March 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

What Lua packages do I need to get started with Lua, especially if I want to use ESL?
by vfclists . 21 Mar '16

21 Mar '16

What Lua packages do I need to get started with Lua, especially if I want to use ESL, ie in addition to those which are automatically included by the Lua installation? Are there some recommended sets and download locations? -- Frank Church ======================= http://devblog.brahmancreations.com

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mod_verto - can't find user
by Shlomi Schwartz 21 Mar '16

21 Mar '16

Hi All, (Sorry for the newbe question) I'm using mod verto with: *<param name="blind-reg" value="true"/>* user login works fine, and I get: *{"jsonrpc":"2.0","id":1,"result":{"message":"logged in","sessid":"E1778D07-8A99-4DBF-88EE-2C57740205D0"}}* however when dialing I can see this warning in the log: *switch_ivr.c:3767 can't find user [1008(a)X.X.X.X] * And no call is initiated, what am I missing? -- This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this on behalf of the addressee you must not use, copy, disclose or take action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply email and delete this message. Thank you.

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Silent Call On Google Cloud
by Burak BorYazılım 21 Mar '16

21 Mar '16

Hi, I have FS 1.2.24 on Red Hat 6.7 Google cloud computing engine. Registration to server is successful and phones can make calls. But there is no sound in calls. Why is that? I think I opened the necessary ports that are explained at https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 Thank you, Regards...

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Disable picture_fast_update INFO requests
by Fabio Margarido 21 Mar '16

21 Mar '16

Hi there. I'm trying a scenario in which FreeSWITCH sits between a WebRTC endpoint and an IMS network, performing media conversion. When a call is established, FS starts to send INFO requests with an application/media_control+xml body related to RFC 5168. The WebRTC endpoint always responds with 415 Unsupported Media Type (the web application uses sip.js) and the IMS network sometimes responds with 500 Server Internal Error. The problem I have is that FS sends that request every second to both sides and keeps doing so for as long as the call is answered. My question is whether it would stop or at least scale down the frequency if both sides replied with 200 OK, or alternatively if there is a way to disable the sending of these requests. I've looked at the documentation and the mailing list history but couldn't find anything conclusive. Thanks in advance.

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Schedule Api
by Gregor Nanger 21 Mar '16

21 Mar '16

Playing with ESL and need some advice. How can I schedule this dialplan command for example: <action application="event" data="Event-Subclass=myevent::notify,Event-Name=CUSTOM,key1=value1,key2=value2"/> I want to shedule from ESL to raise CUSTOM event in x seconds, so I can catch this event in ESL. Should I send Api or Command from my code (ESL)? I tried to send like: sched_api +5 cc event Event-Name=CUSTOM, Event-Subclass=mysub::sub But I get error Invalid Command! Any hint?

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MOD_VERTO Login Incorrect
by Shlomi Schwartz 20 Mar '16

20 Mar '16

Hi All, I have reinstalled FreeSwitch with FusionPBX. After installation I have changed the default password at vars.xml and modified verto.conf.xml like so: ** *<param name="blind-reg" value="true"/>* I'm trying to login like so: *{"jsonrpc":"2.0","method":"login","id":1,"params":{"login":"1000(a)X.X.X.X","loginParams":{},"userVariables":{},"passwd":"my_pass","sessid":"6B3DC5AB-58E0-4375-82C1-E1A98ACBA5ED"}}* However I get a login error: *{"jsonrpc":"2.0","id":1,"error":{"message":"Login Incorrect","code":-32001}}* This setting used to work for me, what am I missing? -- Shlomi Schwartz R&D Technical Leader T: +972-74-700-4511 <http://www.linkedin.com/company/164748> <http://twitter.com/liveperson> <http://www.facebook.com/LivePersonInc> We Create Meaningful Connections <http://info.liveperson.com/201602/liz-welch/liveperson-robert-locascio-on-d…> -- This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this on behalf of the addressee you must not use, copy, disclose or take action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply email and delete this message. Thank you.

