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Lua not working when FS start with background mode[freeswitch -nc]
by zzy 24 Mar '17

24 Mar '17

Debian 8 FreeSWITCH (Version 1.6.15 -32-bec4538 64bit) 1. Login Debian with root. 2. Install the FS by apt-get. wget -O - https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub| apt-key add - echo "deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/jessie main" > /etc/apt/sources.list.d/freeswitch.list apt-get update && apt-get install -y freeswitch-meta-all 3. Run fs_cli after install. 4. I add a gateway in Freeswitch Portal. It will run a run lua script [create_gateway.lua]. This lua script is from source code [/usr/src/freeswitch/scripts/lua/portal/create_gateway.lua].I get an error message in fs_cli: 2017-03-24 10:32:46.006528 [ERR] mod_lua.cpp:203 /usr/share/freeswitch/scripts/portal/create_gateway.lua:13: attempt to index global 'conf_file' (a nil value) stack traceback: /usr/share/freeswitch/scripts/portal/create_gateway.lua:13: in main chunk 5. Stop the FS with [systemctl stop freeswitch.service] command. 6. Run [freeswitch -c] command set the FS in the foreground. 7. I add a gateway in Freeswitch Portal again and get the correct result. 8. I also try this after FS startup in the background ([freeswitch -nc] and [freeswitch -ncwait]). Get the same error as before. Who can tell me why the lua script does not working when FS startup with service mode and background mode? Thanks a lot!

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Dialplan conditions for limitations
by Yücel ALTUNAY 24 Mar '17

24 Mar '17

Hi, I have 3 extensions (1001, 1002, 1003) and i want to limit some calls written below *Ext Can Call Cant't Call* 1001 1002,1003 1002 1001 1003 1003 1001 1002 I wrote condition below; <extension name="Local_Extension"> <condition regex="any"> <regex field="destination_number" expression="^(1001)$"/> <regex field="caller_id_number" expression="^(1001)$"/> ....... <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/>  <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> In this case 1002 and 1003 can call 1001, and can't call each other, but when i want to call 1002 or 1003 from 1001, on 1001's screen i see a incoming call from others. How can i solve this case. Thanks.

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SIMPLE, SRTP and Zoiper
by Rajil Saraswat 24 Mar '17

24 Mar '17

Hello, I have SRTP setup in Freeswitch with Zoiper on Android being the end points. Using Zoiper it is also possible to send text messages. Are these messages encrypted as well? Thanks

3 2

change value in invite sip header
by Ankit Doshi 24 Mar '17

24 Mar '17

Hello All, I have DID Number :123456This DID assigned to 23456 extension. Now when I am dial number so that time sip invite header pass like belowINVITE sip:23456@ipBut I want to send like INVITE sip:123456@ip So which sip URI variable can I try to get this resolved ? Sent from Yahoo Mail on Android

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Configure SRTP and ZRTP on FusionPBX
by Ronggo Kinantoputro 24 Mar '17

24 Mar '17

Hi, I have recently installed FusionPBX (version 4.0.0-151125) on AWS. My purpose is : 1) Make a secure call using SRTP (end to end SRTP session) I've already make some test using softphone (Linphone and Csip Simple for android), with enabling SRTP settings on both end point. The result is from the Active Calls Status(on FusionPBX Web-UI), the Caller Extension has already secure (there's SRTP crypto on the Secure column). But at the Receiver Extension, there's no SRTP crypto on the Secure column. Are those calls has been secured (end to end)? 2) Make a secure call using ZRTP (end to end ZRTP session) I've already make some test using softphone (Linphone and Csip Simple for android), with enabling ZRTP settings on both end point. The result is from the Active Calls Status(on FusionPBX Web-UI), there's no ZRTP crypto on the Secure column for both Caller Extension and Receiver Extension. So I guest the ZRTP session (end to end) still did not works. I have not make any changes with the configuration on FreeSwitch/FusionPBX, so it's still in default configuration. Any advise would be highly appreciated. Thanks.

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Mod_callcenter question
by Kamil Nigmatullin 24 Mar '17

24 Mar '17

I have CRM system and all out clients are devided between managers. CRM system knows who is client's manager. So on incoming call I want to try first this manager, and if he is not answering then route him to another manager. Is there a way to do so somehow passing parameters to callcenter application. I want to get all this call through mod_callcenter to keep statistics and queuing correct. -- Kamil Nigmatullin Skype: kamil.nigmatullin

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Call drop on transfer (REFER) after SIP 302 MOVED
by Piccinin Francesco 23 Mar '17

23 Mar '17

Hi all, we are experiencing a strange issue during call transfer (REFER) after a SIP 302 MOVED. Please find below a brief description of our test environment. Extensions involved: - 8911 (original caller) - 8910 (forwarded to 5536 through phone ui) - 5536 (originator of call transfer) - 5532 (destination of call transfer) Step to reproduce the issue: - 8911 calls 8910 which is forwarded on 5536 (call fwd is configured directly on phone gui) - 5536 correctly answer the call from 8911 - 5536 begins a consult transfer to 5532: Hold step is ok but call drops when 5536 completes the transfer. >From the SIP signaling point of view: - 8910 answers to 8911 with a SIP MOVED. - Freeswitch send an invite to 5536 as 8911. - Call transfer from 5536 to 5532 is managed with a refer. - When 5536 complete the transfer, Freeswitch send a bye first to 8911 (original caller) and then to 5536. On Freeswitch dialplan we manage refer on a different context, setting "sip_redirect_context" variables. Without MOVED involved, all call transfer are correctly managed. It seems that we need to keep alive a-leg after the bridge and then perform the transfer. We tried all the following settings on redirect context: - set + export of variables: force_transfer_context + force_transfer_dialplan - set + export park_after_bridge=true - set continue_on_fail=true - unset of sip_redirect_context But all without success! Please find attached a pcap trace and freeswitch cli log. Thanks in advance for help. Regards Francesco

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Codec late-negotiation change between 1.6.6 and 1.6.15
by David Ponzone 22 Mar '17

22 Mar '17

Does anyone know if there was a change with late-negotiation between 1.6.6 and 1.6.15 ? I am not sure yet about the issue, I am still trying to figure it out. According to the release notes, there were no change in this area. David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr <blocked::http://www.ipeva.fr/> - www.ipeva-studio.com <blocked::http://www.ipeva-studio.com/> Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

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Luis Daniel Lucio Quiroz
by Mario G 22 Mar '17

22 Mar '17

What a scummy thing to do putting people here on linkedin which shares our email addresses. Hope no one uses this guy! Mario

1 0

Dialplan/default.xml
by Yücel ALTUNAY 22 Mar '17

22 Mar '17

Hi, I installed FS 1.6 on Debian and using it on local lan. Do I need extensions in dialplan/defaut.xml except Local Extension? Thanks.

2 1