NO_DEL);
SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, starting to
flush data buffer...\n);
task-runtime = switch_time_now() + 10;
}
Does anyone know how to get it to work?
Thanks,
Mark
-- Forwarded message --
From: Brian West br...@freeswitch.org
Date: Mon, Aug 10, 2009 at 8:53 PM
Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
To: freeswitch-users@lists.freeswitch.org
switch_rtp.c has a simple one for the zrtp cache storing.
/b
On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
Re schedule is done in your callback, take a look at places that use
these apis in the code for details.
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Re: [Freeswitch-users] Loopback and bypass_media
Hi there, application=originate data=(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum) I agree. However, perhaps the ideal is not to specify the carriers at this level, as carriers are added and removed fairly often as costings change. So it would be nice to have some sort of proxy that resolves to a list of carriers: application=originate data=sofia/MyCarriers/bar,sofia/MyCarriers/yum MyCarriers action application=carrier data=sofia/foo/ action application=carrier data=sofia/baz/ action application=carrier data=sofia/etc/ /MyCarriers or something similar. This would achieve the same as loopback in this use case but without dangers of looping? Complicated stuff though. Perhaps have an on answer hook that tries to enable bypass media (re-invite) after the call is setup? That's a good idea - I will look into that. Thanks again. Phillip On Wed, Aug 12, 2009 at 10:22 AM, Rupa Schomakerr...@rupa.com wrote: perhaps we need to add some syntax + logic to originate: application=originate data=(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum) This would acomplish the equiv of loopback/bar,loopback/yum where bar and yum are then further expanded in the dialplan as sofia/foo/bar|sofia/baz/bar and sofia/foo/yum|sofia/baz/yum except that the threads of execution are handled directly by originate. I'm not sure that is really the solution since each () group would still have to be a separate thread to run independently. To me, loopback is the way to accomplish this issue (how I've done it with the same requirements that you have) since all the hard work is layered and works. The problem is that you require bypass_media which doesn't play nice with loopback. Perhaps have an on answer hook that tries to enable bypass media (re-invite) after the call is setup? On Wed, Aug 12, 2009 at 8:29 AM, Phillip Jonespjinthe...@gmail.com wrote: David / Michael - thanks for your your replies. The SoftIVR example is particularly useful. Must admit though - I was hoping not to have to do any custom stuff at this stage. It does appear there is no method to do this by staking bridge lines so I will put an issue in jira to try and get loopback working with bypass_media. In the meantime I will also start looking to build a custom bridging app. As I said though - not a road I wanted to go down. Thanks for your help! Phillip Jones On Tue, Aug 11, 2009 at 6:13 PM, Michael Giagnocavom...@giagnocavo.net wrote: It's also simple enough to write a plugin in one of the scripting languages to add an app to do exactly what you want... -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of David Knell Sent: Tuesday, August 11, 2009 1:55 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Loopback and bypass_media Just to add my $0.02-worth (if you're feeling generous..) - I don't think that the dialplan is expressive enough to do what's needed here, and that's where the trouble's coming from. It's not enormously tricky to build a generic dial this set of numbers according to these rules service using something hanging off the event socket - there's a writeup here: http://www.softivr.com/wiki/index.php/Find_me showing how it could be done on SoftIVR. To roll something similar yourself using the event socket, you'd need to map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge', and have some way of passing messages around between the threads handling the different call legs, assuming that you're using one thread per leg. --Dave ___ 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 -- -Rupa ___ 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
Re: [Freeswitch-users] Loopback and bypass_media
On Wed, Aug 12, 2009 at 10:22 AM, Phillip Jonespjinthe...@gmail.com wrote: Hi there, application=originate data=(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum) I agree. However, perhaps the ideal is not to specify the carriers at this level, as carriers are added and removed fairly often as costings change. So it would be nice to have some sort of proxy that resolves to a list of carriers: application=originate data=sofia/MyCarriers/bar,sofia/MyCarriers/yum MyCarriers action application=carrier data=sofia/foo/ action application=carrier data=sofia/baz/ action application=carrier data=sofia/etc/ /MyCarriers or something similar. This would achieve the same as loopback in this use case but without dangers of looping? Complicated stuff though. Well, that is all done by mod_lcr. I was simplifying to narrow down to just originate. First we need to see if this is worth pursuing over fixing (modifying, whatever) loopback to handle bypass media. If it is, then I'll modify mod_lcr to deal with the situation in question (comma or pipe sep list of numbers to call. mod_lcr would then group as appropriate). Right now, my bridge is setup in a small javascript script that builds the appropriate dialstring (using loopback for external calls, user/ for internal calls) and then when doing the loopback call to mod_lcr to get the dialstring with all providers in the right order. Perhaps have an on answer hook that tries to enable bypass media (re-invite) after the call is setup? That's a good idea - I will look into that. Thanks again. Phillip Let us know how it works for you... -- -Rupa ___ 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] answer command
