[Freeswitch-users] Connecting FreeSWITCH with Asterisk
Dear All, I have tried to connect the FreeSWITCH with Asterisk I have followed steps which is provided in the following link, http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk I have tried to call 2000 from FreeSWITCH, but I have received the following message in Asterisk console NOTICE[6346]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user Velusamy sip:velus...@192.168.6.94sip%3avelus...@192.168.6.94 ;tag=69Q648H9NjrSK I have read Using Authentication topic in the link, But I did understand that topic.. They have mentioned HOSTNAME.DOMAIN.COM in that topic. Which hostname I have to specify here? Please help me Regards, Velusamy.K ___ 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] ESL problem
Dear all, In the previous post, I got the information that using event outbound socket we can implement the IVR and also see the example in libs/esl/perl/server2.pl. I have seen it and understood the flow of the script.But when I was running that script it tells the following error. Can't locate loadable object for module ESL in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at ESL.pm line 11 Compilation failed in require at server2.pl line 1. Then I checked the 11th line in the ESL.pm that line is (bootstrap ESL;) But in perl directory there is no directory called ESL. What would be the issue?. Is ESL necessary is necessary for implementing IVR using event outbound socket? -- Regards, Thangappan.M ___ 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] ESL problem
Make certain that you've built both libesl and the Perl mod. Change directory into /usr/src/freeswitch.trunk/libs/esl (or whatever your path is to libs/esl) and do these commands: make make perlmod Then give it another shot. -MC On Mon, Jul 27, 2009 at 11:16 PM, Thangappan.M thangappan...@gmail.comwrote: Dear all, In the previous post, I got the information that using event outbound socket we can implement the IVR and also see the example in libs/esl/perl/server2.pl. I have seen it and understood the flow of the script.But when I was running that script it tells the following error. Can't locate loadable object for module ESL in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at ESL.pm line 11 Compilation failed in require at server2.pl line 1. Then I checked the 11th line in the ESL.pm that line is (bootstrap ESL;) But in perl directory there is no directory called ESL. What would be the issue?. Is ESL necessary is necessary for implementing IVR using event outbound socket? -- Regards, Thangappan.M ___ 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] Connecting FreeSWITCH with Asterisk
Before you go any further, could you let us know what you are trying to accomplish? What kind of calls need to go to/from Asterisk/FreeSWITCH? Do you require some sort of authentication? Are the FS and Ast machines on the same LAN? It might help for you to pastebin the output from the FS CLI when you make a test call to the Asterisk box as that might give you some clue as to what isn't working. -MC On Mon, Jul 27, 2009 at 11:10 PM, velusamy velu velu.techni...@gmail.comwrote: Dear All, I have tried to connect the FreeSWITCH with Asterisk I have followed steps which is provided in the following link, http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk I have tried to call 2000 from FreeSWITCH, but I have received the following message in Asterisk console NOTICE[6346]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user Velusamy sip:velus...@192.168.6.94sip%3avelus...@192.168.6.94 ;tag=69Q648H9NjrSK I have read Using Authentication topic in the link, But I did understand that topic.. They have mentioned HOSTNAME.DOMAIN.COM in that topic. Which hostname I have to specify here? Please help me Regards, Velusamy.K ___ 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] Connecting FreeSWITCH with Asterisk
Dear, I am just testing that how to connect FreeSWITCH with Asterisk. I don't want any sort of authentication. Yes, the FS and Asterisk are on the same LAN.. My intention is that When I call an extension from FS, the dial plan should bridge a user in Asterisk.. Please give some suggestions... Thanks in Advance. Regards, Velusamy. On Tue, Jul 28, 2009 at 12:32 PM, Michael Collins m...@freeswitch.orgwrote: Before you go any further, could you let us know what you are trying to accomplish? What kind of calls need to go to/from Asterisk/FreeSWITCH? Do you require some sort of authentication? Are the FS and Ast machines on the same LAN? It might help for you to pastebin the output from the FS CLI when you make a test call to the Asterisk box as that might give you some clue as to what isn't working. -MC On Mon, Jul 27, 2009 at 11:10 PM, velusamy velu velu.techni...@gmail.comwrote: Dear All, I have tried to connect the FreeSWITCH with Asterisk I have followed steps which is provided in the following link, http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk I have tried to call 2000 from FreeSWITCH, but I have received the following message in Asterisk console NOTICE[6346]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user Velusamy sip:velus...@192.168.6.94sip%3avelus...@192.168.6.94 ;tag=69Q648H9NjrSK I have read Using Authentication topic in the link, But I did understand that topic.. They have mentioned HOSTNAME.DOMAIN.COM in that topic. Which hostname I have to specify here? Please help me Regards, Velusamy.K ___ 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] ESL problem
Don't forget you need to install libs/esl/perl/ESL.so and libs/esl/ perl/ESL.pm into your system perl library path. /b On Jul 28, 2009, at 1:53 AM, Michael Collins wrote: Make certain that you've built both libesl and the Perl mod. Change directory into /usr/src/freeswitch.trunk/libs/esl (or whatever your path is to libs/esl) and do these commands: make make perlmod Then give it another shot. -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] SIP trunk (PBX - FreeSwitch) routing problem.
Hello brian, It was not exactly at the bottom but before X-PRE-PROCESS cmd=include data=default/*.xml/ I tried to put it higher in the dialplan but it still doesn't work (with the same error). Thanks for your help. Brian West a écrit : I have to guess that you put this at the bottom of the default.xml? /b On Jul 27, 2009, at 10:58 AM, julien wrote: And in the dialplan default.xml : extension name=pbxlyon condition field=destination_number expression=300 action application=set data=hangup_after_bridge=true/ action application=set data=continue_on_fail=true/ action application=bridge data=sofia/gateway/pbxlyon/300/ action application=hangup/ /condition /extension ___ 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] SIP instant messaging presence signaling doesn't work.
