August 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ClueCon Slides: Industrial Grade FreeSWITCH, Scaling, Balancing, High Availability, SIP, WebRTC
by Giovanni Maruzzelli 10 Aug '17

10 Aug '17

*Industrial Grade FreeSWITCH Scaling, Balancing and High Availability for SIP and WebRTC* Scaling your FreeSWITCH platform to serve a growing user base is a critical challenge. We'll go through the best techniques, practices, and implementations for Voice and Video Calls, Conferencing, WebRTC, SIP, Chatting, Presence and Instant Messaging Slides can also be downloaded from http://opentelecom.it/cluecon2017.pdf Enjoy ClueCon! -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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defunct processes created by bgsystem / system
by Dave Horton 10 Aug '17

10 Aug '17

I have the following action in my dialplan: <action application="bgsystem" data = "/opt/scripts/push_handler.php ${uuid} ${sip_contact_user} ${host}"/> each time that command gets executed I see another defunct freeswitch process created: root@cor03-san:/usr/local/freeswitch/log# ps -ef | grep freeswitch root 23010 1 2 06:37 ? 00:00:25 /usr/local/freeswitch/bin/freeswitch -nc -nonat root 23209 23010 0 06:44 ? 00:00:00 [freeswitch] <defunct> root 23242 23010 0 06:45 ? 00:00:00 [freeswitch] <defunct> root 23300 23010 0 06:48 ? 00:00:00 [freeswitch] <defunct> I tried changing 'bgsystem' to 'system' and reloaded the dialplan, but it continued to drop defunct processes. I’ve entered Jira FS-10572 as it seems like this is a bug, but I am wondering if there is a workaround of some kind?

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Persist outbound event socket after channel is destroyed
by Sean Ingham 09 Aug '17

09 Aug '17

I'm using the socket application in the dialplan to establish an outbound event socket to my app. I use the async and full flags to expose full api functionality. I currently use myevents to filter only events related to the current channel, I also use the linger command to ensure the socket stays up until all channel events have been sent to my app (after which FS then tears down the socket). However, what I'm trying to do is prevent FS closing the socket at all, and always leave it up to my app to close the socket when it's finished with it. Is this possible with ESL in outbound mode? In other words, I want the socket to linger longer :) Sean.

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Watch ClueCon live on FaceBook
by Anthony Minessale 09 Aug '17

09 Aug '17

http://facebook.com/freeswitch -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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2017-08-09 06:06:29.494827 [CRIT] switch_core_session.c:1755 LUKE: I'm hit, but not bad.
by Jurijs Ivolga 09 Aug '17

09 Aug '17

Hi, I'm constantly getting following error after around 1 week running freeswitch on quite a low amount of traffic and freeswitch stops to replying to any traffic: 7fcf13e5-4529-464c-87af-67dca8a2eca8 2017-08-09 06:06:06.454889 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/xxxx@myip [7fcf13e5-4529-464c-87af-67dca8a2eca8] 2017-08-09 06:06:06.454889 [CRIT] switch_core_session.c:1799 Thread Failure! 83f02a66-630e-4f59-8716-12a107478059 2017-08-09 06:06:29.494827 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/xxxx@myip [83f02a66-630e-4f59-8716-12a107478059] 2017-08-09 06:06:29.494827 [CRIT] switch_core_session.c:1799 Thread Failure! 2017-08-09 06:06:29.494827 [CRIT] switch_core_session.c:1755 LUKE: I'm hit, but not bad. 2017-08-09 06:06:29.494827 [CRIT] switch_core_session.c:1756 LUKE'S VOICE: Artoo, see what you can do with it. Hang on back there.... Green laserfire moves past the beeping little robot as his head turns. After a few beeps and a twist of his mechanical arm, Artoo reduces the max sessions to 101 thus, saving the switch from certain doom. fs_cli stops to work and freeswitch replies with: SIP/2.0 503 Maximum Calls In Progress Cpu usage is 0. Any hints? Jurijs

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Eavesdrop problems
by Tor Petterson 09 Aug '17

09 Aug '17

Hi I think I've run into a bug using eavesdrop. The scenario is that I setup a webRTC channel which I want to be able to eavesdrop on several calls in succession. I do this by starting eavesdrop then when I switch to eavesdrop on another call, I park the webRTC call before starting eavesdrop again. My problem is that I can only do this a few times before I lose audio on the webRTC call. When the problem occurs I get no entries in the log when I call the eavesdrop application. I tried to do some debugging on this and it seems that the channel thread gets stuck somewhere. Does anyone know a workaround or have an idea where in the code to look for the problem?