I will try to paraphrase my question. Is there any possibility to answer call from CTI application and synchronise answer with answer in SIP client?Maybe we can use SIP functions in our CTI application instead of FS api commands? I'm trying to find the way to make prototype of lineAnswer command in TAPI. Seven Du wrote: It's not Eyebeam but FS hung up the call because it have nothing to do after answer. You should either playback a sound, do the echo command, record, hold the call, bridge to another channel or transfer somewhere else. On Aug 12, 2009, at 4:54 PM, Maxim Tsvetov wrote: I've tried to use answer command from outbound event socket and it's working, but the problem is that FS answering the call, but SIP Client (we tried this with EyeBeam and CISCO 7960) doesn't know that call was answered. So, as long as FS doesn't know what to do with this number it then disconnects the call. 2009-08-12 11:25:07.599250 [NOTICE] mod_dptools.c:649 Channel [sofia/internal/sip:1...@10.107.181.160:42840] has been answered 2009-08-12 11:25:07.599250 [NOTICE] switch_ivr_originate.c:2015 Channel [sofia/internal/1...@10.107.249.12] has been answered 2009-08-12 11:25:07.614875 [ERR] switch_core_io.c:118 sofia/internal/sip:1...@10.107.181.160:42840 has no read codec. 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:503 Hangup sofia/internal/sip:1...@10.107.181.160:42840 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:1016 Hangup sofia/internal/1...@10.107.249.12 [CS_EXECUTE] [NORMAL_CLEARING] 2009-08-12 11:25:07.630500 [NOTICE] switch_core_session.c:1086 Session 133 (sofia/internal/sip:1...@10.107.181.160:42840) Ended Maybe there is the way to acknowledge SIP client that call was answered? Regards, Maxim Tsvetov Diego Viola wrote: I suggest that you learn the differences between mod_commands commands and mod_dptools applications, and also the interfaces where you can access and use them. As said before, mod_dptools is accessible from dialplan, event socket outbound, etc. and mod_commands is accessible from the CLI, event socket (inbound/outbound), XML RPC, etc. That's all described in the wiki I think. Let us know if you have any questions =D. On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola diego.vi...@gmail.com wrote: Michael, you're welcome :). Milena, answer is a mod_dptools command, you can use it from the XML dialplan or from the event socket outbound. mod_commands API are APIs that you execute from the socket, event socket inbound, etc. But you can also execute them from event socket outbound using the api command. I hope that makes sense, correct me if I'm wrong =D. On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins m...@freeswitch.orgwrote: On Tue, Aug 11, 2009 at 9:05 AM, Milena testeado...@gmail.com wrote: Hello Brian, I wanna fix the wiki, but to make sure i got it right, does it only work on outbound event socket? or is there any other scenario where it would work. FYI, Diego Viola fixed the wiki. (Thanks Diego!) -MC ___ 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/answer-command-tp24912812p24931876.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 -- View this message in context: http://www.nabble.com/answer-command-tp24912812p24940548.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] answer command
Well you can only truly answer an inbound call to FS... you can't force answer an outbound call. /b On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote: I will try to paraphrase my question. Is there any possibility to answer call from CTI application and synchronise answer with answer in SIP client?Maybe we can use SIP functions in our CTI application instead of FS api commands? I'm trying to find the way to make prototype of lineAnswer command in TAPI. ___ 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] random route selection
Hi, I would like to implement a random route selection based on some arbitrary percentage. Does anyone know if there is any good way of doing that within freeswitch? If there isn't any api that I can use, does freeswitch has any random generator that I can be used for this purpose? br, JB ___ 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] answer command
If I have two FS extensions A and B. I'm calling from A to B and want to answer from B-side in my CTI application and to make SIP phone to be synchronised to my CTI application. Is it possible to do it? Brian West-3 wrote: Well you can only truly answer an inbound call to FS... you can't force answer an outbound call. /b On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote: I will try to paraphrase my question. Is there any possibility to answer call from CTI application and synchronise answer with answer in SIP client?Maybe we can use SIP functions in our CTI application instead of FS api commands? I'm trying to find the way to make prototype of lineAnswer command in TAPI. ___ 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/answer-command-tp24912812p24941422.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] random route selection
mod_lcr will do random route selection if the rates are the same. But that gives an equal distribution. There is no weighting/percentage supported. On Wed, Aug 12, 2009 at 12:21 PM, Juan Backsonjuanback...@gmail.com wrote: Hi, I would like to implement a random route selection based on some arbitrary percentage. Does anyone know if there is any good way of doing that within freeswitch? If there isn't any api that I can use, does freeswitch has any random generator that I can be used for this purpose? br, JB -- -Rupa ___ 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] answer command
Sip does not support this functionality. The called device would have to support this via some other mechanism such as ctsa which I have seen recently someone was looking at for freeswitch. So the first issue you must resolve is the called device needs to support some way to do this. Mike On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov maxim.tsve...@gmail.com wrote: If I have two FS extensions A and B. I'm calling from A to B and want to answer from B-side in my CTI application and to make SIP phone to be synchronised to my CTI application. Is it possible to do it? Brian West-3 wrote: Well you can only truly answer an inbound call to FS... you can't force answer an outbound call. /b On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote: I will try to paraphrase my question. Is there any possibility to answer call from CTI application and synchronise answer with answer in SIP client?Maybe we can use SIP functions in our CTI application instead of FS api commands? I'm trying to find the way to make prototype of lineAnswer command in TAPI. ___ 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/answer-command-tp24912812p24941422.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] Confused about conferences
I have been reading all the docs about conferences I can find and am
getting somewhat confused. What I am trying to do is set up a dialplan
where I have subscribers with extensions in the 1xx range, and then to
set an ability to have a series of conference rooms for each subscriber
in the 21xx range where if the user enters is own conference he is the
moderator, but if not he is just a normal user.
I want to be able for the moderator to do things like mute or kick people.
So dialplan would probably have something like this in it
extension name=user_conference
condition field=destination_number
expression=^(2${caller_id_number})$
action application=set data=i_am_moderator=true
action application=answer/
action application=conference data=$...@default+flags{moderator}/
/condition
condition field=destination_number expression=^(21\d{2})^
action application=set data=i_am_moderator=false
action application=answer/
action application=conference data=$...@default/
/condition
/extension
extension name=kick
condition field=i_am_moderator expression=true
action application=play_and_get_digits data=3 3 3 7000 #
/ask-for-extension.wav /invalid.wav conf-user-id \d+
WHAT GOES HERE???
