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Verto pass variables dialplan
by Igor Olhovskiy 06 Nov '14

06 Nov '14

Hi! Is there any way to get in dialplan (XML) parameters, passed from browser via js? In my case I appended extraHeaders. For ex, 2014-11-06 17:25:19.364834 [ALERT] mod_verto.c:1284 READ 178.54.1.162:52512 [{ "jsonrpc": "2.0", "method": "verto.invite", "params": { "sdp": "......", "dialogParams": { "useVideo": false, "useStereo": false, "tag": "webvideotag", "login": "XXXXX(a)webrtc-dev.webcall.today", "destination_number": "9001", "caller_id_name": "XXXXXX", "caller_id_number": "XXXXX", * "extraHeaders": {** ** "PageUrl": "https://my.webcall.today/"** ** },* "callID": "db94fb82-f113-62e5-9e61-e091715b3c4f", "remote_caller_id_name": "Outbound Call", "remote_caller_id_number": "9001" }, "sessid": "6a4c4a47-9de0-8af3-38e0-971ba1f63198" }, "id": 4 }] Is there any way to get this PageUrl parameter in dialplan variables? Freeswitch 1.5.14b -- Regards, Igor.

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Does freeswitch support acc format?
by Aqs Younas 06 Nov '14

06 Nov '14

Hi, I am new to freeswitch. Does anybody know freeswitch support acc format?

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Recommended FS SOHO configuration, test scripts
by Bill Ross 06 Nov '14

06 Nov '14

Folks; I am creating a hybrid OpenWrt (recent trunk) / Freeswitch-1.4.12 office in a box distribution intended for SOHO's, small business and, end users looking for secure telephony (ZRTP) solution. I am fine tuning the configuration and have a Freeswitch guru question, requiring knowledge of how freeswitch is used in the field. In particular, which modules I should compile freeswitch with. My current (compile successfully, not fully tested) modules.conf is here: http://www.rossco.org/Files/modules.conf-1.4.production.txt any suggestions regarding module additions / deletions would be greatly appreciated. Also, any pointers to scripts for automated feature / load / regression testing would be greatly appreciated. Thanks; Bill Ross

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By pass only video rtp
by Doron Kruh 06 Nov '14

06 Nov '14

Hello, Is there a way to bypass media only of the Video RTP stream and still process the audio RTP of a video call through Freeswitch? Thanks, Doron. -- Doron Kruh, R&D Team Leader Lexifone , Hapalyam 7 Haifa. Mobile: 972-52-3823361 Office: 972-4-9126793

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PostgreSQL - p_key
by Joel White 06 Nov '14

06 Nov '14

Is it feasible to add Primary Keys to all FreeSWITCH tables or would this break some functionality? I am asking because I am working on a multi-master replication setup

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Gateway Status 908
by Spencer Thomason 05 Nov '14

05 Nov '14

Hello, I’m a bit confused by gateway timeout status 908. On a REGISTER timeout, the status is set to 908, gateway_ptr->failure_status = 908; Is this a typo? Should it not be 408? I know the 908-retry-seconds parameter was added recently, I guess i’m just confused that its not 408-retry-seconds. Thanks, Spencer

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FreeSWITCH 1.4.13 Released!!
by Ken Rice 05 Nov '14

05 Nov '14

New Post on freeswitch.org from krice387 check it out at http://ift.tt/1rUJmfP FreeSWITCH 1.4.13 Released!! The FreeSWITCH Team is happy to announce the release of FreeSWITCH 1.4.13!!!! Tarball is available at http://ift.tt/1rUJls8 as well as packages for Debian and Centos6 in their respective repositories! This Release includes: Fix SegFault in switch_core_media.c – http://ift.tt/1ocyP4k Updates for Xcode 6 on OSX Mavericks and Yosemite – http://ift.tt/1rUJlsa Allow sub millisecond resolution for option ping times – http://ift.tt/1rUJlsc Add ability to log commands executed in mod_xml_rpc Fix error when resuming a call on hold – http://ift.tt/1rUJmMJ Fix missing Notify message record-route tag – http://ift.tt/1ocyRZS Add new hard_mute control to allow apps to request low level mute e.g. from the rtp stack level. Its used in mod_conference to avoid reading audio while muted and possibly reduce some transcoding load Improve SIP OPTIONS ping generation by distributing them across an interval – http://ift.tt/1rUJlsi variable digits_dialed_filter to set regular expressions with () captures and anything matched will be replaced with X’s in the CDR (useful for filtering things like CreditCard numbers) – http://ift.tt/1ocyS00 fix leak of nua handle due to reference counting. Effects all calls with auth/challenge on INVITE as well as several other minor tweaks and changes There are also several other minor changes and bug fixes in this release! Along with this release we had to fix a small error in the main FreeSWITCH GIT repository on the v1.4 branch. On your next pull of the branch you may see a merge error. To resolve this error from your local copy of the git repo while on the v1.4 branch issue the following command:git reset --hard origin/v1.4 Please note, if you have local patches you will want to rebase from the main FreeSWITCH git repository to resolve the issue.