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Why do inbound calls receive "invalid number"?
by Nate 09 Aug '17

09 Aug '17

Good day/evening everyone, Calls from my mobile are showing up in the logs, but they aren't ringing the extension I have configured for Destination. I've searched, purchased (and been reading) the eBook Freeswitch 1.8, but am still a bit perplexed by what the issue is. Can anyone please assist with interpreting of the following log? Many many thanks!!! [WARNING] sofia_reg.c:1792 SIP auth challenge (REGISTER) on sofia profile 'internal' for [204(a)newplymouth.poiema.org] from ip 192.168.1.93 2017-08-09 10:13:51.062063 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/64225551212(a)206.191.159.247 [64db53af-fa7b-4878-9bdd-fd45a2f8f312] 2017-08-09 10:13:51.062063 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/64225551212(a)206.191.159.247) Running State Change CS_NEW (Cur 1 Tot 7) 2017-08-09 10:13:51.062063 [DEBUG] sofia.c:9873 sofia/internal/ 64225551212(a)206.191.159.247 receiving invite from 206.191.159.247:5060 version: 1.6.19 -36-7a77e0b 64bit 2017-08-09 10:13:51.062063 [DEBUG] sofia.c:10044 IP 206.191.159.247 Rejected by acl "domains". Falling back to Digest auth. 2017-08-09 10:13:51.062063 [DEBUG] sofia.c:2334 detaching session 64db53af-fa7b-4878-9bdd-fd45a2f8f312 2017-08-09 10:13:51.062063 [WARNING] sofia_reg.c:1792 SIP auth challenge (INVITE) on sofia profile 'internal' for [6468808998(a)newplymouth.poiema.org] from ip 206.191.159.247 2017-08-09 10:13:51.062063 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/64225551212(a)206.191.159.247) State NEW 2017-08-09 10:13:51.482817 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/64225551212(a)192.241.183.87 [8b74f385-3024-40fe-a83d-892e5ecd7bc3] 2017-08-09 10:13:51.482817 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/64225551212(a)192.241.183.87) Running State Change CS_NEW (Cur 2 Tot 8) 2017-08-09 10:13:51.482817 [DEBUG] sofia.c:9873 sofia/internal/ 64225551212(a)192.241.183.87 receiving invite from 192.241.183.87:5060 version: 1.6.19 -36-7a77e0b 64bit 2017-08-09 10:13:51.482817 [DEBUG] sofia.c:10044 IP 192.241.183.87 Rejected by acl "domains". Falling back to Digest auth. 2017-08-09 10:13:51.482817 [DEBUG] sofia.c:2334 detaching session 8b74f385-3024-40fe-a83d-892e5ecd7bc3 2017-08-09 10:13:51.482817 [WARNING] sofia_reg.c:1792 SIP auth challenge (INVITE) on sofia profile 'internal' for [6468808998(a)newplymouth.poiema.org] from ip 192.241.183.87 2017-08-09 10:13:51.482817 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/64225551212(a)192.241.183.87) State NEW 2017-08-09 10:13:52.182503 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/64225551212(a)107.6.13.77 [af5c7dd9-7848-4b59-8672-d62181ee1933] 2017-08-09 10:13:52.182503 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/64225551212(a)107.6.13.77) Running State Change CS_NEW (Cur 3 Tot 9) 2017-08-09 10:13:52.182503 [DEBUG] sofia.c:9873 sofia/internal/ 64225551212(a)107.6.13.77 receiving invite from 107.6.13.77:5060 version: 1.6.19 -36-7a77e0b 64bit 2017-08-09 10:13:52.182503 [DEBUG] sofia.c:10044 IP 107.6.13.77 Rejected by acl "domains". Falling back to Digest auth. 2017-08-09 10:13:52.182503 [DEBUG] sofia.c:2334 detaching session af5c7dd9-7848-4b59-8672-d62181ee1933 2017-08-09 10:13:52.182503 [WARNING] sofia_reg.c:1792 SIP auth challenge (INVITE) on sofia profile 'internal' for [6468808998(a)newplymouth.poiema.org] from ip 107.6.13.77 2017-08-09 10:13:52.182503 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/64225551212(a)107.6.13.77) State NEW 2017-08-09 10:14:01.182250 [WARNING] switch_core_state_machine.c:687 sofia/internal/64225551212(a)206.191.159.247 Abandoned 2017-08-09 10:14:01.182250 [NOTICE] switch_core_state_machine.c:690 Hangup sofia/internal/64225551212(a)206.191.159.247 [CS_NEW] [WRONG_CALL_STATE] 2017-08-09 10:14:01.182250 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/64225551212(a)206.191.159.247) Running State Change CS_HANGUP (Cur 3 Tot 9) 2017-08-09 10:14:01.182250 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/64225551212(a)206.191.159.247) Callstate Change DOWN -> HANGUP 2017-08-09 10:14:01.182250 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/64225551212(a)206.191.159.247) State HANGUP

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Re: [Freeswitch-users] How to reduce default sleep between transfers?
by Ahmed habiba 08 Aug '17