/condition
/extension
in the conference.conf.xml file, I would change the caller controls to
include
caller-controls
group name=somekeys
control action=transfer digits=9 data=kick XML default/
/group
/caller-controls
My question (at the moment) is
In the WHAT GOES HERE place how do it
Kick extension ${conf-user-id} (DOES IT REQUIRE A SCRIPT TO CALL THE
Conference API?)
Re-Enter the moderator back into the conference
Re-Enter the ordinary user who happened to press 9 back into the conference
I am assuming I can't stop the non moderator getting the control - since
all users get the same controls.
--
Alan Chandler
http://www.chandlerfamily.org.uk
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[Freeswitch-users] fs installation is broken in version: FreeSWITCH Version 1.0.trunk (14500M)
I just did a rebootstrap on a fs box, it turned out the new revision has this at the end of mod_sofia.h: char *sofia_glue_get_extra_headers(switch_channel_t *channel, const char *prefix); .mine void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const *sip, const char *prefix); === void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const *sip, const char *prefix); .r14490 no wonder why it wouldn't compile :P ___ 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] fs installation is broken in version: FreeSWITCH Version 1.0.trunk (14500M)
You have a merge conflict. svn revert sofia_glue.c /b On Aug 12, 2009, at 12:59 PM, Milena wrote: I just did a rebootstrap on a fs box, it turned out the new revision has this at the end of mod_sofia.h: char *sofia_glue_get_extra_headers(switch_channel_t *channel, const char *prefix); .mine void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const *sip, const char *prefix); === void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const *sip, const char *prefix); .r14490 no wonder why it wouldn't compile :P ___ 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] The application hash
I was trying to explore the documentation for the application hash which is in the default dialplan. Its vaguely obvious what its doing, but I wanted to be sure. It appears to be listed as a application under dp_tools, but when I click on it I get taken to a page that talks about limit_hash rather than hash. Is there any documentation for this? Am I looking in the wring place? -- Alan Chandler http://www.chandlerfamily.org.uk ___ 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] random route selection
Write you a small C app to randomly return them based on the percentages... Currently we do something similar to this but use a random round robin based thing using a simple sql backend and doing a select order by random sort of thing... Contact me off list if you need some profession help figuring this out K From: Juan Backson juanback...@gmail.com Reply-To: freeswitch-users@lists.freeswitch.org Date: Thu, 13 Aug 2009 01:21:07 +0800 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] random route selection Hi, I would like to implement a random route selection based on some arbitrary percentage. Does anyone know if there is any good way of doing that within freeswitch? If there isn't any api that I can use, does freeswitch has any random generator that I can be used for this purpose? br, JB ___ 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] fs installation is broken in version: FreeSWITCH Version 1.0.trunk (14500M)
It's modified because it wouldn't compile with those at the end of the file 2009/8/12 Michael Jerris m...@jerris.com The M in the version number means modified. You had local code changes that conflicted when you updated trunk. Revert the changes to that file and it should be fine. Mike On Aug 12, 2009, at 10:59 AM, Milena testeado...@gmail.com wrote: I just did a rebootstrap on a fs box, it turned out the new revision has this at the end of mod_sofia.h: char *sofia_glue_get_extra_headers(switch_channel_t *channel, const char *prefix); .mine void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const *sip, const char *prefix); === void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const *sip, const char *prefix); .r14490 no wonder why it wouldn't compile :P ___ 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] The application hash
hash is just like db but its all in memory.. you can interchange db and hash. /b On Aug 12, 2009, at 1:25 PM, Alan Chandler wrote: I was trying to explore the documentation for the application hash which is in the default dialplan. Its vaguely obvious what its doing, but I wanted to be sure. It appears to be listed as a application under dp_tools, but when I click on it I get taken to a page that talks about limit_hash rather than hash. Is there any documentation for this? Am I looking in the wring place? -- Alan Chandler http://www.chandlerfamily.org.uk ___ 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] fs installation is broken in version: FreeSWITCH Version 1.0.trunk (14500M)
Ok, done and fixed, thank you very much :) 2009/8/12 Milena testeado...@gmail.com It's modified because it wouldn't compile with those at the end of the file 2009/8/12 Michael Jerris m...@jerris.com The M in the version number means modified. You had local code changes that conflicted when you updated trunk. Revert the changes to that file and it should be fine. Mike On Aug 12, 2009, at 10:59 AM, Milena testeado...@gmail.com wrote: I just did a rebootstrap on a fs box, it turned out the new revision has this at the end of mod_sofia.h: char *sofia_glue_get_extra_headers(switch_channel_t *channel, const char *prefix); .mine void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const *sip, const char *prefix); === void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const *sip, const char *prefix); .r14490 no wonder why it wouldn't compile :P ___ 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] ClueCon Presentations - Where?
Hi there, Does anyone have the URL for where I might find all the electronic versions of the presentations made at ClueCon last week? 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] ClueCon Presentations - Where?