Hi everybody, I intend to use Freeswitch with two Ekiga Softphones. SIP Instant messaging works between the two softphones but SIP presence signaling is not managed by the softphones. I try to use other softphones (QuteCom and SIPCommunicator) and it is the same. I have the following error in my FreeSwitch console: [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete subscriptions for failed notify Is there any special configuration for SIP instant messaging presence? Thanks. G.C. Hi everybody, I intend to use Freeswitch with two Ekiga Softphones. SIP Instant messaging works between the two softphones but SIP presence signaling is not managed by the softphones. I try to use other softphones (QuteCom and SIPCommunicator) and it is the same. I have the following error in my FreeSwitch console: [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete subscriptions for failed notify Is there any special configuration for SIP instant messaging presence? Thanks. G.C. ___ 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] SIP trunk (PBX - FreeSwitch) routing problem.
Well press F8 and increase the debug level.. then try again you'll prob. see that its not finding it NOR matching it anywhere in your dialplan. /b On Jul 28, 2009, at 4:32 AM, julien wrote: Hello brian, It was not exactly at the bottom but before X-PRE-PROCESS cmd=include data=default/*.xml/ I tried to put it higher in the dialplan but it still doesn't work (with the same error). Thanks for your help. ___ 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] SIP trunk (PBX - FreeSwitch) routing problem.
julien jgonza...@sqli.com wrote: It was not exactly at the bottom but before X-PRE-PROCESS cmd=include data=default/*.xml/ Why not put it in the default directory, from which it will be included by the above line? If necessary, you could comment out any entries in default.xml that might be matched first. I've debugged this kind of problem before, and the best solution has always been to read the logs carefully to see which extensions matched (or didn't match). Also, if necessary, check out freeswitch/log/freeswitch.xml.fsxml to see where your extension ends up in the final dial plan. ___ 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] SIP instant messaging presence signaling doesn't work.
You must turn on the option to manage presence in the sip profile. Mike On Jul 28, 2009, at 5:43 AM, Gregory Charles gregory.char...@sogeti.com wrote: Hi everybody, I intend to use Freeswitch with two Ekiga Softphones. SIP Instant messaging works between the two softphones but SIP presence signaling is not managed by the softphones. I try to use other softphones (QuteCom and SIPCommunicator) and it is the same. I have the following error in my FreeSwitch console: [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete subscriptions for failed notify Is there any special configuration for SIP instant messaging presence? Thanks. G.C. Hi everybody, I intend to use Freeswitch with two Ekiga Softphones. SIP Instant messaging works between the two softphones but SIP presence signaling is not managed by the softphones. I try to use other softphones (QuteCom and SIPCommunicator) and it is the same. I have the following error in my FreeSwitch console: [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete subscriptions for failed notify Is there any special configuration for SIP instant messaging presence? Thanks. G.C. ___ 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] SIP trunk (PBX - FreeSwitch) routing problem.
Thanks for the tip Brian. It seems that the extension matches successfully in the dialplan (PASS, instead of FAIL for all other entries of the dialplan) : Dialplan: sofia/internal/[EMAIL PROTECTED] parsing [default-pbxlyon] continue=false Dialplan: sofia/internal/[EMAIL PROTECTED] Regex (PASS) [pbxlyon] destination_number(300) =~ /300/ break=on-false But it leads nowhere. After the match the connection to the PBX fails : 2009-07-28 16:16:43.963836 [NOTICE] switch_channel.c:602 New Channel sofia/external/300 [46fca878-7b81-11de-a9c2-0f49fee5280a] 2009-07-28 16:16:43.963836 [DEBUG] mod_sofia.c:2751 (sofia/external/300) State Change CS_NEW - CS_INIT 2009-07-28 16:16:43.963836 [DEBUG] switch_core_session.c:933 Send signal sofia/external/300 [BREAK] 2009-07-28 16:16:43.973759 [DEBUG] switch_core_state_machine.c:397 (sofia/external/300) Running State Change CS_INIT 2009-07-28 16:16:43.973759 [DEBUG] switch_core_state_machine.c:480 (sofia/external/300) State INIT 2009-07-28 16:16:43.973759 [DEBUG] mod_sofia.c:83 sofia/external/300 SOFIA INIT 2009-07-28 16:16:43.975221 [DEBUG] mod_sofia.c:111 (sofia/external/300) State Change CS_INIT - CS_ROUTING 2009-07-28 16:16:43.975221 [DEBUG] switch_core_session.c:933 Send signal sofia/external/300 [BREAK] 2009-07-28 16:16:43.975221 [DEBUG] sofia.c:3215 Channel sofia/external/300 entering state [calling][0] 2009-07-28 16:16:43.975221 [DEBUG] switch_core_state_machine.c:480 (sofia/external/300) State INIT going to sleep 2009-07-28 16:16:43.975221 [DEBUG] switch_core_state_machine.c:397 (sofia/external/300) Running State Change CS_ROUTING 2009-07-28 16:16:43.975221 [DEBUG] switch_core_state_machine.c:483 (sofia/external/300) State ROUTING 2009-07-28 16:16:43.975221 [DEBUG] mod_sofia.c:130 sofia/external/300 SOFIA ROUTING 2009-07-28 16:16:43.975221 [DEBUG] switch_ivr_originate.c:63 (sofia/external/300) State Change CS_ROUTING - CS_CONSUME_MEDIA 2009-07-28 16:16:43.975221 [DEBUG] switch_core_session.c:933 Send signal sofia/external/300 [BREAK] 2009-07-28 16:16:43.976763 [DEBUG] switch_core_state_machine.c:483 (sofia/external/300) State ROUTING going to sleep 2009-07-28 16:16:43.976763 [DEBUG] switch_core_state_machine.c:397 (sofia/external/300) Running State Change CS_CONSUME_MEDIA 2009-07-28 16:16:43.976763 [DEBUG] switch_core_state_machine.c:502 (sofia/external/300) State CONSUME_MEDIA 2009-07-28 16:16:44.82786 [DEBUG] sofia.c:3215 Channel sofia/external/300 entering state [terminated][404] 2009-07-28 16:16:44.82786 [NOTICE] sofia.c:3775 Hangup sofia/external/300 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] It looks to me that it's more a problem from the gateway than from the dialplan? Don't you think so? Do you think the way I defined my gateway is good for a connexion to a PBX ? Thanks for your replies Brian and Jason. Brian West a écrit : Well press F8 and increase the debug level.. then try again you'll prob. see that its not finding it NOR matching it anywhere in your dialplan. /b On Jul 28, 2009, at 4:32 AM, julien wrote: Hello brian, It was not exactly at the bottom but before X-PRE-PROCESS cmd=include data=default/*.xml/ I tried to put it higher in the dialplan but it still doesn't work (with the same error). Thanks for your help. ___ 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] SIP trunk (PBX - FreeSwitch) routing problem.