08 Aug '17

As mentioned by Brian you need to change the default password of the extensions in the "freeswitch/conf/vars.xml” as below: <X-PRE-PROCESS cmd="set" data=“default_password=xyzabc”/> *Note xyzabc is just example > > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] How to reduce default sleep between transfers? > Date: August 8, 2017 at 7:04:00 PM GMT+3 > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > That sleep is there on purpose in the configs and has you looked at fs_cli it complains that you’ve not changed the default password! CRIT messages:) > > > > On Tue, Aug 8, 2017 at 10:24 AM Jose David Jurado Alonso <josedavid(a)zennio.com <mailto:josedavid@zennio.com>> wrote: > Hi, > > I'm not sure but you can try to modify some of this timeouts: > > /etc/freeswitch# grep -r "10000" * > ... > dialplan/default.xml: <action application="sleep" data="10000"/> > ivr_menus/new_demo_ivr.xml: timeout="10000" > ivr_menus/demo_ivr.xml: timeout="10000" > skinny_profiles/internal.xml: <param name="digit-timeout" value="10000"/> > ... > > Probably, the correct variable to be modified is in "default.xml". > > > 2017-08-07 14:34 GMT+02:00 Euan Millar <euan(a)ensemblepourladifference.org <mailto:euan@ensemblepourladifference.org>>: > Hi, > > I am quite new to Freeswitch so this is probably a simple config question... > > When transferring a call I notice a 10 second sleep which can make users think that something has gone wrong. It is slightly unnatural between my IVR applications. > > I see this in the log: > > EXECUTE sofia/internal/1003@192.168.0.4:5060 <http://1003@192.168.0.4:5060/> sleep(10000) > > Is there a way to reduce this value? > > Many thanks, > > Euan > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > -- > Brian West > brian(a)freeswitch.org <mailto:brian@freeswitch.org> > Twitter: @FreeSWITCH , @briankwest > > http://www.freeswitchbook.com <http://www.freeswitchbook.com/> > http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/> > Book a phone call (CST) <https://freeswitch.com/appointment> > > Allison prompts for FreeSWITCH: > > https://www.gofundme.com/allison-prompts-for-freeswitch <https://www.gofundme.com/allison-prompts-for-freeswitch> > Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > Skype:briankwest > >

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How to reduce default sleep between transfers?
by Euan Millar 08 Aug '17

08 Aug '17

Hi, I am quite new to Freeswitch so this is probably a simple config question... When transferring a call I notice a 10 second sleep which can make users think that something has gone wrong. It is slightly unnatural between my IVR applications. I see this in the log: EXECUTE sofia/internal/1003@192.168.0.4:5060 sleep(10000) Is there a way to reduce this value? Many thanks, Euan

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att_xfer source leg and lua dialplan
by animal_alpha 08 Aug '17

08 Aug '17

Hello everybody! I am using Freeswitch with mod_lua and have the following setup: User 1004 is calling user 101, user 101 does an att transfer to 103, but user 103 see the caller-id of 1004(basically 1004 is calling both 101 and 103: mod_dialplan_xml.c:637 Processing 1004 <1004>->103 in context 1_local_103) 101, not 1004 should call 103 How can i get caller-id of 101 on 103(set -bleg on transfer?? )? 101 user's context is : <context name="1_local_101"> <extension name="101"> <condition field="destination_number" expression="^(101)$"> <action application="bind_meta_app" data="# b s execute_extension::att_xfer XML features"/> <action application="bridge" data="{sip_invite_domain=1.sips.local}sofia/internal/101(a)1.sips.local"/> </condition> </extension> <extension name="empty"> <condition field="destination_number" expression="^(.*)$"> <action application="hangup" data="NO_ROUTE_DESTINATION"/> </condition> </extension> </context> 103 user's context: <context name="1_local_103"> <extension name="103"> <condition field="destination_number" expression="^(103)$"> <action application="export" data="ignore_display_updates=true"/> <action application="bind_meta_app" data="# b s execute_extension::att_xfer XML features"/> <action application="bridge" data="{sip_invite_domain=1.sips.local}sofia/internal/103(a)1.sips.local"/> </condition> </extension> <extension name="empty"> <condition field="destination_number" expression="^(.*)$"> <action application="hangup" data="NO_ROUTE_DESTINATION"/> </condition> </extension> </context> features context: <context name="features"> <extension name="att_xfer"> <condition field="destination_number" expression="^att_xfer$"> <action application="read" data="3 4 'tone_stream://%(10000,0,350,440)' digits 30000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="att_xfer" data="{sip_invite_domain=1.sips.local}sofia/internal/${digits}@$${domain}"/> </condition> </extension> <extension name="empty"> <condition field="destination_number" expression="^(.*)$"> <action application="hangup" data="NO_ROUTE_DESTINATION"/> </condition> </extension> </context> -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/att-xfer-source-leg-and-lua-d… Sent from the freeswitch-users mailing list archive at Nabble.com.

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