October 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS_PATH for Gateways
by Sharath Kumar 27 Oct '17

27 Oct '17

Can we add an fs_path when dialing out via a gateway ? I know this has been asked before and in the archives it has been said to support and also not support fs_path in a gateway. Just wanted to confirm. thank you, Shaks.

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Multiple "Changing expire time to 60 by request of proxy" requests result in delayed audio
by Erik M. Devane - Comms Guy 27 Oct '17

27 Oct '17

Hello, I have a fairly generic upstream VoIP provider. Every day or so of calls, I receive: 2017-10-26 16:54:06.100048 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:54:31.120584 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:55:09.139687 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:55:48.160453 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:56:04.180054 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:56:26.180559 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:56:43.199818 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:57:24.219901 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:58:08.260016 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:58:29.259860 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:59:00.281354 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 If there are just one or two, nothing seems to happen. When I see ten or more, there's an audio delay of five seconds or so. Can anyone help point me in the right direction? Thank you, Erik

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Gigaset auto answer
by Nicola von Thadden 27 Oct '17

27 Oct '17

Hi, I'm trying to get auto answer working on my Gigaset N510 IP DECT system. I made the changes to the DECT config (enabled "Call Manager" for the extension I want to use_ and tried to call it using the default prefix (8xxxx). Unlike the other phones I work with, the Gigaset device requires authentication for that. Since it is a subscriber to the freeswitch system, the system does not have any information to fulfill it: [ERR] sofia_reg.c:2561 Cannot locate any authentication credentials to complete an authentication request for realm '"test"' The INVITE itself looks as I would expect it to look but the phone answers with: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 46.38.244.168;rport=5060;branch=z9hG4bKK94yXK0rDUcXj From: "Nico" <sip:1000@xxxxx>;tag=6gtHSvy9p62HH To: <sip:1012@xxxxx:5060>;tag=2471884501 Call-ID: dee1f01e-2fc3-1236-8f91-52542680391a CSeq: 113890892 INVITE Contact: <sip:1012@xxxxx:5060> WWW-Authenticate: Digest realm="test", nonce="d0533fa35f7a53f86112305a818d732", algorithm=MD5, qop="auth" User-Agent: N510 IP PRO/42.243.00.000.000 Content-Length: 0 Does anyone have an idea how I can either get freeswitch to retry with authentication or how to get the Gigaset device to accept the call without further auth? Thanks Nico

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Freeswitch crashed daily
by Achintha 27 Oct '17

27 Oct '17

hi all Our freeswitch (Version 1.6.18-35-6e79667~64bit) handled around 7000 calls per day. Its coredb configured with postgresql (bdr) database (freeswitch DB) and following modules. mod_xml_curl : Dynamic Dialplans (from rest Service) mod_json_cdr : for CDR (from rest Service) and one module developed by our self (mod_ards) freeswitch crashed daily. here i have upload a core dump trace on pastbin https://pastebin.freeswitch.org/view/bf2b479d -- Best Regards.. Achintha Udukumbura

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call forward failing with enterprise originates
by sangeeth＠xoyal.com 26 Oct '17

26 Oct '17

Hi We have an extension(1000) registered at an IP phone which have a call forwarding to an external number. and the other extension(1002) on other ip phone. when i make an enterprise originate to two destinations. the 1000 gives me infinite invite requests instead of forwarding. and stop ringing. i refered to the Jira https://freeswitch.org/jira/browse/FS-6814 but looking if it had been resolved yet . BR, sangeeth kumar.

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mod_conference video_no_video_avatar_png flag with video-mode passthrough
by Richard Screene 26 Oct '17

26 Oct '17

Hello, In mod_conference, is it possible to use the video_no_video_avatar_png flag with video-mode passthrough? I can’t get it working but that could indicate that either (a) its not possible or (b) I’ve done something wrong. What I’m trying to achieve is: in a conference with both audio/video and audio only members the video stream follows the active speaker. However, if one of the audio only participants is speaking then a placeholder image is transmitted to the audio/video members instead. Obviously, I’d prefer to reduce the amount of transcoding/canvas manipulation on the server hence the use of the passthrough video-mode. If there any better ways of achieving this then please let me know! Many thanks for reading this, Richard

1 0

Share Screen stop work after ~20 seconds (FS 1.6.18/1.6.19/master)
by José Lopes 26 Oct '17

26 Oct '17

Hello, I have a Freeswitch WebRTC SIP connected with a Freeswitch MCU. When i use Video Conference on WebRTC, when I share the Screen, the screen is shared, but after 20 seconds the screen freeze and it stops to be presented. The video call of Screen Share is still active. I have this situation after upgrade to FreeSwitch 1.6.18. On Freeswitch 1.6.17 this situation does not happen. I tried also on Freeswitch 1.6.19 and master and this situation still happens. There is any configuration that I need to use on the new versions related to this scenario, so this will not happen? Additional Notes: * On Freeswitch MCU, I made a change that when it is video from Screen Share, the conference member is with flags : mute and deaf. After 20 seconds the video freeze , but after some seconds the screen share start working again. * I tried a Freeswitch WebRTC with MCU, and this situation doesn't happen. Thanks in advance Best regards, *José Lopes*