The remote end said 404 /b On Jul 28, 2009, at 10:00 AM, julien wrote: 2009-07-28 16:16:44.82786 [DEBUG] sofia.c:3215 Channel sofia/external/300 entering state [terminated][404] ___ 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] originate in dialplan
Hello,
Please tell me, how can I execute originate new call and uuid_bridge in dial
plan.
I tried to make like thise:
action application=originate data=user/$${destination_end_point}
playback(${hold_music})/
action application=originate data=user/$${destination_end_point},
playback($${hold_music})/
action application=uuid_bridge data=$${uuid} $${client_uuid}/
result:
[ERR] switch_core_session.c:1239
switch_core_session_execute_application() Invalid Application originate
[ERR] switch_core_session.c:1239
switch_core_session_execute_application() Invalid Application uuid_bridge
Best regards.
vkozak___
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[Freeswitch-users] CELT codec code number
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, today I finylly got a working Ekiga Softphone version which is able to use high quality celt codec with FS :) On my way to get it work with FS I found that Ekiga currently uses codec code 95 in SDP while FS uses 114. Changing FS to 95 made it works so I'm now able to listen to my PBX-based MP3-Player on Windows Desktop instead of using Ubuntu. veerrry cool work of FS team ! Concerning the codec code 95, 114 or whatever I found the link below, which states that every codec code between 96 and 127 is OK but it seems they prefer 97 ... http://tools.ietf.org/html/draft-valin-celt-rtp-profile-02#section-5 You can get windows Ekiga 3.2.5 (with celt 0.5.1) here: http://wwwuser.gwdg.de/~mrickma/ekiga/ekiga-setup-3.2.5-release.exe regards Helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKbyPL4tZeNddg3dwRAiDMAKChqgWeirYklgra5nN7NGwZSpK6wQCgjYox Q/okubHauhgjtoiogzFM9mI= =Ml4q -END PGP SIGNATURE- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] originate in dialplan
What exactly are you trying to accomplish with this dialplan entry? That
will help us answer your question.
-MC
2009/7/28 Kozak Vladimir vko...@abisoft.spb.ru
Hello,
Please tell me, how can I execute originate new call and uuid_bridge in
dial plan.
I tried to make like thise:
action application=originate
data=user/$${destination_end_point} playback(${hold_music})/
action application=originate
data=user/$${destination_end_point}, playback($${hold_music})/
action application=uuid_bridge data=$${uuid} $${client_uuid}/
result:
[ERR] switch_core_session.c:1239
switch_core_session_execute_application() Invalid Application originate
[ERR] switch_core_session.c:1239
switch_core_session_execute_application() Invalid Application uuid_bridge
Best regards.
vkozak
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Re: [Freeswitch-users] CELT codec code number
On Jul 28, 2009, at 11:14 AM, Helmut Kuper wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, today I finylly got a working Ekiga Softphone version which is able to use high quality celt codec with FS :) On my way to get it work with FS I found that Ekiga currently uses codec code 95 in SDP while FS uses 114. Changing FS to 95 made it works so I'm now able to listen to my PBX-based MP3-Player on Windows Desktop instead of using Ubuntu. It should work if they use different codec numbers I suspect we are sending on 114 and receiving on 95 which is what should take place. This is one of those areas most people fail to implement properly. We send the remote our RTP map they send us theirs... Can you get a packet capture of this taking place so I can verify who is at fault? /b veerrry cool work of FS team ! Concerning the codec code 95, 114 or whatever I found the link below, which states that every codec code between 96 and 127 is OK but it seems they prefer 97 ... http://tools.ietf.org/html/draft-valin-celt-rtp-profile-02#section-5 You can get windows Ekiga 3.2.5 (with celt 0.5.1) here: http://wwwuser.gwdg.de/~mrickma/ekiga/ekiga-setup-3.2.5-release.exe ___ 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] originate in dialplan
Здравствуйте, Kozak. Try, for example action application="bridge" data=""/ Вы писали 28 июля 2009 г., 18:16:45: Hello, Please tell me, how can I execute originate new call and uuid_bridge in dial plan. I tried to make like thise: action application="originate" data=""/ action application="originate" data=""/ action application="uuid_bridge" data=""/ result: [ERR] switch_core_session.c:1239 switch_core_session_execute_application() Invalid Application originate [ERR] switch_core_session.c:1239 switch_core_session_execute_application() Invalid Application uuid_bridge Best regards. vkozak -- С уважением, Evgeniy mailto:zolo...@altron.ua ___ 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] CELT codec code number
