June 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] ICE issue
by Bastien LE REST 13 Jun '16

13 Jun '16

Bonjour, I thought the issue was solved since the calls work just fine now with PJSIP on iOS, however the calls don't work with PJSIP on Android and I don't understand why, the PJSIP configuration is exactly the same on both devices. FS finds a candidate for RTP but it doesn't find a candidate for RTCP. Please note that I only have 3 candidate. FS trace: https://pastebin.freeswitch.org/view/8d4df983 Could you please have a look at the trace, and let me know what am I doing wrong? Thank you very much in advance for your help in this matter. Kind regards, Bastien. On 4 June 2016 at 13:44, Bastien LE REST <ba.lerest(a)gmail.com> wrote: > Hi, > > When I enable ICE using PJSIP, FreeSwitch answers "488 Not Acceptable > Here". When I disable ICE, I don't have any problems, the calls work just > fine. I tried to use another SIP client with ICE and the call works fine. > So I think the problem has to do with PJSIP. > > I'm working on the migration from an Asterisk architecture to a FreeSwitch > architecture. I have to deal with PJSIP and ICE support because my client > is a mobile application who is already published on the stores. > > PJSIP trace : https://pastebin.freeswitch.org/view/ddeaae4b > FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d > > Could you please have a look at the trace and tell me what am I doing > wrong? Thank you in advance. > > Kind regards, > Bastien. >

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Reject Inbound Call and Callback
by Deepika Yadav 13 Jun '16

13 Jun '16

Hi, I want to reject an incoming call from a mobile number in the public context and callback to it how should the dialplan look like: <include> <extension name="public_did"> <condition field="destination_number" expression="^(88888888)$"> <action application="set" data="domain_name=$${domain}"/> <action application="log" data="INFO domain is $${domain}"/> <action application="set" data="callerID = $${caller_id_number}"/> <action application="log" data="INFO src number is $${callerID}"/> <action application="respond" data="503"/> <action application="bridge" data="{sip_auth_username=1003,sip_auth_password=R@D ~o$121$@IIITD}sofia/external/1003@52.38.67.78 9769658019"/>   </condition> </extension> </include> Regards, Deepika

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Problems with wait_for_silence API call
by Matthew 13 Jun '16

13 Jun '16

Hello All, I've having a problem with the wait_for_silence API call in lua. When I attempt to orginate a call and then check for silence on the line, I'm getting unpredictable results. Sometimes it seems to work, but more times than not it times out ( switch_ivr_wait_for_silence: TIMEOUT 0 ). To test this, I answer the call, speak a short greeting and then mute the handset. In other words, I know the script should be detecting silence on the other end of the call. Here is a good portion of the test script I'm using ... local cid_name = argv[1] local cid_number = argv[2] local gw_string = argv[3] local queue_ext = argv[4] dialout_string = "{origination_caller_id_name='" .. cid_name .. "'," .. "origination_caller_id_number=" .. cid_number .. "," .. "ignore_early_media=true}" .. gw_string dialout_session = freeswitch.Session( dialout_string ) while dialout_session:ready() and not dialout_session:answered() do freeswitch.msleep( 250 ) end if dialout_session:ready() and dialout_session:answered() then dialout_session:consoleLog( "INFO", "dialout: connected to " .. gw_string ) ) dialout_session:execute( "voice_start" ) dialout_session:sleep( 2000 ) dialout_session:execute( "wait_for_silence", "300 30 15 15000" ) dialout_session:execute( "voice_stop" ) dialout_session:consoleLog( "INFO", "dialout: amd_detect = " .. amd_detect ) end And here is an example of how I would invoke the script ... luarun dialout.lua TEST 1234567890 sofia/gateway/twilio/+1234567890 7000 Any ideas why the wait_for_silence API call would be misbehaving? Any suggestions on how I could debug this problem further? Thanks, -Matthew

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Channel/session variables availability on execute_on_sip_reinvite
by Florent Krieg 13 Jun '16

13 Jun '16

Hello everybody, My use case is pretty simple: I would like to react on SIP Re-INVITE without sdp (empty body/content-length 0) so I have put an execute_on_sip_reinvite instruction in my bridge string. I have tried both to call the 'info' app and to call a custom shell system script, passing the ${uuid} var and calling fs_cli uuid_dump $uuid, but unfortunately it seems that when the SIP message reaches the application it has been processed by the Sofia stack and I can't detect if there is an SDP or not (using sdp variables, or any other actually). Sometimes it gives me 0.0.0.0 as the remote_media_ip, sometimes it just builds an SDP based on the previous one. I have also tried to modify the re-INVITE by adding custom headers or putting information in any mandatory SIP header (like, say, Via), but these updates doesn't seem to be available via any channel variable. Is there any FS configuration parameter I could change to be able to access this information? Is there any other way I could do what I want? Thanks in advance if you have any clue :) Florent

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Need help on call bridge !!
by turja 13 Jun '16

13 Jun '16

Hello i am new user of Freeswitch. i am trying to write a dial plan that when i call to 1234(callee) first call bridge to 1233(another number) after 1233 hangup the call bridge 1234(callee) ....but can not ...after hangup from 1233(another number)the call hangs up not bridging with 1234(callee)..... how to do that in freeswitch. Thanks for help ! Turja -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Need-help-on-call-bridge-tp75… Sent from the freeswitch-users mailing list archive at Nabble.com.

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add modules when installing from the latest branch
by Matt Broad 13 Jun '16

13 Jun '16

Hi, this may be a really stupid question but here goes: I have upgraded a server to debian 8 and have installed freeswitch using the latest branch https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian… : My question is how do I activate other modules? In previous versions I would have amended the moudules conf file and re-compile. I can't find the modules file in this install thanks Matt

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ReInvte in bypass/proxy mode | a > a+v
by Mirko Brankovic 13 Jun '16

13 Jun '16

Hi, Can someone help me figure out the problem i am hitting. Im trying to upgrade call from audio to audio + video (opus 48k and vp8- but this one is only in passtrough mode). So set up is browser (sip.js/node.js) >> kamailio wss >> freswitch (bypass mode bridge without answer) >> kamailio >> browser (node.js) Problem is that freeswitch receives reInvite on existing audio call with new sdp containing also video codecs, but directly from client so no proxy (kamailio) in path, and exiting invite towards kamailio for outgoing leg is missing video in a=group BUNDLE, so only audio and a=sendrcv is also being stripped (that is the difference to 1 on 1 video call). I understood that bypass mode shouldn't change/touch sdp. Also passtrough codec shouldn't be the problem i guess since both clients can decode it. Any help is appreciated. Thanks, Mirko

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SHA-2 Implementation
by Abdul Hakeem 12 Jun '16

12 Jun '16

Hello, Any plans of moving FS from SHA-1 to SHA-2 crypto hash support ? I understand SHA-1 will be deprecated in Chrome by December, this will cause all calls to Verto to fail. Cheers, AH

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Authentication of SIP phones against LDAP directory?
by Kevin Long 12 Jun '16

12 Jun '16

Hello, Can Freeswitch authenticate SIP phone logins from an LDAP directory ? Hoping to integrate this so my SIP users can use the same password/username they use for all other applications. Regards, Kevin Long

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Auto record by call direction
by Ashwin Rath 12 Jun '16

12 Jun '16

Hello To record a call (inbound or outbound) i set the following variable for a user in the directory folder <variable name="user_record" value="enabled"/> However i would like to record inbound calls only. Is there some way to selectively auto record inbound calls only ?

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