September 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Week in Review (Master Branch) September 5th-11th
by Ken Rice 14 Sep '15

14 Sep '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1FJl1lk FreeSWITCH Week in Review (Master Branch) September 5th-11th Hello, again. This past week in the FreeSWITCH master branch we had 35 commits. This week we had one feature: the handling of a=sendonly, a=sendrecv, a=recvonly to change who is sending video during a call. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-8053 Handle a=sendonly, a=sendrecv, a=recvonly to change who is sending video during a call Improvements in build system, cross platform support, and packaging: FS-7168 [Debian] Update packages so that FS core libraries are setup as runtime dependencies FS-7697 Chown the /etc/freeswitch/tls directory so that the freeswitch user can have read/write for TLS certificate generation FS-7966 [Windows] Explore use of nuget for wix build dependency FS-7967 [build] SmartOS compatibility FS-1772 [mod_voicemail] Fixed a build error FS-8072 [Debian] Fixed a missed space The following bugs were squashed: FS-8130 Port video buffer to also support audio and remove original STFU jitter buffer, add some more resilience to video packet loss, add codec control mechanism for both call-specific debug and codec/call specific parameters, make opus function better in packet loss and latent situations, use new codec control prams to make jitter buffer lookahead FEC optionally enabled or disabled mid-call, and add a parameter to allow jitter buffer lookahead to be enabled. FS-8131 [mod_voicemail] Fixed issues with allowing an empty password change and then locking out the user FS-8136 [mod_h26x] Do not load passthru video codecs by default FS-8140 [mod_sofia] Fixed a user_name typo in sofia_handle_sip_i_invite FS-8142 Fixed a thread cache thread-safety and caching FS-8075 [mod_hiredis] Fix for failover when you pull power on redis, while redis clients under load test FS-8144 [mod_opus] Readability and code formatting cleanup FS-8126 [switch_core] Fixed the pruning of a media bug causing all media bugs on a session to be pruned FS-8143 [mod_rayo] Fixed a crash caused by client disconnecting from mod_rayo while a message is being delivered to that client. This is caused by the XMPP context’s JID -> XMPP stream mapping not being cleaned up on XMPP stream destruction. FS-8147 [event_handlers] Fixed the process spawning segfault FS-8149 [mod_xml_cdr] Fixed curl dependency in makefile And, this past week in the FreeSWITCH 1.4 branch we had 9 new commits merged in from master. And the FreeSWITCH 1.4.21 release is here! Go check it out! The following bugs were fixed: FS-8143 [mod_rayo] Fixed a crash caused by client disconnecting from mod_rayo while a message is being delivered to that client. This is caused by the XMPP context’s JID -> XMPP stream mapping not being cleaned up on XMPP stream destruction. FS-8127 [mod_conference] Fixed an audio issue caused by the codec not updating often enough when detecting rate change FS-8099 [mod_lua] Restored LUA dialplan ACTIONS functionality FS-8142 Fixed thread cache thread-safety and caching

1 0

Extract Channel Variables inline When using Lua script
by Jonathan Hunter 14 Sep '15

14 Sep '15

Hi Guys, Sorry for the post here but I just wanted to check if this was possible. I am using one large .lua script, and Im setting up channel variables before the call is connected, so for example I am setting in line; table.insert(xml, [[<action application="set" data="closedop=voicemail" inline="true"/>]]); table.insert(xml, [[<action application="set" data="closedopact=2000" inline="true"/>]]); This then terminates on a dialplan, where I would like to use the value of closedop and closedopact as an anti-action; <anti-action application="]]..closedop..[[" data="default ]]..h.Customer_Domain..[[ ]]..closedopact..[["/> I can see in the logs the value is set; Action set(closedop=voicemail) INLINE EXECUTE sofia/internal/07917190439(a)62.239.9.74 set(closedop=voicemail) 2015-09-14 20:41:47.828526 [DEBUG] mod_dptools.c:1435 sofia/internal/ 07917190438(a)62.239.9.74 *SET [closedop]=[voicemail]* Dialplan: sofia/internal/07917190439(a)62.239.9.74 Action set(closedopact=2000) INLINE EXECUTE sofia/internal/07917190439(a)62.239.9.74 set(closedopact=2000) 2015-09-14 20:41:47.828526 [DEBUG] mod_dptools.c:1435 sofia/internal/ 07917190438(a)62.239.9.74 *SET [closedopact]=[2000]* So I want to set the application to voicemail, and the extension to 2000. How can I extract this? I have tried using ${closedop}, and session:getvariables, however this is only suitable when session established I believe? Plus I'm doing this inline, so setting variables, then wanting to get them again before answer. Can anyone advise? Thanks Jon