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How to stop the current dialplan application which is being executed
by Naveen Tamanam 19 Mar '16

19 Mar '16

Hi, How to stop the current dialplan application which is being executed. I'm aware of uuid_break, which is useful to stop the audio which is being sent on channel currently. But I'm looking for the way to stop the current application. UseCase: Here is the use case, why I asked this question, and what I have experienced. I'm playing some IVR application on channel, I switched back to playback application by calling uuid_break to stop the IVR, but I still hear the sound regarding IVR application(ivr greet short sound) during gap/silence of current playback sound. -- Thanks & Regards, Naveen Tamanam

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Freeswitch IVR: call drop on transfer to busy extensions
by Francesco Piccinin 19 Mar '16

19 Mar '16

Hello Giovanni and Bote, here you are updated pastebin on suggested url: Freeswitch dialplan files https://pastebin.freeswitch.org/24597 FSCLI internal call log: https://pastebin.freeswitch.org/24598 FSCLI external call log: https://pastebin.freeswitch.org/24599 Looking forward to your feedback. thanks Regards Francesco

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SRTP calls fail 'because secure media is administratively disabled'
by Eternal21 18 Mar '16

18 Mar '16

I am using FreeSWITCH demo configuration to test SRTP calls. I am placing a call between endpoint 1000 and 1001. The call works fine when SRTP is disabled on the softphones. The problem starts when I enable SRTP on the softphones. I also enabled SRTP on FreeSWITCH server, by including the following command in vars.xml: <X-PRE-PROCESS cmd="set" data="sip_secure_media=true"/> When I try to place a call between the two endpoints now, I end up with one side immediately hanging up. Specifically, the error seems to be: [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled Followed by: [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] Couldn't find any info on how to resolve this issue online. Here's the relevant log: 2016-03-18 09:45:21.664536 [INFO] switch_rtp.c:3605 Activating Audio Secure RTP SEND [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [INFO] switch_rtp.c:3583 Activating Audio Secure RTP RECV [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000(a)10.10.10.77! [DEBUG] switch_channel.c:3461 (sofia/internal/1000(a)10.10.10.77) Callstate Change RINGING -> EARLY [DEBUG] mod_sofia.c:2330 Ring SDP: v=0 o=FreeSWITCH 1458288533 1458288534 IN IP4 10.10.10.77 s=FreeSWITCH c=IN IP4 10.10.10.77 t=0 0 m=audio 20188 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uhco9Z9FgciE1vh5LQqGyNXxts0etxavW51gJKVP [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16@8000hz 1 channel 20ms [DEBUG] switch_core_codec.c:221 sofia/internal/1000(a)10.10.10.77 Push codec L16:100 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] [DEBUG] sofia.c:6756 Channel sofia/internal/1000(a)10.10.10.77 entering state [early][183] [DEBUG] switch_rtp.c:6645 Correct audio ip/port confirmed. [DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [completing][200] [DEBUG] sofia.c:6766 Remote SDP: v=0 o=VOCAL 61793 12653 IN IP4 10.10.10.112 s=- c=IN IP4 10.10.10.112 t=0 0 m=audio 1442 RTP/SAVP 0 a=rtpmap:0 PCMU/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i7qQOAKGWeBs/QtXQcYRD3g7u2pGv/BDqrvKE6Av a=ptime:20 [DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [ready][200] [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] [DEBUG] sofia.c:1355 Channel is already hungup. 2016-03-18 09:45:26.303802

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major build change in master.
by Michael Jerris 18 Mar '16

18 Mar '16

please note this important change that went into master today. As of today, we are no longer using system versions of libyuv and libvpx due to major conflicts with system versions of these libraries. These are now built static into the freeswitch core. Also note, mod_vpx no longer exists, it is automatically loaded as part of the core and you will no longer have mod_vpx.so or have to manually load it. I'll have more details coming, but let me know if you have any questions. Please note for anyone doing cross compiling, this probably needs a bit more work for you, I'll be working on fixing that this week. Mike

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