using 95 is wrong. That is not part of the dynamic range for unassigned codecs. This needs to be fixed on their side. MIke On Jul 28, 2009, at 12:23 PM, Brian West wrote: On Jul 28, 2009, at 11:14 AM, Helmut Kuper wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, today I finylly got a working Ekiga Softphone version which is able to use high quality celt codec with FS :) On my way to get it work with FS I found that Ekiga currently uses codec code 95 in SDP while FS uses 114. Changing FS to 95 made it works so I'm now able to listen to my PBX-based MP3-Player on Windows Desktop instead of using Ubuntu. It should work if they use different codec numbers I suspect we are sending on 114 and receiving on 95 which is what should take place. This is one of those areas most people fail to implement properly. We send the remote our RTP map they send us theirs... Can you get a packet capture of this taking place so I can verify who is at fault? /b veerrry cool work of FS team ! Concerning the codec code 95, 114 or whatever I found the link below, which states that every codec code between 96 and 127 is OK but it seems they prefer 97 ... http://tools.ietf.org/html/draft-valin-celt-rtp-profile-02#section-5 You can get windows Ekiga 3.2.5 (with celt 0.5.1) here: http://wwwuser.gwdg.de/~mrickma/ekiga/ekiga-setup-3.2.5-release.exe ___ 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] AbsoluteTimeout: The FreeSWITCH way
Hello everyone, I need to set a maximum call duration. What is the current recommended way to implement this in FreeSWITCH? I'm looking for something similar to AbsoluteTimeout() in Asterisk. Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.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] AbsoluteTimeout: The FreeSWITCH way
action application=sched_hangup data=+600/ On Tue, Jul 28, 2009 at 7:28 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Hello everyone, I need to set a maximum call duration. What is the current recommended way to implement this in FreeSWITCH? I'm looking for something similar to AbsoluteTimeout() in Asterisk. Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] AbsoluteTimeout: The FreeSWITCH way
What needs to happen at the end of the timeout? In any case you can use the sched_XXX APIs: sched_api sched_transfer sched_hangup You can get fancy or just hangup up on the call after X number of seconds... :) -MC On Tue, Jul 28, 2009 at 10:28 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Hello everyone, I need to set a maximum call duration. What is the current recommended way to implement this in FreeSWITCH? I'm looking for something similar to AbsoluteTimeout() in Asterisk. Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] AbsoluteTimeout: The FreeSWITCH way
You can also schedule a playback then a hangup, what comes after the ! is the hangup cause. sched_broadcast,Schedule a broadcast in the future,[+]time path [aleg|bleg|both],mod_dptools action application=sched_broadcast data=+600 playback! normal_clearing::/path/to/file / Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca Am 28-Jul-09 um 1:50 PM schrieb Michael Collins: What needs to happen at the end of the timeout? In any case you can use the sched_XXX APIs: sched_api sched_transfer sched_hangup You can get fancy or just hangup up on the call after X number of seconds... :) -MC On Tue, Jul 28, 2009 at 10:28 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Hello everyone, I need to set a maximum call duration. What is the current recommended way to implement this in FreeSWITCH? I'm looking for something similar to AbsoluteTimeout() in Asterisk. Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] AbsoluteTimeout: The FreeSWITCH way
also take a look at execute_on_answer if you want it to be scheduled from answer instead of from that point in the dialplan. Mike On Jul 28, 2009, at 1:48 PM, Saeed Ahmad wrote: action application=sched_hangup data=+600/ On Tue, Jul 28, 2009 at 7:28 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Hello everyone, I need to set a maximum call duration. What is the current recommended way to implement this in FreeSWITCH? I'm looking for something similar to AbsoluteTimeout() in Asterisk. Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] originate in dialplan
A call to B.
A place to musikOnHold.
I wont play short musik rington B before make bridge A with B.
If use API comands, it's work:
//1. A to musikOnHold.
sendmsg
call-command: execute
execute-app-name: execute_extension
execute-app-arg: hold_extension
//2. originate B and play rington
api originate B_END_POINT playback(MUSIK_PATH)
//3. bridge
api uuid_bridge A_UUID B_UUID
I wont joint point 2 and point 3 in dial plan.
Result API comands:
//1. A to musikOnHold.
sendmsg
call-command: execute
execute-app-name: execute_extension
execute-app-arg: hold_extension
//2. originate B and play rington to B and bridge A with B
sendmsg
call-command: execute
execute-app-name: execute_extension
execute-app-arg: xxx_extension
dialplan.xml
extension name=xxx_extension
condition field=destination_number expression=^xxx_extension$
???//originate B
???//play to B
???//bridge with A
/condition
/extension
- Original Message -
From: Michael Collins
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, July 28, 2009 8:07 PM
Subject: Re: [Freeswitch-users] originate in dialplan
What exactly are you trying to accomplish with this dialplan entry? That will
help us answer your question.
-MC
2009/7/28 Kozak Vladimir vko...@abisoft.spb.ru
Hello,
Please tell me, how can I execute originate new call and uuid_bridge in
dial plan.