1 0

mod_callcenter timeouts and on break
by Kamil Nigmatullin 14 Sep '15

14 Sep '15

Hello everybody, Could anybody help me understand the following thing. When agent didn't answer, lets say 3 times, system changes its status to "On break". Is there any timer that sends agent back to status "Available" or should it be done manually only? Thank you -- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin

3 3

ACL for ESL issue ?
by Pavel 14 Sep '15

14 Sep '15

Hello, I was trying to enable esl connections from outside of fs host. To do so I've followed https://wiki.freeswitch.org/wiki/Mod_event_socket#Configuration and changed default event_socket.conf.xml from: <configuration name="event_socket.conf" description="Socket Client"> <settings> <param name="nat-map" value="false"/> <param name="listen-ip" value="::"/> <param name="listen-port" value="8021"/> <param name="password" value="ClueCon"/> </settings></configuration> to: <configuration name="event_socket.conf" description="Socket Client"> <settings> <param name="nat-map" value="false"/> <param name="listen-ip" value="0.0.0.0"/> <param name="listen-port" value="8021"/> <param name="password" value="ClueCon"/> </settings></configuration> and issued: reload mod_event_socket. Trying to telnet to fs host on port 8021 I observe: Content-Type: text/rude-rejectionContent-Length: 24 Access Denied, go away.Content-Type: text/disconnect-noticeContent-Length: 67 Disconnected, goodbye.See you at ClueCon! http://www.cluecon.com/Connection closed by foreign host. And in fs log i can see the following: mod_event_socket.c:2603 IP "someiphere" Rejected by acl "loopback.auto" But as far as I understand the event_socket.conf.xml doesn't mention any ACL set up against ESL connection ?Would someone please be so kind to point what am I missing ?Thanks.Regards,Pavel.

4 6

Using G729 between apps and SIP trunk
by Michael Nielsen 14 Sep '15

14 Sep '15

I've got FreeSWITCH running and works really well. I would like to use G729 as codec whenever possible. My SIP Trunk does not support G729, only G711. My apps which are connecting to my FreeSWITCH are supporting G729, license has been paid for these. Can I setup FreeSWITCH to use G729 whenever possible, and for SIP Trunks conversion happens, such as: app/G729 <> FS <> app/G729 app/G729 <> FS <> SIP Trunk/G711

4 5

blind transfer behaviour
by huseyin kalyoncu 14 Sep '15

14 Sep '15

Hello, When using blind transfer feature with the following setup: transferee<--->kamailio_proxy<--->freeswitch | | transferor<------------- | If the transferee does not accept the transfer with a 488 reply to REFER, Freeswitch is still ending the call with a BYE message. Is this behaviour correct? Attached is a call flow diagram. Highlighted row is showing the BYE message after 488 reply to REFER. Thanks Huseyin

1 0

Recommendations for TLS certificate?
by Michael Nielsen 14 Sep '15

14 Sep '15

Anyone who can recommend a TLS certificate provider for FS?

4 5

Freeswitch does not upgrade user_agent
by Volodymyr Fedorov 14 Sep '15

14 Sep '15

Greets to all, seems after upgrade to 1.4.21 freeswitch doesn't update field "user_agent" in sqlitedb table "sip_registrations" after phone re-registers. Someone experience the same problems ? Thanks! -- Best regards, Volodymyr

1 0

gsmopen problems
by Pete Kay 12 Sep '15

12 Sep '15

Hi I am using gsmopen to make outbound calls. I am expeiencing some problems. When I have 2 dongles running even they are correctly recognized with "gsm list" , I can only make one calls. The second interface call will get Disconnection Out Of Order error. Also, when the remote end hangs up, the channel is still not getting killed. Therefore, no more call can be made even after remote side hangs up and I need to uuid_kill the channel. Is this an issue or my configuration is wrong? Pete

3 7

Clean installation of FreeSWITCH with basic functionality
by Michael Nielsen 11 Sep '15

11 Sep '15

Dear all, I'm playing around with FreeSWITCH and everything plays very nicely. I'm using the freeswitch-meta-vanilla Debian package on Debian 7. I'm "only" using FreeSWITCH for calls between subscribers, voicemail and outgoing calls to a SIP gateway - so the vanilla installation seems rather "overkill" - especially for a production environment. I've tried to - on a clean Debian - install the freeswitch-meta-default including freeswitch-conf-minimal. It does install some things, but it seems as if some basic functionality is not working - such as fs_cli etc. What is the best way for a very clean FreeSWITCH similar to the vanilla installation, but without any users, dial plans and such...?

6 12