They are all getting gathered up and put online... files.freeswitch.org/cluecon_2009 just keep an eye there some of the videos are up also. /b On Aug 12, 2009, at 2:18 PM, Christian Jensen wrote: Hi there, Does anyone have the URL for where I might find all the electronic versions of the presentations made at ClueCon last week? 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] Cluecon 2009
I to would like to put my thanks on the table. I have been going to conferences for a very long time and often question the value of taking time off to attend these venues. When I was asked to attend by a client I was very hesitant. I am very pleased that I decided to attend. Now the skeptical among you may say that I am only pleased because I won the MacBook. I cannot deny there is truth in that statement. However, on Thursday morning I was sitting in the conference room waiting for the first presentation. I was thinking that it had been a valuable three days. I had been able to connect with some clients at the conference, made some new contacts, met people face to face that I had only met online, listened to good presentations, learned some valuable new information, and lastly received some insight from Anthony and Michael Jerris on fixing a bug that had been plaguing me for sometime. As I sat pondering, I clicked on the picture of the MacBook and thought the only way the conference could end better was if I won the MacBook. Never did I actually think that would happen and the conference did end better. I will be much more motivated to consider attending next year, even knowing lightening does not strike the same spot twice. Jonathan Augenstine On Fri, Aug 7, 2009 at 3:54 PM, David Knell d...@3c.co.uk wrote: Just a quick note to say thanks to Cluecon's organisers for putting together such a useful, informative and packed three days. I've come away with a head full of ideas, a bunch of new contacts and a collection of things to do; I'd thoroughly recommend that anyone interested in IP telephony blocks out the first week of August 2010, right now..! Cheers -- Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ 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] Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS
Hello, anybody has a clue what this message means? [WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS What does VETO mean here? Best regards Peter ___ 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] Cluecon 2009
Dave, Thanks, Hope to see you there next year... /b On Aug 7, 2009, at 5:54 PM, David Knell wrote: Just a quick note to say thanks to Cluecon's organisers for putting together such a useful, informative and packed three days. I've come away with a head full of ideas, a bunch of new contacts and a collection of things to do; I'd thoroughly recommend that anyone interested in IP telephony blocks out the first week of August 2010, right now..! Cheers -- Dave ___ 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] Cluecon 2009
Remember next year we'll have more Mac Book's to give away and iPod Touches with engraved sponsor logos on them. :) /b On Aug 12, 2009, at 2:57 PM, jonathan augenstine wrote: I to would like to put my thanks on the table. I have been going to conferences for a very long time and often question the value of taking time off to attend these venues. When I was asked to attend by a client I was very hesitant. I am very pleased that I decided to attend. Now the skeptical among you may say that I am only pleased because I won the MacBook. I cannot deny there is truth in that statement. However, on Thursday morning I was sitting in the conference room waiting for the first presentation. I was thinking that it had been a valuable three days. I had been able to connect with some clients at the conference, made some new contacts, met people face to face that I had only met online, listened to good presentations, learned some valuable new information, and lastly received some insight from Anthony and Michael Jerris on fixing a bug that had been plaguing me for sometime. As I sat pondering, I clicked on the picture of the MacBook and thought the only way the conference could end better was if I won the MacBook. Never did I actually think that would happen and the conference did end better. I will be much more motivated to consider attending next year, even knowing lightening does not strike the same spot twice. Jonathan Augenstine ___ 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] Cluecon 2009
Macbook ... that's nothing, I got $1500 worth of coffee :-) On Wed, Aug 12, 2009 at 12:57 PM, jonathan augenstinejaugenst...@gmail.com wrote: I to would like to put my thanks on the table. I have been going to conferences for a very long time and often question the value of taking time off to attend these venues. When I was asked to attend by a client I was very hesitant. I am very pleased that I decided to attend. Now the skeptical among you may say that I am only pleased because I won the MacBook. I cannot deny there is truth in that statement. However, on Thursday morning I was sitting in the conference room waiting for the first presentation. I was thinking that it had been a valuable three days. I had been able to connect with some clients at the conference, made some new contacts, met people face to face that I had only met online, listened to good presentations, learned some valuable new information, and lastly received some insight from Anthony and Michael Jerris on fixing a bug that had been plaguing me for sometime. As I sat pondering, I clicked on the picture of the MacBook and thought the only way the conference could end better was if I won the MacBook. Never did I actually think that would happen and the conference did end better. I will be much more motivated to consider attending next year, even knowing lightening does not strike the same spot twice. Jonathan Augenstine On Fri, Aug 7, 2009 at 3:54 PM, David Knell d...@3c.co.uk wrote: Just a quick note to say thanks to Cluecon's organisers for putting together such a useful, informative and packed three days. I've come away with a head full of ideas, a bunch of new contacts and a collection of things to do; I'd thoroughly recommend that anyone interested in IP telephony blocks out the first week of August 2010, right now..! Cheers -- Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ 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 -- Want to buy my photo's ? : http://www.shutterstock.com/gallery.mhtml?id=309295rid=309295 ___ 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] Question about sharing conference between servers
Hello, I have spent the past couple of weeks toying around with FS to evaluate the possibility of using it for a large scale conference server for our organization. The plan is to have several FS servers initiate calls to participants and connect them together, but not have to transfer all of the calls to one server supporting the conference. My problem is that I do not see a simple way to link the servers together. Has anyone done anything like this, or know of a way to create a connection between servers without an actual caller on the line? I started to go down a path of registering a soft-phone on each machine and place a call those extensions prior to initiating the conference call -- then connecting, but that feels like a kludge.Any thoughts, suggestions or guidance would be greatly appreciated.T ___ 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] Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS
On Wed, Aug 12, 2009 at 4:04 PM, Peter P GMX prometheus...@gmx.net wrote: Hello, anybody has a clue what this message means? [WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS What does VETO mean here? Best regards Peter Means that state transition should not occur. The only thing that it would cause that (I think) is a bug in the the openzap code. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Cluecon 2009
So you're the one that drank 16 gallons of coffee! Good luck sleeping! /b On Aug 12, 2009, at 3:25 PM, Terry Moore-Read wrote: Macbook ... that's nothing, I got $1500 worth of coffee :-) ___ 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] Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS
Isn't progress_media already past progress in the state machine? so the state machine can't move backwards in states right? /b On Aug 12, 2009, at 3:33 PM, Moises Silva wrote: On Wed, Aug 12, 2009 at 4:04 PM, Peter P GMX prometheus...@gmx.net wrote: Hello, anybody has a clue what this message means? [WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS What does VETO mean here? Best regards Peter Means that state transition should not occur. The only thing that it would cause that (I think) is a bug in the the openzap code. ___ 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] Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS
Correct, so the question is why ozmod_libpri attempting to move from progress_media to progress ... may be a delayed libpri event? or some crap along those lines. On Wed, Aug 12, 2009 at 4:42 PM, Brian West br...@freeswitch.org wrote: Isn't progress_media already past progress in the state machine? so the state machine can't move backwards in states right? /b On Aug 12, 2009, at 3:33 PM, Moises Silva wrote: On Wed, Aug 12, 2009 at 4:04 PM, Peter P GMX prometheus...@gmx.net wrote: Hello, anybody has a clue what this message means? [WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS What does VETO mean here? Best regards Peter Means that state transition should not occur. The only thing that it would cause that (I think) is a bug in the the openzap code. ___ 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 -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS
Yes you can get a progress after you get a progress with media ... I have seen it. /b On Aug 12, 2009, at 3:57 PM, Moises Silva wrote: Correct, so the question is why ozmod_libpri attempting to move from progress_media to progress ... may be a delayed libpri event? or some crap along those lines. ___ 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] ClueCon Presentations - Where?
