February 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Change any character # with mod_translate?
by José Lopes 07 Feb '17

07 Feb '17

Hello, I am trying to change on the destination number, the character # to %23. I know that this is possible using lua script, but that it is also possible using mod_translate? Thanks in advance. Best Regards, Jose Lopes

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Japanese Audio Prompts
by Dayton Turner 07 Feb '17

07 Feb '17

Hi list, I'm wondering if anyone has put together (and can share) an audio pack for FreeSWITCH in Japanese? Some googling doesn't uncover much, sadly! -Dayton <http://www.getpostbox.com><http://www.getpostbox.com>

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Transcoding G729 to PCMA fails
by Igor Potjevlesch 07 Feb '17

07 Feb '17

Hello, I don't figure out why I can't make my calls transcoded from PCMA to G729. Here is the scenario: I send the call to external interface with G729. The call is properly sent to the leg B with PCMA. When the call is pick up by leg B, FS sends 488 to leg A after 200K/ACK with leg A. The module mod_com_g729 is properly loaded and has channels available. Thanks for your advice to diagnose this. Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

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generate deliberate packet loss?
by jungle boogie 07 Feb '17

07 Feb '17

Hi All, I would like to make a sip call through my freeswitch server to a device registered onsip and cause packet loss on either side of the call (not simultaneously, though). Is something I would be able to do through fs_cli? I know a little bit about the uuid_jitterbuffer, but that seems to cause delays with all the packets eventually arriving. Thanks!

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Prevent Call Transfer from a specific ip network(s)
by Lợi Đặng 07 Feb '17

07 Feb '17

Hi guys, I have 2 network-ip list A, and B that has processed similarly in xml_dialplan. Call Scenario is pretty simple: A/B -> FS -> C. For some reasons not listed, I'm using `proxy-refer=true` to proxy REFER msg to C. Using this setting make `force_transfer_context` useless. But I want to apply REFER from A only, REFER from B needed to be rejected with 403/405 or something. All REFERs are within dialog (RFC3515) I know I can place a SIP proxy before FS to achieve this. But I don't want to add a SIP component at the moment. So, can you guide me how to do this with FreeSwitch? rgds, Loi Dang Thanh Phone : +84. 1224.735.448 Email : loi.dangthanh(a)gmail.com

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How are you using FreeSWITCH?
by Ken Rice 06 Feb '17

06 Feb '17

Hey Guys, A lot of you are using FreeSWITCH for some fairly complex things. However that is not always clear to people discovering FreeSWITCH for the first time. I would like to build a list of examples. If you don't mind this info being compiled on Confluence. Please Reply with who, what, and where so we can document it and show off some of the really cool FreeSWITCH based or FreeSWITCH using deployments out there. Thanks Ken

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Have att_xfer search dialplan
by Rovshan Baghirov 05 Feb '17

05 Feb '17

Hi, When invoking transfer dialplan application I specify data="dialplan_entry XML domain_name". How can I also have att_xfer search dialplan entry and then make attended transfer to that entry. I'm trying various channel names but cannot get it right and keep getting mod_sofia.c:4641 Cannot locate registered user error. Thanks in advance. Best Regards, Rovshan

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Conference Talking Indicator
by Colin Morelli 05 Feb '17

05 Feb '17

Hey all, I believe the answer here is no, but I wanted to confirm. I'm using mod_conference and trying to get an indication of when someone in the conference is talking. I believe this is driven off of the "energy" setting. The problem is, when using the energy setting, mod_conference seems to clip a bit of the audio at the beginning and end of someone talking. I tried changing the energy settings around but I could never seem to eliminate this entirely, so I've turned off the noise gating of conferences. However, is there a way (or would it be reasonable to add) a separate number that is used for determining if a person is talking? In other words, separate the energy level that is used to determine if audio is bridged into the conference from the energy level that is used to determine if someone is speaking or not (with the default being that they're the same, of course). Is this reasonable? Is there something else you would suggest to fix the clipping issue I'm having so that I can turn the energy setting back on? Best, Colin

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A note from the Lead Developer. FreeSWITCH 1.8 Feature Requests
by babak 04 Feb '17

04 Feb '17

HiBuilt in support in Freeswitch database for muli-server Master-Master replication and also in modules like call center or conference for multi server environment for load balancing and High Availability. Regards

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FreeSWITCH at FOSDEM, SIP and WebRTC Load Balancing and High Availability
by Giovanni Maruzzelli 03 Feb '17

03 Feb '17

See you all, in person or in the intertube, at: https://fosdem.org/2017/schedule/event/freeswitch/ sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT

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