I tried to make like thise:
action application=originate
data=user/$${destination_end_point} playback(${hold_music})/
action application=originate
data=user/$${destination_end_point}, playback($${hold_music})/
action application=uuid_bridge data=$${uuid} $${client_uuid}/
result:
[ERR] switch_core_session.c:1239
switch_core_session_execute_application() Invalid Application originate
[ERR] switch_core_session.c:1239
switch_core_session_execute_application() Invalid Application uuid_bridge
Best regards.
vkozak
___
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[Freeswitch-users] DTMF confusion
Hello, If I wanted a bridged call to a gateway to use inband DTMF for incoming recognition and outgoing generation I'm unclear on what to do because the wiki clearly states[1] not to use the start_dtmf and start_dtmf_generate together for cause of loops. Wouldn't it be technically possible to generate DTMF only on the outbound leg and recognize DTMF only on the inbound leg without interference? I assume I'm not understanding something correctly here - could somebody elaborate? The end goal for me is to detect the absense of telephone-event rtpmap and enable inband DTMF from a gateway. Thanks! - Jesse [1] http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_managed users?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I've just tried new mod_managed under Win32 and I get a weird behavior.
I try the example below:
public class DemoScript : IApiPlugin
{
public void Execute(ApiContext context)
{
context.Stream.Write(string.Format(DemoScripts executed with
args '{0}' and event type {1}.,
context.Arguments, context.Event == null ? none :
context.Event.GetEventType()));
}
public void ExecuteBackground(ApiBackgroundContext context)
{
Log.WriteLine(LogLevel.Notice, DemoScripts on a background
thread #({0}), with args '{1}'.,
System.Threading.Thread.CurrentThread.ManagedThreadId,
context.Arguments);
}
}
It's just like the ApiDemo from Demo.cs
So When I copy DemoScript.csx to managed dir the console log is:
freeswi...@zwierko-laptop Loading
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx from
domain DemoScripts.csx_3
2009-07-28 20:57:16.71 [INFO] switch_cpp.cpp:1130 Compiling
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx
2009-07-28 20:57:16.97 [ERR] switch_cpp.cpp:1130 There were 1 errors
compiling
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx.
2009-07-28 20:57:16.97 [ERR] switch_cpp.cpp:1130 CS5001: Program
'c:\Users\Zwierko\AppData\Local\Temp\fc8jnlir.exe' does not contain a
static 'Main' method
suitable for an entry point
Adding
public static void Main()
{
}
solves the issue. Is this how it's supposed to work?
Another strange thing is that when I compile this class to DLL (release)
it does not work at all...
freeswi...@zwierko-laptop Loading
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\FSScripts.dll from
domain FSScripts.dll_5
freeswi...@zwierko-laptop
freeswi...@zwierko-laptop managed DemoScript 111
API CALL [managed(DemoScript 111)] output:
2009-07-28 21:13:03.542000 [ERR] switch_cpp.cpp:1130 API plugin
DemoScript not found.
2009-07-28 21:13:03.542000 [ERR] mod_managed.cpp:393 Execute failed for
DemoScript 111 (unknown module or exception).
And another issue with scripts. I use script code as example:
public class ScriptDemo
{
static void Main()
{
switch (FreeSWITCH.Script.ContextType)
{
case ScriptContextType.Api:
{
var ctx = FreeSWITCH.Script.GetApiContext();
ctx.Stream.Write(Script executing as API with args:
+ ctx.Arguments);
break;
}
case ScriptContextType.ApiBackground:
{
var ctx = FreeSWITCH.Script.GetApiBackgroundContext();
Log.WriteLine(LogLevel.Notice, Executing as
APIBackground with args: + ctx.Arguments);
break;
}
case ScriptContextType.App:
{
var ctx = FreeSWITCH.Script.GetAppContext();
Log.WriteLine(LogLevel.Notice, Executing as App
with args: + ctx.Arguments);
break;
}
}
}
}
console log is as follows:
freeswi...@zwierko-laptop Loading
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx from
domain Script.csx_8
2009-07-28 21:19:36.289000 [INFO] switch_cpp.cpp:1130 Compiling
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx
2009-07-28 21:19:36.438000 [INFO] switch_cpp.cpp:1130 File
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx compiled
successfully.
2009-07-28 21:19:36.451000 [ERR] switch_cpp.cpp:1130 Entry point:
ScriptDemo.Main is not public. This may cause errors with Mono.
2009-07-28 21:19:36.458000 [NOTICE] switch_cpp.cpp:1130 Loaded App
Script.csx, aliases 'Script.csx', into domain Script.csx_8.
2009-07-28 21:19:36.459000 [NOTICE] switch_cpp.cpp:1130 Loaded Api
Script.csx, aliases 'Script.csx', into domain Script.csx_8.
2009-07-28 21:19:36.459000 [INFO] switch_cpp.cpp:1130 Finished loading
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx into
domain Script.csx_8.
managed ScriptDemo 111
2009-07-28 21:19:53.452000 [ERR] switch_cpp.cpp:1130 API plugin
ScriptDemo not found.
API CALL [managed(ScriptDemo 111)] output:
2009-07-28 21:19:53.452000 [ERR] mod_managed.cpp:393 Execute failed for
ScriptDemo 111 (unknown module or exception).
Again, am I doing something wrong in here?
Thanks, for any feedback
Lukasz Zwierko
Michael Giagnocavo wrote:
Ah, that’s embarrassing. I added them and tried building FreeSWITCH.Managed
from svn and it worked fine now. (I’ll kick off a new complete build in a
minute.)
-Michael
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Diego Toro
Sent: Sunday, July 26, 2009 8:47 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_managed users?