On Wed, Aug 12, 2009 at 2:25 PM, Brian West br...@freeswitch.org wrote: They are all getting gathered up and put online... files.freeswitch.org/cluecon_2009 just keep an eye there some of the videos are up also. /b FYI, I've uploaded the first batch and they should get synched up on files.freeswitch.org/cluecon_2009/presentations any time... -MC ___ 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] Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS
On Wed, Aug 12, 2009 at 4:01 PM, Brian West br...@freeswitch.org wrote: Yes you can get a progress after you get a progress with media ... I have seen it. Yes, you definitely can and I believe that some of the PRI specs suggest that this is totally legal, even though it's kind of silly. -MC ___ 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] Cluecon 2009
That's the trouble with a 8am conference in a town where the bars close at 4am :-) On Wed, Aug 12, 2009 at 1:42 PM, Brian Westbr...@freeswitch.org wrote: So you're the one that drank 16 gallons of coffee! Good luck sleeping! /b On Aug 12, 2009, at 3:25 PM, Terry Moore-Read wrote: Macbook ... that's nothing, I got $1500 worth of coffee :-) ___ 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 -- Want to buy my photo's ? : http://www.shutterstock.com/gallery.mhtml?id=309295rid=309295 ___ 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] Cluecon 2009
And it didn't help we had an open bar two of the nights! /b On Aug 12, 2009, at 4:27 PM, Terry Moore-Read wrote: That's the trouble with a 8am conference in a town where the bars close at 4am :-) ___ 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] Cluecon 2009
it helped me! oh... well, I helped myself! -giovanni On Wed, Aug 12, 2009 at 11:30 PM, Brian Westbr...@freeswitch.org wrote: And it didn't help we had an open bar two of the nights! /b On Aug 12, 2009, at 4:27 PM, Terry Moore-Read wrote: That's the trouble with a 8am conference in a town where the bars close at 4am :-) ___ 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] Question about sharing conference between servers
On Wed, Aug 12, 2009 at 2:55 PM, Tina Martinez t...@a2unlimited.com wrote:
Hello,
I have spent the past couple of weeks toying around with FS to evaluate the
possibility of using it for a large scale conference server for our
organization. The plan is to have several FS servers initiate calls to
participants and connect them together, but not have to transfer all of the
calls to one server supporting the conference. My problem is that I do not
see a simple way to link the servers together. Has anyone done anything
like this, or know of a way to create a connection between servers without
an actual caller on the line? I started to go down a path of registering a
soft-phone on each machine and place a call those extensions prior to
initiating the conference call -- then connecting, but that feels like a
kludge.
Any thoughts, suggestions or guidance would be greatly appreciated.
T
Tina,
Welcome to the FreeSWITCH community! I'm pretty sure that FS can do what you
need. My first question to you is this: do envision having some sort of call
control mechanism, like a script or something that knows which callers to
connect to and which ones to put into conferences, etc.? The reason I ask is
that controlling calls using the event-socket allows for great power and
flexibility.
As for connecting the FS servers you have several options. You can use
ACLs to allow the servers to call each other via SIP without needing
authentication. You could also use gateways to allow FS machines to do SIP
registrations with each other. (I suppose it depends on your exact needs,
but ACLs are pretty easy and clean.) Once you have FS machines able to dial
each other then it's pretty much a matter of configuring your dialplans to
route the calls. You can create conferences on the fly so that part is easy.
The tricky part will be controlling how to link the conferences together,
although FS has a handy API to add a conference member to an existing
conference by dialing out. (See
http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference)
Let's say you have two servers, FS_A and FS_B. Each server can dial the
other and each server has a conference running, named FS_A_Conf and
FS_B_Conf, respectively. You can have either conference dial the other
server. For example, from the FS_A command line:
conference FS_A_Conf dial
{originate_timeout=30}sofia/internal/3...@fs_b.ip.address FS_A_Conf
Where 3900 is an extension set up in FS_B's dialplan to route to FS_B_Conf
and FS_A_Conf are the caller ID number and name that FS_B should
receive from FS_A.
You still need to figure out other things like if you want the ability to
break apart the two conferences after they've been connected together,
etc. This is definitely doable but you'll need to handle the call control
stuff some how.
HTH,
MC
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Re: [Freeswitch-users] Confused about conferences
You've thought through some of the difficult points, which is good. You're
right that the moderator can't have different controls (unless you're
controlling the conference yourself from outside, using, say, the event
socket).
Before I go further, I want to make sure I understand what you're proposing.