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is it possible to use mod_callcenter queues without answering the line
by Khalil Khamlichi 26 Oct '17

26 Oct '17

Hi, is it possible to use mod_callcenter queues without answering the line, that is the customer is held in the queue and hearing ringing (without the call being answered). In fact this is a mandatory requirement when calls have a cost and we don't want to charge customer for waiting time but only for talk time. Thanks in advance. Khalil

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[ERR] mod_lua.cpp:103 attempt to call a nil value in recordFile function
by Akshaya j 26 Oct '17

26 Oct '17

Hi team, Spec: OS - Debian FS server Version -1.6.15 I use Zoiper clients (in Ubuntu machine) directly connected to FS server. I'm supposed to write voicemail scripts where i get the call and bridge to Avaya gateway if there is no response i load a lua script. The lua script contains........... if session:ready() then session:speak("Record your message now, to end recording press pound"); session:setInputCallback('onInput',''); test = session:recordFile(recording_filename, 60, 30, 5) session:consoleLog("info", "session:recordFile() = " .. test ) And my Input CallBack function has got..... function onInput(session, input_type, data) session:consoleLog("info", "inside" ) if input_type == "dtmf" then return "break" end end Now my problem is I dont see any log which is inside the callback function.I can record the voice and getting saved but there is an error.... [DEBUG] switch_rtp.c:7255 RTP RECV DTMF #:1280 [INFO] switch_channel.c:515 RECV DTMF #:1280 [ERR] mod_lua.cpp:103 attempt to call a nil value stack traceback: [C]: in function 'recordFile' /usr/local/freeswitch/scripts/test.lua:19: in main chunk I can witness that the session is receiving the DTMF signal but then the function is not getting executed. I tried commenting the func call of callback and I can witness the DTMF is not sensed and hence the record termination happens only after silence_threshold. I badly need to process the keypress DTMF for more options without console err. NOTE: instead of recordFile() I tried speak() even by then i get the same error "attempt to call a nil value" in speak function. Please save me FS.!

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outbound route question
by alex 26 Oct '17

26 Oct '17

I create two sip extension 1000 and 1001,i have to calls 1000 to 1001. This is my outbount route: <include> <context "name=default"> <extension name="outgoing"> <condition field="destination_number" expression="^(100)$"> <action application="bridge" data="sofia/freetdm/wp1/a/$1"/> </condition> </extension> </context> </include> But it didn't take the route. Why is that? what should i do? This are CLI log: 2017-10-13 05:02:36.296708 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000(a)192.168.1.70 [b12d303a-1f58-4731-b568-78e2b256c28b] 2017-10-13 05:02:36.316705 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1234 in context public 2017-10-13 05:02:36.316705 [NOTICE] switch_ivr.c:2201 Transfer sofia/internal/1000(a)192.168.1.70 to XML[1001@default] 2017-10-13 05:02:36.316705 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1001 in context default 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 Once changed type 'reloadxml' at the console. 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *1 execute_extension::dx XML features 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1000.2017-10-13-05-02-46.wav 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *3 execute_extension::cf XML features 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *4 execute_extension::att_xfer XML features 2017-10-13 05:02:46.336708 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1001@192.168.1.69:5060 [4e2e818f-98f5-461b-ad4c-ad34eb933191] 2017-10-13 05:02:46.336708 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2017-10-13 05:02:46.396708 [NOTICE] sofia.c:7391 Ring-Ready sofia/internal/1001@192.168.1.69:5060! 2017-10-13 05:02:46.396708 [INFO] switch_ivr_originate.c:1220 Sending early media 2017-10-13 05:02:46.416708 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000(a)192.168.1.70! 2017-10-13 05:02:50.016708 [NOTICE] sofia.c:8474 Hangup sofia/internal/1000(a)192.168.1.70 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2017-10-13 05:02:50.016708 [NOTICE] switch_ivr_originate.c:3629 Hangup sofia/internal/1001@192.168.1.69:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2017-10-13 05:02:50.036718 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2017-10-13 05:02:50.036718 [INFO] mod_dptools.c:3508 Originate Failed. Cause: ORIGINATOR_CANCEL 2017-10-13 05:02:50.036718 [NOTICE] switch_core_session.c:1731 Session 2 (sofia/internal/1001@192.168.1.69:5060) Ended 2017-10-13 05:02:50.036718 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1001@192.168.1.69:5060 [CS_DESTROY] 2017-10-13 05:02:50.056713 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000(a)192.168.1.70) Ended 2017-10-13 05:02:50.056713 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000(a)192.168.1.70 [CS_DESTROY] Thanks for answer.

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