Hi Michael,
Thank you for your job with mod_managed, I get lastest version with
[Freeswitch-users] Using tone_detect application
Hi, I'm doing some tests between 2 FS to understand how the tone_detect application works. I'm trying to detect a SIT tone, but I can't seem to detect the 3 tones. I only get the first activated tone. If I have all tones activated - - - I detect only the first tone of my wav file. If I have 2nd segment tones and up activated - - - I detect only the second tone of my wav file. If I have the 3rd segment tones activated - - - I detect only the third tone of my wav file. I see that tone_detect can detect all 3 tones, but never at the same time. Does anyone have an idea of what I could be doing wrong ? Thanks ! I'm using FreeSWITCH Version 1.0.trunk (14397). My dialplan looks like this : extension name=incoming condition field=destination_number expression=^(514[0-9][0-9][0-9][0-9][0-9][0-9][0-9])$ action application=answer/ action application=tone_detect data=SIT1LO 913.8 w +25000 set SIT1LO=true/ !--1st segment low -- action application=tone_detect data=SIT1HI 985.2 w +25000 set SIT1HI=true/ !--1st segment high -- action application=tone_detect data=SIT2LO 1370.6 w +25000 set SIT2LO=true/ !--2nd segment low -- action application=tone_detect data=SIT2HI 1428.5 w +25000 set SIT2HI=true/ !--2nd segment high -- action application=tone_detect data=SIT3LO 1776.7 w +25000 set SIT3LO=true/ !--3rd segment low -- action application=tone_detect data=SITITU1 950 w +25000 set SITITU1=true/ !--2nd segment low -- action application=tone_detect data=SITITU2 1400 w +25000 set SITITU2=true/ !--2nd segment high -- action application=tone_detect data=SITITU3 1800 w +25000 set SITITU3=true/ !--3rd segment low -- action application=bridge data=sofia/internal/$...@192.168.100.9/ /condition /extension ___ 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] Distortion on approx first 200ms of G722prompts on DECT based CPE
Hi, [Thanks for the advice Anthony] I tried send_silence_when_idle=true and restarted, but did not notice any change/improvement. But I had limited time to test, so will need to test more thoroughly with this CPE. An additional test was to configure the following media path scenarios: A: Policom-FS-DECT CPE B: Policom-DECT CPE (media not via FS) [The only change was to add param name=inbound-bypass-media value=true in sip profile for B/) In scenario A, whenever the Policom's VAD kicked in, it resulted in the first 200ms of the restarting audio being distorted in the DECT CPE. In scenario B, no problems when the Policom's VAD kicked in. I also addressed the issue to the CPE vendor yesterday, who responded today : We have observed the RTP-stream and found the following : - the RTP-stream is completely interrupted in the pause between the announcements - no RTP-packets are send at all during these pauses - jitter buffer runs empty and our device will automatically mute the connection when no packets comes in - last RTP-packets before pauses don't contain the info about comfort-noise We would like to ask you the following : - please check if it's possible to send dummy-packets during pause instead of sending no packets at all In scanning the wiki on the topic of CNG, I found at http://wiki.freeswitch.org/wiki/VAD_and_CNG : In FreeSWITCH the CNG options select whether or not FreeSWITCH will generate CN RTP packets. suppress-cng sofia profile option and suppress_cng channel variable used to set of this setting. When both sides are supporting RFC3389 (they agree in SDP message exchange, rtpmap:13), FreeSWITCH will send CN packets. Note: Allowing CNG in FreeSWITCH does not mean it will generate any comfort noise into the media channel. In case one of the parties in bridge do not handle VAD and asynchronous RTP media, there should be an issue as the one might think hearing perfect silence and might think the connection has been dropped. Another example is when on one side is Asterisk or CallWeaver. For handling these endpoints, there has been added (r9543) a new channel variable: bridge_generate_comfort_noise which will generate fake audio So the options here seem to be : a) Get FS to send CNG packet(s) before going into 'pauses'. From the vendor's analysis they are not seeing this when testing (FYI, these observations were made calling into vmail). Could this be because the CPE is perhaps not supporting RFC3389 - FS did not see the rtpmap:13 in the SDP ? b) Make sure FS keeps sending packets during pauses and silence. I am not clear on the difference between the 'send_silence_when_idle=true' and 'bridge_generate_comfort_noise=true' options. Ideally I would still want to leverage VAD, but then need the CNG messages to be forwarded in scenarios where I have media passing through FS on a call between two customers and when a customer is interacting with vmail (or other IVR type application). Any advise appreciated. Best Regards Keith From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 28 July 2009 02:06 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Distortion on approx first 200ms of G722prompts on DECT based CPE you can set the global var send_silence_when_idle=true in vars.xml On Mon, Jul 27, 2009 at 4:23 PM, Keith Laaks wrote: Hi All, I am testing a range of G722 capable DECT based CPE. With one range, I have noticed that the first 200ms or so of each separate prompt file being played back is played out distorted from the DECT handset. When having a normal conversation, the quality is excellent, but when accessing your vmail, all the individual audio files making up the menu choices exhibit the distortion, which is pretty annoying. The same unit using G729, alaw or ulaw works 100%. I wonder if anybody else has uncounted this issue? My guess at this point - There may be a short break in the RTP between the separate files being played out by FS that makes up any menu. During this time the DECT handset's AGC probably goes to MAX amplification (as its not receiving any input during the short break in RTP). Then, when the RTP returns at the start of the next file, the AGC boosts the audio into clipping zone and takes 200ms to dampen down back to normal good levels. Looks like in these devices the G722 encode/decode is actually done in the DECT handset and not the voip-base unit. Is there any parameter that can be set in FS to ensure that the RTP keeps flowing, sending 'silence' between prompts ? Would be interesting to validate the above 'guess'. Best Regards Keith ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Re: [Freeswitch-users] mod_managed users?
Hello Lukasz,
Thanks for testing mod_managed. I apologize for the problems you've
encountered, and I'll try to sort them out for you.
A few things first:
- Scripting support: This is made to allow true scripts, as invoked
as an EXE - similar to the Lua and spidermonkey support. So, without a Main(),
it won't compile as an EXE. If you aren't using it as a script, then an empty
Main method will work fine.
- Entry points must be public for Mono. I'll update the demo code to
make sure that Main is public. This is a bug in Mono's lightweight code
generation -- it won't skip the JIT access checks.