What you're essentially saying is that when the command to kick someone is
pressed the person should be transferred out of the conference, checked for
moderator status, asked whom to kick (if so), and then let back in while the
system kicks that person. The other commands would work similarly. Does this
sound like a correct summary?
If so, I think what you've got is mostly right, but I'm not sure it will
work, having not tried to do it that way myself (I went the event socket
route). In theory, it should look something like this:
In the scripts entering the conference, you'll need to save the current
conference number. Something like (in both cases):
action application=set data=conf-id=$1/
Extension doesn't actually check for anything, so you'll need to check
for kick with a condition. Try this:
extension name=kick
condition field=destination_number expression=^kick$/
condition field=$(i_am_moderator) expression=true
action application=play_and_get_digits data=3 3 3 7000 #
/ask-for-extension.wav /invalid.wav conf-user-id \d+
Followed by (I picked up the kick syntax from the wiki):
action application=conference data=$(conf-id) kick
$(conf-user-id)/
action application=conference
data=$(conf-id)@default+flags{mod}/
anti-action application=conference data=$(conf-id)@default/
/condition
/extension
In theory, then, you've transferred out with the 9 key, you check the
moderator flag you've made, and do stuff based on the conf-id and the data
you enter. Obviously, if this works, the moderator won't be able to hear
what's going on in the conference while he's entering stuff.
Bigger problem: the conf-user-id he enters has to be the member ID that FS
chose for the user he's trying to kick. That number will make sense if
you're following the system closely, but for someone who doesn't know FS
internals, it will be impossible to know unless you broadcast it to him
somehow. To my knowledge, there's no way to use any other identifier (like
caller-id) to kick them with.
BB
On Wed, Aug 12, 2009 at 10:56 AM, Alan Chandler
a...@chandlerfamily.org.ukwrote:
I have been reading all the docs about conferences I can find and am
getting somewhat confused. What I am trying to do is set up a dialplan
where I have subscribers with extensions in the 1xx range, and then to
set an ability to have a series of conference rooms for each subscriber
in the 21xx range where if the user enters is own conference he is the
moderator, but if not he is just a normal user.
I want to be able for the moderator to do things like mute or kick people.
So dialplan would probably have something like this in it
extension name=user_conference
condition field=destination_number
expression=^(2${caller_id_number})$
action application=set data=i_am_moderator=true
action application=answer/
action application=conference data=$...@default
+flags{moderator}/
/condition
condition field=destination_number expression=^(21\d{2})^
action application=set data=i_am_moderator=false
action application=answer/
action application=conference data=$...@default/
/condition
/extension
extension name=kick
condition field=i_am_moderator expression=true
action application=play_and_get_digits data=3 3 3 7000 #
/ask-for-extension.wav /invalid.wav conf-user-id \d+
WHAT GOES HERE???
/condition
/extension
in the conference.conf.xml file, I would change the caller controls to
include
caller-controls
group name=somekeys
control action=transfer digits=9 data=kick XML default/
/group
/caller-controls
My question (at the moment) is
In the WHAT GOES HERE place how do it
Kick extension ${conf-user-id} (DOES IT REQUIRE A SCRIPT TO CALL THE
Conference API?)
Re-Enter the moderator back into the conference
Re-Enter the ordinary user who happened to press 9 back into the conference
I am assuming I can't stop the non moderator getting the control - since
all users get the same controls.
--
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [Freeswitch-users] Confused about conferences
Whoops. All of my parens () should be curly braces {}. Wasn't paying
attention.
BB
On Wed, Aug 12, 2009 at 2:40 PM, Bradley Brashier bjbrash...@gmail.comwrote:
You've thought through some of the difficult points, which is good. You're
right that the moderator can't have different controls (unless you're
controlling the conference yourself from outside, using, say, the event
socket).
Before I go further, I want to make sure I understand what you're
proposing. What you're essentially saying is that when the command to kick
someone is pressed the person should be transferred out of the conference,
checked for moderator status, asked whom to kick (if so), and then let back
in while the system kicks that person. The other commands would work
similarly. Does this sound like a correct summary?
If so, I think what you've got is mostly right, but I'm not sure it will
work, having not tried to do it that way myself (I went the event socket
route). In theory, it should look something like this:
In the scripts entering the conference, you'll need to save the current
conference number. Something like (in both cases):
action application=set data=conf-id=$1/
Extension doesn't actually check for anything, so you'll need to check
for kick with a condition. Try this:
extension name=kick
condition field=destination_number expression=^kick$/
condition field=$(i_am_moderator) expression=true
action application=play_and_get_digits data=3 3 3 7000 #
/ask-for-extension.wav /invalid.wav conf-user-id \d+
Followed by (I picked up the kick syntax from the wiki):
action application=conference data=$(conf-id) kick
$(conf-user-id)/
action application=conference
data=$(conf-id)@default+flags{mod}/
anti-action application=conference data=$(conf-id)@default/
/condition
/extension
In theory, then, you've transferred out with the 9 key, you check the
moderator flag you've made, and do stuff based on the conf-id and the data
you enter. Obviously, if this works, the moderator won't be able to hear
what's going on in the conference while he's entering stuff.
Bigger problem: the conf-user-id he enters has to be the member ID that FS
chose for the user he's trying to kick. That number will make sense if
you're following the system closely, but for someone who doesn't know FS
internals, it will be impossible to know unless you broadcast it to him
somehow. To my knowledge, there's no way to use any other identifier (like
caller-id) to kick them with.
BB
On Wed, Aug 12, 2009 at 10:56 AM, Alan Chandler
a...@chandlerfamily.org.uk wrote:
I have been reading all the docs about conferences I can find and am
getting somewhat confused. What I am trying to do is set up a dialplan
where I have subscribers with extensions in the 1xx range, and then to
set an ability to have a series of conference rooms for each subscriber
in the 21xx range where if the user enters is own conference he is the
moderator, but if not he is just a normal user.
