September 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

RFC 4662 Support in Freeswitch?
by Colton Conor 06 Sep '17

06 Sep '17

Does Freeswitch support RFC 4662 for BLF Resource list URI? Asterisk does: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30278158 They have a good overview page Kamillio does: https://kamailio.org/docs/modules/3.2.x/modules_k/rls.html Good info from Grandsteam about why to use this: http://www.grandstream.com/ sites/default/files/Resources/GXP21x0_Eventlist_BLF_Guide.pdf Same with Polycom: http://community.polycom.com/polycom/attachments/ polycom/VoIP/19112/1/Technical%20Brief%20-%20Busy%20Lamp%20Field.pdf Others have asked this question on the freeswitch users list, but responses were limited.

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leg B video
by Madovsky 05 Sep '17

05 Sep '17

Hi all, is there an easy way to know if the leg B has video or not? Thanks Franck

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B-leg transfer
by David Villasmil 05 Sep '17

05 Sep '17

I have this dialplan i'd like to do, let's see if i can explain it :) I have a call coming in on 5080, which i forward to another box. When B-leg is answered, I want to disconnect the A-leg and bridge B-leg to a conference. I'm trying with the following: <extension name="Move_b_leg_to_conference" continue="false"> <condition field="destination_number" expression="^9999$"> <action application="callcenter" data="support@default"/> </condition> </extension> <extension name="Forward_to_proxy" continue="true"> <condition field="destination_number" expression="^(.*)$"> <action application="export" data="nolocal:execute_on_answer=transfer -bleg 9999@public"/> <action application="bridge" data="sofia/gateway/proxy/$1"/> <action application="hangup" data="NORMAL_CLEARING"/> </condition> </extension> The idea is to answer the call with the second extension and execute_on_answer to transfer the b-leg to extension 9999 on public, which sends it over to the conference... But this doesn't seem to be working, the call seems to be sent to 9999@public: 2017-09-04 23:09:21.243247 [NOTICE] sofia.c:8159 Channel [sofia/external/4345412121] has been answered EXECUTE sofia/external/4345412121 transfer(-bleg 9999@public) 2017-09-04 23:09:21.243247 [DEBUG] switch_ivr.c:2165 (sofia/external/ 12345(a)1.2.3.4) State Change CS_EXECUTE -> CS_ROUTING 2017-09-04 23:09:21.243247 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/12345(a)1.2.3.4 to XML[9999@public@public] I've also tried doing uuid_transfer on the bleg like so: <extension name="Forward_to_proxy" continue="true"> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>  <action application="bridge" data="{hangup_after_bridge=true,origination_uuid=${new_uuid}}sofia/gateway/proxy/$1"/> <action application="uuid_kill" data="${uuid}"/> </condition> </extension> and also instead of uuid_transfer, i tried putting the bleg on hold like: <action application="export" data="nolocal:api_on_answer=uuid_hold ${new_uuid}"/> But is seems as soon as the aleg dies, the whole thing dies Help is much appreciated! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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FS with Homer. error: tport_logging: capture socket error: Protocol error
by Denys Pozniak 05 Sep '17

05 Sep '17

Hello! I have FreeSWITCH installation with Homer. But after some period of time FS stops sending traffic to Homer with error in console: freeswitch@freeswitch01> error: tport_logging: capture socket error: Protocol error error: tport_logging: capture socket error: Protocol error error: tport_logging: capture socket error: Protocol error Please help me to find the source of this issue. System has next parameters: freeswitch@freeswitch01> version FreeSWITCH Version 1.6.17~64bit ( 64bit) root@freeswitch:/# cat /etc/lsb-release DISTRIB_ID=Ubuntu DISTRIB_RELEASE=16.04 DISTRIB_CODENAME=xenial DISTRIB_DESCRIPTION="Ubuntu 16.04.2 LTS" root@freeswitch:/# cat sofia.conf.xml <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> <param name="debug-presence" value="0"/> <param name="capture-server" value="udp:1.2.3.4:9999"/> </global_settings> <profiles> <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml"/> </profiles> </configuration> And each profile has : <param name="sip-capture" value="yes"/> BR, Denys

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Playback of AMR-WB files.
by ha.ppy.neko 04 Sep '17

04 Sep '17

Hi, I have FreeSwitch 1.6.19 with mod_amrwb and mod_native_file installed. But when I try to playback amr-wb file I see the following error "cannot play or record native files with variable length data". Is it possible to play amr and amr-wb files with FS and if so how? -- Best regards, Ivan

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Channel Event Issue
by Freeswitch 04 Sep '17

