July 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Week in Review (Master Branch) July 4th-July 11th
by Ken Rice 13 Jul '15

13 Jul '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1O2b2vR FreeSWITCH Week in Review (Master Branch) July 4th-July 11th Hello, again. This passed week in the FreeSWITCH master branch we had 54 commits. We had a bunch of new commits for features this week! Those pull requests are really helping to get things reviewed and accepted. Keep it up community! 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-7780 Add new channel variable max_session_transfers. If set, this variable is used to count the number of session transfers allowed instead of the max_forwards variable. If not set, the existing behavior is preserved. FS-7783 Add channel variable for capturing DTMF input when using play_and_get_digits when the response does not match FS-7772 [mod_opus] Add functionality to keep FEC enabled on the encoder by modifying the bitrate if packet loss changes (Opus codec specific behaviour). FS-7799 [mod_png] Add API command uuid_write_png FS-7801 [mod_opus] Added support to set CBR mode FS-7685 [mod_say_nl] Fix Dutch numbers pronunciation FS-7198 Add coma separated values and reverse ranges for time-of-day and day-of-week matches FS-7809 [mod_opus] Added 60 ms ptime for Opus at 8 khz ( opus@8000h@60i ) FS-7405 [mod_dialplan_xml] Fix condition regex=”all” to work with time conditions FS-7819 [mod_opus] Restore bitrate (if there’s no more packet loss) and added step for 60 ms FS-7773 [mod_sofia] Adding additional transfer events when the fire-transfer-events=true profile parameter is set FS-7820 FreeSWITCH automated unit test and micro benchmark framework Improvements in build system, cross platform support, and packaging: FS-7628 [mod_erlang_event] Removed unused variables causing a compilation error FS-7776 Add mod_kazoo to packaging The following bugs were squashed: FS-7778 [mod_sofia] Fixed a bug causing a SQL statement to fail because of a double quote instead of a single quote FS-7754 [freetdm] Fixed a bug relating to single digit dial-regex with analog devices FS-7785 [mod_opus] Fix for invalid ptime 30 ms for opus@8000h . Replaced 30 ms with 40 ms. FS-7762 [mod_av] Handle buffer allocation failures of large buffers And, this passed week in the FreeSWITCH 1.4 branch we had no new commits merged in from master.

1 0

FreeSWITCH Week in Review (Master Branch) June 27th-July 3rd
by Ken Rice 13 Jul '15

13 Jul '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1KIMIBl FreeSWITCH Week in Review (Master Branch) June 27th-July 3rd Hello, again. This passed week in the FreeSWITCH master branch we had 49 commits. This week we had a bunch of new features with most of them being helpful little improvements, but we also had two new modules merged in! The 2600hz guys added mod_kazoo and William King merged in mod_smpp. You can find out more about mod_smpp by going here. And, the 2600hz patches are all slated to be merged in by the 1.6 release. 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-7732 Continue recording with uuid_transfer FS-7752 [mod_rayo] Increase maximum number of elements from 30 to 1024 to allow adhearsion to create large grammars to navigate IVR menus. FS-7750 [mod_commands] Allow for uuid_setvar to handle arrays FS-7758 [mod_loopback] Emit an event if a loopback bowout occurs FS-7759 [mod_sofia] Added the channel variable ignore_completed_elsewhere to suppress setting the completed elsewhere cause FS-7771 Set a channel variable if the recording is terminated due to silence hits FS-7760 Added xml fetch for channels to externally support nightmare transfer depends on channel-xml-fetch-on-nightmare-transfer profile param (default is disabled) FS-7730 [mod_smpp] Added mod_smpp as an event handler module and fixed the default configs to provided sample load option for mod_sms and mod_smpp FS-7774 Add mod_kazoo Improvements in build system, cross platform support, and packaging: OPENZAP-238 [freetdm] Fix some GSM compilation errors and do a bit of code cleanup OPENZAP-237 [freetdm] Use __func__ instead of __FUNCTION__ to comply with c99 in gcc 5.1 The following bugs were squashed: FS-7734 [mod_nibblebill] Fixed a deadlock FS-7726 Fixed a bug with recording a video session on DTMF command FS-7721 Fixed a segfault caused when using session:recordFile() and session:unsetInputCallback in a lua script FS-7429 [mod_curl] Fixed to output valid json FS-7746 [mod_verto] Fixed a device permission error in verto client FS-7753 [mod_local_stream] Fixed some glitching and freezing video when using hold/unhold FS-7761 [core] Fix shutdown races running api commands during shutdown FS-7767 [mod_sofia] Fixed a segfault caused by invalid arguments to sip_dig FS-7744 [mod_conference] Fixed a bug causing the first user’s video stream to stop when another verto user calls the conference FS-7486 [mod_sofia] Fixed the handling of queued requests FS-7775 [mod_conference] Fix threading issue causing stuck worker threads FS-7777 [mod_imagick] Fixed a regression causing a segfault when playing png & pdf in conference And, this passed week in the FreeSWITCH 1.4 branch we had 2 commits merged in from master. FS-7486 [mod_sofia] Fixed the handling of queued requests FS-7750 [mod_commands] Set uuid_setvar to handle arrays