I want to be able for the moderator to do things like mute or kick people.
So dialplan would probably have something like this in it
extension name=user_conference
condition field=destination_number
expression=^(2${caller_id_number})$
action application=set data=i_am_moderator=true
action application=answer/
action application=conference data=$...@default
+flags{moderator}/
/condition
condition field=destination_number expression=^(21\d{2})^
action application=set data=i_am_moderator=false
action application=answer/
action application=conference data=$...@default/
/condition
/extension
extension name=kick
condition field=i_am_moderator expression=true
action application=play_and_get_digits data=3 3 3 7000 #
/ask-for-extension.wav /invalid.wav conf-user-id \d+
WHAT GOES HERE???
/condition
/extension
in the conference.conf.xml file, I would change the caller controls to
include
caller-controls
group name=somekeys
control action=transfer digits=9 data=kick XML default/
/group
/caller-controls
My question (at the moment) is
In the WHAT GOES HERE place how do it
Kick extension ${conf-user-id} (DOES IT REQUIRE A SCRIPT TO CALL THE
Conference API?)
Re-Enter the moderator back into the conference
Re-Enter the ordinary user who happened to press 9 back into the
conference
I am assuming I can't stop the non moderator getting the control - since
all users get the same controls.
--
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS
then probably we should check the current state and ignore the libpri event when already in progress with media. On Wed, Aug 12, 2009 at 5:10 PM, Michael Collins m...@freeswitch.org wrote: On Wed, Aug 12, 2009 at 4:01 PM, Brian West br...@freeswitch.org wrote: Yes you can get a progress after you get a progress with media ... I have seen it. Yes, you definitely can and I believe that some of the PRI specs suggest that this is totally legal, even though it's kind of silly. -MC ___ 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 -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Confused about conferences
Bradley Brashier wrote:
...
Before I go further, I want to make sure I understand what you're
proposing. What you're essentially saying is that when the command to
kick someone is pressed the person should be transferred out of the
conference, checked for moderator status, asked whom to kick (if so),
and then let back in while the system kicks that person. The other
commands would work similarly. Does this sound like a correct summary?
Absolutely what I am trying to do.
...
Followed by (I picked up the kick syntax from the wiki):
action application=conference data=$(conf-id) kick
$(conf-user-id)/
This is a significant new fact for me. What you seem to be doing is
calling the commands referenced in the conference api here
http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
by using application=conference and then the data string as the second
part of the command. Am I correct in the assumption that you can do this.
Bigger problem: the conf-user-id he enters has to be the member ID that
FS chose for the user he's trying to kick. That number will make sense
if you're following the system closely, but for someone who doesn't know
FS internals, it will be impossible to know unless you broadcast it to
him somehow. To my knowledge, there's no way to use any other identifier
(like caller-id) to kick them with.
If my assumption about the use of the API above is correct, then
couldn't I do something like
action application=set data=member-list=${conference(${conf-id}
list)}/
action application=execute_extension data=do-kick/
and the figure out a regular expression to get participant-id and
caller-id out of the resultant string.
Or is this not how its done?
--
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [Freeswitch-users] Confused about conferences
This is a significant new fact for me. What you seem to be doing is
calling the commands referenced in the conference api here
http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
by using application=conference and then the data string as the second
part of the command. Am I correct in the assumption that you can do this.
I agree that that's what it looks like. What I don't know is if it works. I
got this example from the page
http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom. I never did
exactly what you're trying, and never tried using the API in this fashion.
action application=set data=member-list=${conference(${conf-id}
list)}/
action application=execute_extension data=do-kick/
and the figure out a regular expression to get participant-id and
caller-id out of the resultant string.
Or is this not how its done?
Even assuming that our understanding on the kick command is correct, and
there's an accessible API here, I don't believe the above would actually
populate your variable with the data that you want. I don't think it
actually issues the command with the set application. Instead, you'd need
to do something like
action application=conference data=${conf-id} list/
and then you'd have to capture the output somehow, which I also don't
believe is possible.
I'm hardly a master of XML, though, or of dialplans, so feel free to try it
if you want. Just make sure you're logging everything so you can watch it
all unfold.
BB
On Wed, Aug 12, 2009 at 3:33 PM, Alan Chandler
a...@chandlerfamily.org.ukwrote:
Bradley Brashier wrote:
...
Before I go further, I want to make sure I understand what you're
proposing. What you're essentially saying is that when the command to
kick someone is pressed the person should be transferred out of the
conference, checked for moderator status, asked whom to kick (if so),
and then let back in while the system kicks that person. The other
commands would work similarly. Does this sound like a correct summary?
Absolutely what I am trying to do.
...
Followed by (I picked up the kick syntax from the wiki):
action application=conference data=$(conf-id) kick
$(conf-user-id)/
This is a significant new fact for me. What you seem to be doing is
calling the commands referenced in the conference api here
http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
by using application=conference and then the data string as the second
part of the command. Am I correct in the assumption that you can do this.
Bigger problem: the conf-user-id he enters has to be the member ID that
FS chose for the user he's trying to kick. That number will make sense
if you're following the system closely, but for someone who doesn't know
FS internals, it will be impossible to know unless you broadcast it to
him somehow. To my knowledge, there's no way to use any other identifier
(like caller-id) to kick them with.
If my assumption about the use of the API above is correct, then
couldn't I do something like
action application=set data=member-list=${conference(${conf-id}
list)}/
action application=execute_extension data=do-kick/
and the figure out a regular expression to get participant-id and
caller-id out of the resultant string.
Or is this not how its done?