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srtp-sdp header tag.
by divyesh kamothi 05 Nov '14

05 Nov '14

Hello team I updated freeswitch to 1.2 to 1.5. now here issue is when in using srtp. with freeswitch 1.2 :: In SDP value after a=crypto:0 <= this is 0 with freeswitch 1.5 :: In SDP value after a=crypto:8 <= this is 8 is there any configuration or other way using which i can change digit to 0. -- Thanks and Regards Divyesh Kamothi PH : - 00-91-84605-18040

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Freeswitch Week in Review(Master Branch) October 5th- 12th
by Ken Rice 05 Nov '14

05 Nov '14

New Post on freeswitch.org from kathleen check it out at http://ift.tt/1AdJhhd Freeswitch Week in Review(Master Branch) October 5th- 12th Hello, again. This week in the FreeSWITCH master branch we had 28 commits. Some of the feature work done this week included improvements to the jitter buffer, the addition of uuid_send_info enhancement that allows setting the Content-Type of the SIP INFO message in mod_commands, and a 908-retry-seconds gateway param to set reg retry time when getting a 908 for backup interfaces to connect quickly in mod_sofia. New features that were added: da43bde Add some calculations to jitter buffer related to judging the optimal size eaaf946 FS-6897: uuid_send_info enhancement that allows setting the Content-Type of the SIP INFO message in mod_commands [Jira: http://ift.tt/1txlhzp] a4f840b More jitter buffer improvements 855cc4b Add 908-retry-seconds gateway param to set reg retry time when getting a 908 for backup interfaces to connect quickly in mod_sofia The following bugs were squashed: 397ec5a Fixed a jitter buffer logic error 490efb7 FS-6710: fix incorrect comparison for Min-SE values between SIP INVITE and local configuration in mod_sofia [Jira: http://ift.tt/1AdJhhm] 7ec7c92 OPENZAP-220 fix blocked into read and add cause for a correct hangup 4a5e36d Fixed switch_pgsql_next_result_timed() to use switch_micro_time_now() to be consistent 2051a86 FS-6889 Fix for Reinvite Codec Error when using bypass_media_resume_on_hold and bypass_media_after_hold in mod_sofia [Jira: http://ift.tt/1E2VIsW] d48057e FS-6890 Fix to account for lost frames during ptime detection 28bc992 Fix error in SRGS grammar parser in mod_rayo 43c2c6d FS-6815 Fix for fs_encode issue [Jira: http://ift.tt/1txlkv7] 66dafbd FS-6902 Add patch to make native SILK encoded playback problem obvious and fail on record and playback [Jira: http://ift.tt/1txlhzr] e4e9b1b Fix to have resume media on hold not send invite back out at the holder but rather enable media in the 200ok in mod_sofia In terms of stability these were the use cases that were fixed: 2514de9 Fix obvious seg in setting a record file name to every participant and not checking for the recording member which does not have a session in mod_conference b5294c5 Fix crash on transport=tls with non-TLS profile in mod_sofia Improvements in cross platform build supports: 6146efd FS-6870 Fix for build issue with vs2010 [Jira: http://ift.tt/1E2VGBl] The complete list of commits can be found here:2014_10_5-2014_10_12

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ICTFax Version 3.0 released
by Tahir Almas 04 Nov '14

04 Nov '14

Please to announce the release of ICTFax Version 3.0 , for more information please visit http://www.ictinnovations.com/content/ictfax-version-30-fax-over-ip-solutio… http://www.icfax.org regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT

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