As to the main problem of your DLL not working, can you send me the full source
code, or all the logging output from loading it? Try managedreload my.dll to
reload the DLL and see how it is registering them. It should output something
like Registering API FullName with Aliases fullname, shortname.
Thanks,
Michael
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Lukasz
Zwierko
Sent: Tuesday, July 28, 2009 1:26 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_managed users?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I've just tried new mod_managed under Win32 and I get a weird behavior.
I try the example below:
public class DemoScript : IApiPlugin
{
public void Execute(ApiContext context)
{
context.Stream.Write(string.Format(DemoScripts executed with
args '{0}' and event type {1}.,
context.Arguments, context.Event == null ? none :
context.Event.GetEventType()));
}
public void ExecuteBackground(ApiBackgroundContext context)
{
Log.WriteLine(LogLevel.Notice, DemoScripts on a background
thread #({0}), with args '{1}'.,
System.Threading.Thread.CurrentThread.ManagedThreadId,
context.Arguments);
}
}
It's just like the ApiDemo from Demo.cs
So When I copy DemoScript.csx to managed dir the console log is:
freeswi...@zwierko-laptop Loading
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx from
domain DemoScripts.csx_3
2009-07-28 20:57:16.71 [INFO] switch_cpp.cpp:1130 Compiling
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx
2009-07-28 20:57:16.97 [ERR] switch_cpp.cpp:1130 There were 1 errors
compiling
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx.
2009-07-28 20:57:16.97 [ERR] switch_cpp.cpp:1130 CS5001: Program
'c:\Users\Zwierko\AppData\Local\Temp\fc8jnlir.exe' does not contain a
static 'Main' method
suitable for an entry point
Adding
public static void Main()
{
}
solves the issue. Is this how it's supposed to work?
Another strange thing is that when I compile this class to DLL (release)
it does not work at all...
freeswi...@zwierko-laptop Loading
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\FSScripts.dll from
domain FSScripts.dll_5
freeswi...@zwierko-laptop
freeswi...@zwierko-laptop managed DemoScript 111
API CALL [managed(DemoScript 111)] output:
2009-07-28 21:13:03.542000 [ERR] switch_cpp.cpp:1130 API plugin
DemoScript not found.
2009-07-28 21:13:03.542000 [ERR] mod_managed.cpp:393 Execute failed for
DemoScript 111 (unknown module or exception).
And another issue with scripts. I use script code as example:
public class ScriptDemo
{
static void Main()
{
switch (FreeSWITCH.Script.ContextType)
{
case ScriptContextType.Api:
{
var ctx = FreeSWITCH.Script.GetApiContext();
ctx.Stream.Write(Script executing as API with args:
+ ctx.Arguments);
break;
}
case ScriptContextType.ApiBackground:
{
var ctx = FreeSWITCH.Script.GetApiBackgroundContext();
Log.WriteLine(LogLevel.Notice, Executing as
APIBackground with args: + ctx.Arguments);
break;
}
case ScriptContextType.App:
{
var ctx = FreeSWITCH.Script.GetAppContext();
Log.WriteLine(LogLevel.Notice, Executing as App
with args: + ctx.Arguments);
break;
}
}
}
}
console log is as follows:
freeswi...@zwierko-laptop Loading
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx from
domain Script.csx_8
2009-07-28 21:19:36.289000 [INFO] switch_cpp.cpp:1130 Compiling
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx
2009-07-28 21:19:36.438000 [INFO] switch_cpp.cpp:1130 File
c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx compiled
successfully.
2009-07-28 21:19:36.451000 [ERR] switch_cpp.cpp:1130 Entry point:
ScriptDemo.Main is not public. This may cause errors with Mono.
2009-07-28 21:19:36.458000 [NOTICE] switch_cpp.cpp:1130 Loaded App
Script.csx, aliases
Re: [Freeswitch-users] Distortion on approx first 200ms of G722prompts on DECT based CPE
Have a look at the sip traffic. I believe in the default configuration that FreeSWITCH will use negotiated CNG (payload 13) if the other end supplies it. The description you got from your vendor is entirely accurate but they are supposed to handle this situation. When we stop sending RTP for a duration longer than the RTP period we send the marker bit in the next packet which tells the remote end that we have resumed after a pause so it can clear its jitter buffer. If the SDP contains negotiation for CNG we will send it per the RFC. Some devices assume everyone just automatically does CNG w/o negotiating it if your vendor is such a device you may need a special param custom coded into FS to force the CNG even when it's not negotiated. On Tue, Jul 28, 2009 at 3:54 PM, Keith Laaks kei...@voxtelecom.co.zawrote: Hi, [Thanks for the advice Anthony] I tried “send_silence_when_idle=true “ and restarted, but did not notice any change/improvement. But I had limited time to test, so will need to test more thoroughly with this CPE. An additional test was to configure the following media path scenarios: A: Policom-FS-DECT CPE B: Policom-DECT CPE (media not via FS) [The only change was to add param name=inbound-bypass-media value=true in sip profile for B/) In scenario A, whenever the Policom’s VAD kicked in, it resulted in the first 200ms of the restarting audio being distorted in the DECT CPE. In scenario B, no problems when the Policom’s VAD kicked in. I also addressed the issue to the CPE vendor yesterday, who responded today : “We have observed the RTP-stream and found the following : - the RTP-stream is completely interrupted in the pause between the announcements - no RTP-packets are send at all during these pauses - jitter buffer runs empty and our device will automatically mute the connection when no packets comes in - last RTP-packets before pauses *don't contain the info about comfort-noise* We would like to ask you the following : - please check if it's possible to send dummy-packets during pause instead of sending no packets at all “ In scanning the wiki on the topic of CNG, I found at http://wiki.freeswitch.org/wiki/VAD_and_CNG : “In FreeSWITCH the CNG options select whether or not FreeSWITCH will generate CN RTP packets. suppress-cng sofia profile option and suppress_cng channel variable