4 3

Force codec on A leg
by stefano 13 Jul '15

13 Jul '15

Hello, I have this scenario: A = user B = my FreeSwitch C = Dialogic IVR HMP A leg place a call with codec in sdp in this order g729 PCMU PCMA B reroute the call to C C answer with codec PCMU, PCMA I'd like to remove codec PCMU,PCMA from originator (A leg) and use only g729 offered and let the FS make the transcoding vs C. I have used on the sip_profile: on my dialplan: <action application="export" data="nolocal:absolute_codec_string=PCMA,PCMU"/> before outdialing to the IVR How can I discard the use of PCMU,PCMA from the A leg? Thanks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Force-codec-on-A-leg-tp759617… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 0

Eavesdrop a call.
by Igor Olhovskiy 13 Jul '15

13 Jul '15

Hi! I'm using Freeswitch 1.4.20 (git b95362f 2015-07-03 16:42:15Z 64bit)) (stable one) and trying to get eavesdropping. But no luck. Main scheme: DID - > extension (210). 210 answers eavesdropping from extension 110. While giving a command *eavesdrop <channel-uuid> * I've got log full of messages *2015-07-09 13:29:00.325667 [ERR] switch_core_io.c:1531 Write Buffer 0 bytes Failed!* and silence on 110 extension While giving a command *userspy 210@${domain_name}* or *userspy <channel-uuid>* I've got music-on-hold on 110, no errors on log. In both cases used <channel-uuid> of extension 210 (answered one) Is this a bug or I'm doing something wrong? -- Best regards, Igor

3 4

Send a alert by email when freeswith server is down
by Manish Talwar 13 Jul '15

13 Jul '15

Hi, Is there any way to send an alert by email or SMS to someone whenever FreeSwitch is down (either by manually shutdown or server crashed/restart). Please let me know the steps for its configuration if possible. Thanks, Regards, Manish Talwar

4 3

LAN being blocked
by Joli Martinez 12 Jul '15

12 Jul '15

Hello, I have installed Freeswitch and fusionPBX. My LAN network is 192.168.21.0/24. PBX is 250. I am trying to register a phone to Freeswitch. When I do tcpdump on the LAN interface on port 5060, I see traffic hitting the box from the phone, but on the Freeswitch CLI I see nothing. I am assuming an ACL is blocking the traffic. Where can I allow the LAN side to register? I have tried the acl.xml and added the following and it still doesn't work. After I did this I reloaded the ACL and also restarted FS. <list name="lan" default="allow">  <node type="allow" cidr="192.168.21.0/24"/> </list>  <list name="domains" default="deny">  <node type="allow" domain="$${domain}"/> <node type="allow" cidr="192.168.21.1/32"/>  <node type="allow" cidr="192.168.21.0/24"/>

3 6

freeswitch not listening on 5060
by Joli Martinez 12 Jul '15

12 Jul '15

Hello, I installed Freeswitch with fusionPBX. I am able to get my public SIP trunk to register on port 5080 and when I do a netstat I see it listening on port 5080. I do not see it listening on port 5060 therefore I cant get any internal extensions to resister. I have checked the internal.xml and it is set to the internal sip port which in the vars.xml is set to 5060. What am I missing to get Freeswitch to listen on both 5080 for external and 5060 for internal. The box has two interfaces, but only one is supposed to be used for freeswitch. Also There is an SBC infront of Freeswich. Thanks,

2 1

socket application execution
by Babak Yakhchali 11 Jul '15

11 Jul '15

Hi On what conditions socket application execution ends in dialplan and the next application execution starts? for example in this dialplan when info is executed? <extension name="socket_test"> <condition field="destination_number" expression="^999$"> <action application="socket" data="172.16.90.1:8084 async full"/> <action application="info"/> </condition> </extension> thanks

2 1

E1 cards
by Nikolay Zaytsev 11 Jul '15

11 Jul '15

Hi everyone) Could you advise me please which card is the best choice for using with Freeswitch? I need the card with 2 E1 ports. So far I have been thinking about Sangoma A102DE, but maybe you will advise me something else) Thank you for your time, Best regards, Nikolay

1 0

calling intercept from event socket
by Babak Yakhchali 11 Jul '15

11 Jul '15

Hi I'm using event socket to do something like freeswitch callcenter module. members(aleg) call and are connected to socket application, this is my dialplan: <extension name="socket"> <condition field="destination_number" expression="^999$"> <action application="set" data="socket_resume=true"/> <action application="socket" data="172.16.90.1:8084 async full"/> <action application="info"/> </condition> </extension> these are executed using socket: connect myevents set park_after_bridge=true answer playback local_stream://moh now a new channel is originated to agent(bleg) and on answer I'm calling intercept on aleg from socket to bridge it to agent channel. if I set socket_resume=true in dialplan, after intercept both legs are stuck (dead silent). if I do not set socket_resume=true legs are bridged but execution continues through dialplan (I placed info to test this). problem is if socket_resume=false and agent transfers the call to 999 during the call (same socket app) call hangsup. but if socket_resume=true intercept is not working as expected but transfer is done. how can I solve this? thanks

2 4