--
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS
Well you really can't ignore it... it happens with our ISDN stack too. Thats what the VETO handles. /b On Aug 12, 2009, at 5:28 PM, Moises Silva wrote: then probably we should check the current state and ignore the libpri event when already in progress with media. ___ 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 sharing conference between
On Wed, Aug 12, 2009 at 6:10 PM, Tina Martinez t...@a2unlimited.com wrote: Michael, Thanks for the welcome, and for the response to my question. The call control and dynamic setup of conferences I have working (pretty cool stuff). The tricky part, as you said, is linking the servers together. Basically, what I need to do is establish a connection that will not be dependent on a live person being on the call. And I would prefer to avoid having to register actual phone extensions for every server -- and for every conference call. I apologize if I'm slow, but I'm new to working an application like this. No worries. :) In your example, you stated, 3900 is an extension set up in FS_B's dialplan, does this extension have to be a live person (or soft-phone connected using an auto-answer mechanism)? or can I setup something where there is not a phone actually connected? No actual telephone is needed. Here's an example dialplan snippet that you could drop right into conf/dialplan/default/ in a new file. (I prefer to put my own custom dialplan entries into a separate file instead of editing default.xml) include extension name=Sample Conference at 3900 condition field=destination_number expression=^3900$ action application=conference data=c...@fs_b/ /condition /extension /include Put the above into conf/dialplan/default/01_ConfB.xml on the FS_B server. (You can make a similar file on the FS_A server or any other server if you'd like.) You will also need to create a public extension which will route the inbound calls appropriately. (If that doesn't make any sense right now then don't worry, just do it. :) Put the following into a file named conf/dialplan/public/01_ConfB.xml: include extension name=Sample Conference at 3900 condition field=destination_number expression=^3900$ action application=transfer data=3900 XML default/ /condition /extension /include (Again, you can do the same on all of your servers - this will allow all servers to receive calls and route them to x3900.) Now that you've got 3900 set up you can test it. Press F6 (or type reloadxml) at the CLI for FS_B. Then, have a phone that is registered to FS_B make a call to 3900. It should be alone in the conference. Now you'll need to set up some sort of dialplan routing to call from FS_A to FS_B, unless you have a SIP phone registered to FS_A that can dial a SIP URI. The SIP URI is: sip:3...@fs_b.ip.address. For kicks, let's add a simple dialplan extension on FS_A that allows you to dial 23900 to get to FS_B's 3900 extension. Put this into conf/dialplan/default/01_Dial_ConfB.xml on FS_A: include extension name=Sample Conference at 3900 condition field=destination_number expression=^2(3900)$ action application=bridge data=sofia/internal/$...@fs_b.ip.address / /condition /extension /include Save, and do the reloadxml (or F6) thing on FS_A CLI. Now on FS_A you can dial 23900 and it will ring right into 3900 on FS_B so that the phone registered at FS_A is in the conference on FS_B. Got it? Have fun tinkering and let us know how it all goes. -MC Right now I'm able to place a call from one machine to another, but I'm calling an X-Lite phone on the second server. Also, if it is possible to call a virtual extension, I have no problem incorporating application logic that would clean-up the orphaned conferences on all machines when a conference call is complete. I'm more concerned with being able to setup the links quickly and in an elegant fashion. - T ___ 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] Setting max inbound for UA
Is there a way to limit the number of calls a UA can receive in the FS configs? I'm doing some testing with XLite as the UA, and can not figure out how to keep line 2 from answering if line 1 is in use. THanks. -str ___ 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] Setting max inbound for UA
Check out mod_limit... Other wise you'll have to look specifically at the UA you are trying to use, some like polycom and sipura offer a way to disable call waiting Remember with SIP there is no such thing as a line, its a SESSION and you can have as many sessions as the software allows (and most software doesn't put sane limits based on CPU/RAM/Bandwidth etc) From: String Larson strin...@gmail.com Reply-To: freeswitch-users@lists.freeswitch.org Date: Wed, 12 Aug 2009 19:42:02 -0500 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Setting max inbound for UA Is there a way to limit the number of calls a UA can receive in the FS configs? I'm doing some testing with XLite as the UA, and can not figure out how to keep line 2 from answering if line 1 is in use. THanks. -str ___ 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] OpenZAP/Sangoma in Windows
I have been testing analog support under Windows with good results so far. I am waiting on another driver fix to solve some small problems with the api. I have been running this code for 6 weeks or so now under a home test environment(light call traffic) and the reliabilty has been fine - no errors or other abnormalites. The openzap for windows code is very simular to the others now(with the new LibSangoma) so we should be in pretty good shape. I would love to hear more from you regarding your testing of PRI support under windows. -Jeff Moises Silva wrote: On Tue, Aug 11, 2009 at 2:16 AM, Peter Olsson peter.ols...@visionutveckling.se wrote: Hi, I'm trying to evaluate the OpenZAP/Sangoma-support in Windows, using PRI E1 connections. Thanks for testing this :-) I have been meaning to install FreeSWITCH on Windows but just could not find the time. 1. Has anyone tested it in Windows at all? I know the build-files for Visual Studio has only been checked in for a couple of months, so that's why I'm asking. The drivers have been tested quite well but not using FreeSWITCH. 2. Does anyone have any directions how to configure the driver within Windows? Should I use BitStream or HDLC, and how should the channel groups be configured? You should use HDLC for the D-channel and Bitstream for the B-Channels. Typically you would create 2 groups, one with channels 1-23 and the other with just channel 24. The first group would work in bitstream and TDM_CHAN_VOICE_API operational mode and the second in HDLC/API mode. You can find me in #openzap, #freeswitch or #freeswitch-dev as moy if you have more questions. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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://n2.nabble.com/OpenZAP-Sangoma-in-Windows-tp3422060p3435444.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