used to set of this setting. When both sides are supporting RFC3389 (they agree in SDP message exchange, rtpmap:13), FreeSWITCH will send CN packets. Note: Allowing CNG in FreeSWITCH does not mean it will generate any comfort noise into the media channel. In case one of the parties in bridge do not handle VAD and asynchronous RTP media, there should be an issue as the one might think hearing perfect silence and might think the connection has been dropped. Another example is when on one side is Asterisk or CallWeaver. For handling these endpoints, there has been added (r9543) a new channel variable: bridge_generate_comfort_noise which will generate fake audio” So the options here seem to be : a) Get FS to send CNG packet(s) before going into ‘pauses’. From the vendor’s analysis they are not seeing this when testing (FYI, these observations were made calling into vmail). Could this be because the CPE is perhaps not supporting RFC3389 – FS did not see the rtpmap:13 in the SDP ? b) Make sure FS keeps sending packets during pauses and silence. I am not clear on the difference between the ‘send_silence_when_idle=true’ and ‘bridge_generate_comfort_noise=true’ options. Ideally I would still want to leverage VAD, but then need the CNG messages to be forwarded in scenarios where I have media passing through FS on a call between two customers and when a customer is interacting with vmail (or other IVR type application). Any advise appreciated. Best Regards Keith *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* 28 July 2009 02:06 *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Distortion on approx first 200ms of G722prompts on DECT based CPE you can set the global var send_silence_when_idle=true in vars.xml On Mon, Jul 27, 2009 at 4:23 PM, Keith Laaks wrote: Hi All, I am testing a range of G722 capable DECT based CPE. With one range, I have noticed that the first 200ms or so of each separate prompt file being played back is played out distorted from the DECT handset. When having a normal conversation, the quality is excellent, but when accessing your vmail, all the individual audio files making up the menu choices exhibit the distortion, which is pretty annoying. The same unit using G729, alaw or ulaw works 100%. I wonder if anybody else has uncounted this issue? My guess at this point – There may be a short break in the RTP between the
[Freeswitch-users] ext-ext transfer from gateway
Hello, I'm looking for a way to bridge 2 external PBX devices without keeping the FS gateway occupied during the conversation. My configuration: FS is registered as a sip-endpoint (G - gateway) in a regular PBX, gateway configured on FS side, inbound and outbound working fine. G is limited to 2 calls max. Scenario: 1. PBX device A dials G and routed by FS dialplan (e.g. music on hold player). 2. session transferred to PBX device B via same gateway G. (Transfer is made from a module loaded to FS with switch_ivr_session_transfer function.) 3. B answers, A and B are talking. Problem: calls are connected via gateway G and blocking it until A or B hooks on. Question: is there a way to avoid the problem with some gateway config tune? Or if not, is it possible to make call transfer using the gateway like a regular device or is there any other way to make FS leave the call alone? -- View this message in context: http://www.nabble.com/ext-ext-transfer-from-gateway-tp24708822p24708822.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] CELT codec code number
Brian West wrote: I totally missed this at first... but 95 wouldn't dynamically work because its not 96-127 /b The way to CELT plugin registers should be ok, the code is based on the Speex plugin. Opal tries to assign a number in the dynamic range first and if nothing is free in that area, goes backwards starting at 95 to find an empty spot (opal/mediafmt.cxx - OpalMediaFormatInternal::OpalMediaFormatInternal()). Maybe some of the other codecs is registering a lot of formats without the shared flag set and using up all dynamic IDs, or there's a problem somewhere in the ID handling 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] how to enable ESL for ruby?
Hi Brian, Sorry responding late. I still cannot get this work, can you take a look? http://pastebin.freeswitch.org/9877 Everything works fine on Linux but not on my MAC. I have the default ruby framework and port install on /opt/local/bin/ruby, however, even I changed the Makefile to use the default ruby framework, it just doesn't work though the ESL.so compiled successfully. Where I'm wrong? Thanks. 2009/5/27 Brian West br...@freeswitch.org On May 26, 2009, at 11:20 AM, dujinfang wrote: Thanks Brain. Got ESL.so, however on my Mac it is #include ruby.h instead of ruby/ruby.h. Actually since we do -framework Ruby it should be ruby/ruby but I think the line above the -framework Ruby should be removed since you're doing i tthe Mac way. /b ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] how to enable ESL for ruby?
Just to mention, there is also a Ruby library here for FreeSWITCH, similar to ESL, it might interest you. http://code.rubyists.com/projects/fs http://github.com/bougyman/freeswitcher/tree/master http://blog.rubyists.com/2009/05/19/ruby-freeswitch-love On Wed, Jul 29, 2009 at 12:45 AM, Seven Du dujinf...@gmail.com wrote: Hi Brian, Sorry responding late. I still cannot get this work, can you take a look? http://pastebin.freeswitch.org/9877 Everything works fine on Linux but not on my MAC. I have the default ruby framework and port install on /opt/local/bin/ruby, however, even I changed the Makefile to use the default ruby framework, it just doesn't work though the ESL.so compiled successfully. Where I'm wrong? Thanks. 2009/5/27 Brian West br...@freeswitch.org On May 26, 2009, at 11:20 AM, dujinfang wrote: Thanks Brain. Got ESL.so, however on my Mac it is #include ruby.h instead of ruby/ruby.h. Actually since we do -framework Ruby it should be ruby/ruby but I think the line above the -framework Ruby should be removed since you're doing i tthe Mac way. /b ___ 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