04 Sep '17

Hi, since updating from FreeSWITCH 1.6.8 to FreeSWITCH 1.6.17 (Windows Server 2012 R2), the Event Socket Handler(inbound mode) sporadically stops sending Channel related Event Info. Once in this situation, the Event Socket Handler does not recover until either reset of the ESL connection by the Host that established the ESL connection or reload of mod_event_socket. Shortly before Event Socket connection drops it seems that there is a problem trying to close a channel. Example Log for failure case: Channel with uuid d9be476b-6a9d-4c96-95fb-5d8728bcc08e is never closed after leavig the conference. d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.399989 [NOTICE] switch_channel.c:1104 New Channel sofia/external/149@dialin [d9be476b-6a9d-4c96-95fb-5d8728bcc08e] d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.399989 [INFO] mod_dialplan_xml.c:637 Processing 149 <149>->4600 in context public d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.399989 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/149@dialin to XML[4600@default] d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.419958 [INFO] mod_dialplan_xml.c:637 Processing 149 <149>->4600 in context default d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.439966 [NOTICE] mod_dptools.c:1312 Channel [sofia/external/149@dialin] has been answered d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:55.940144 [NOTICE] sofia.c:1012 Hangup sofia/external/149@dialin [CS_EXECUTE] [NORMAL_CLEARING] d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:55.940144 [INFO] conference_loop.c:1469 Channel leaving conference, cause: NORMAL_CLEARING Example Log for regular case: Channel with uuid 982453be-a321-4825-a944-ecfffa715914 is closed correctly after leaving the conference): 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.340132 [NOTICE] switch_channel.c:1104 New Channel sofia/external/149@dialin [982453be-a321-4825-a944-ecfffa715914] 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.340132 [INFO] mod_dialplan_xml.c:637 Processing 149 <149>->4600 in context public 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.360126 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/149@dialin to XML[4600@default] 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.360126 [INFO] mod_dialplan_xml.c:637 Processing 149 <149>->4600 in context default 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.380148 [NOTICE] mod_dptools.c:1312 Channel [sofia/external/149@dialin] has been answered 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:56:11.900321 [NOTICE] sofia.c:1012 Hangup sofia/external/149@dialin [CS_EXECUTE] [NORMAL_CLEARING] 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:56:11.900321 [INFO] conference_loop.c:1469 Channel leaving conference, cause: NORMAL_CLEARING 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:56:11.900321 [NOTICE] switch_core_session.c:1683 Session 9744 (sofia/external/149@dialin) Ended 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:56:11.900321 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/149@dialin [CS_DESTROY] The channel with uuid d9be476b-6a9d-4c96-95fb-5d8728bcc08e will still be displayed at fs_cli on request of "show channels" freeswitch@1250-0110> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context d9be476b-6a9d-4c96-95fb-5d8728bcc08e,inbound,2017-08-29 10:59:35,1503997175,sofia/external/149@dialin,CS_EXECUTE,149,149,dialin,4600,playback,{playBackId=2573}us/announcement.wav,XML,default,L16,8000,128000,PCMA,8000,64000,,1250-0110,,,,ACTIVE,,,,,,,149,149,dialin,4600,XML,public 1 total. However, trying to retrieve the channel data for this channel e.g. "uuid_dump" fails. freeswitch@1250-0110> uuid_dump d9be476b-6a9d-4c96-95fb-5d8728bcc08e -ERR No such channel! Same issue exists with FreeSWITCH 1.6.19. We are continuously sending api commands to the channel. Could this be a race condition, receiving an api, while the Channel is trying to close ? Would bgapi help ? Why is it not an issue with FreeSWITCH 1.6.8 ? Regards, Tobias

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Stop packet radius
by Phonecall SRL - Nicolas Paleari 02 Sep '17

02 Sep '17

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detect_speech via Event Socket not working
by Henning Heedfeld 01 Sep '17

01 Sep '17

Hi, I cannot get the detect_speech app working. I fire the command as following: SendMsg 31ada607-aede-4073-86ff-4dcc318ee839 call-command: execute execute-app-name: detect_speech execute-app-arg: unimrcp:uni2 builtin:speech/transcribe This is the only result I see in fs_cli: 2017-09-01 17:15:26.518812 [DEBUG] switch_ivr.c:623 sofia/internal/1001@tst Command Execute detect_speech(unimrcp:uni2 builtin:speech/transcribe) EXECUTE sofia/internal/1001@tst detect_speech(unimrcp:uni2 builtin:speech/transcribe) When I use the play_and_detect_speech app instead, the MRCP connection is established immediately and everything is working fine. SendMsg 31ada607-aede-4073-86ff-4dcc318ee839 call-command: execute execute-app-name: play_and_detect_speech execute-app-arg: /tmp/beep.wav detect:unimrcp:uni2 builtin:speech/transcribe What am I doing wrong? Thanks for any hints, Henning

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Fwd: Call drops after 32 seconds with ACK Timeout
by Микита Нікітенко 01 Sep '17

01 Sep '17

Hello! I have server at digital ocean with a floating ip address and I have routed sip traffic to this floating ip. At this server installed kamailio which balancing between freeswitch servers. I have a problem that call drops after 32 seconds with error ACK Timeout. Traffic comming throug the kamailio proxy. kamailio.cfg - https://pastebin.com/x0CL3TEZ kamailio sip messages log - https://pastebin.com/wyT3cBbD freeswitch logs with sip debug - https://pastebin.com/zPysfQkV I'm struggling for a week and have no idea how to fix it. Thank you very much. Mykyta Nikitenko

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Inbound call - Dialed number on transfer
by Gregor Nanger 01 Sep '17

01 Sep '17

Can you please help me with best practice. I am transfering incoming call to extenstion via bridge to user/user@domain. But I want that sip client sees on which number (external DID) call was placed. Which variable should I set? Thank you, Gregor

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