November 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Vulnerability Testing
by Jerry Allen 04 Nov '14

04 Nov '14

I believe I have my system "ready for use" Are there any exploit testing tools that test for the ability to make calls etc. off of a system? I have tested with sipvicious and sipsak but I am not positive that I am testing correctly. I would like a real hack test. The system has been "wide open" for a few weeks, fine-tuned fail2ban as well as the addition of a firewall that protects my FS machine (hopefully) and there have been no outbound calls made that I did not make there have been hundreds of various hackers/scanners that fail2ban blocked etc. I am not exactly sure where to go to make sure it is not a big bomb waiting to be set off. Jerry Allen

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XML DialPlan help
by Sean Devoy 04 Nov '14

04 Nov '14

I swear I have read this, but now that I need it I can't find it. My user wants to dial international. I want to require them to do SOME extra step other than dialing 0113010... to call the Netherlands. Oddly enough, it is cheaper than US termination - go figure. Ideally, I would like them dial some extension like 777, hear a new dialtone and then accept the international number. I might even like to play a message that says "All charges for International are their responsibility and not part of their current package" before the dialtone. Ideas? Thanks again, Sean

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Freeswitch Week in Revew (Master Branch) September 28th-October 5th
by Ken Rice 04 Nov '14

04 Nov '14

New Post on freeswitch.org from kathleen check it out at http://ift.tt/1E2VPVl Freeswitch Week in Revew (Master Branch) September 28th-October 5th Hello, again. This week in the FreeSWITCH master branch we had 48 commits. We had a few new features added, like, tab completion for api commands in mod_gsmopen, a new feature to filter the SDP on bypass_media calls to remove or limit codecs, and a way to globally disable system commands by setting global var disable_system_api_commands=true. The following bugs were squashed: d619017 FS-6757 FS-6713 FS-6868 FS-6863 FS-6858 5 bugs one typo. From commit 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira: http://ift.tt/1uqVaxG;] da51603 FS-6757 FS-6713 FS-6868 FS-6863 FS-6858 #resolve #comment 5 bugs one typo. From commit 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira: http://ift.tt/1sh2sks] fbe857e FS-6644 Fix ptime from known broken endpoints on re-invite [Jira: http://ift.tt/1E2VGkM] 0150c86 FS-6854 Patch for SIP lockout on WebRTC in mod_verto [Jira: http://ift.tt/1uqV8WN] 644b41f FS-6874 Fix for multipart invite format in mod_sofia [Jira: http://ift.tt/1uqV9d2] 35aeae0 FS-6822 #comment The code in question appears to have been added by me (18f20e24). I think this patch is the correct solution. [Jira: http://ift.tt/1uqV9d6] 8db31f9 Fix some recovery issues with dynamic payloads ae5d865 FS-6884 #comment Fix for simple warnings in mod_sofia and mod_conference [Jira: http://ift.tt/1E2VGBa] 01bf422 FS-6888 Fix regression from refactoring new feature [Jira: http://ift.tt/1uqVaxL] bde2e2d 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] b2ae5f4 Fixed a few bugs in the new features in mod_sofia afd6875 FS-6781 Adding a check to ensure consistency in mod_valet_parking [Jira: http://ift.tt/1uqVaxP] New features that were added: a94fbe8 Add tab completion for api commands in mod_gsmopen 92a66fb Improve adaptive jitter buffer ascending check in switch_stfu 24084ad Add new feature to filter the SDP on bypass_media calls to remove or limit codecs. 8e408e9 FS-6865 #resolve add XMPP priority to dingaling [Jira: http://ift.tt/1E2VIJi] 789e148 FS-6880 Improvements to filtering invites in the core. [Jira: http://ift.tt/1uqVaxR] b3d7191 FS-6870 vs2010 and vs2012 fix [Jira: http://ift.tt/1E2VGBl] 91ffe17 Use OPUS_APPLICATION_VOIP always to get FEC and filtering in mod_opus 9e91753 FS-6886 Addition rtp_disable_hold when bypass_media is active and addition of ignoring unhold in mod_sofia [Jira: http://ift.tt/1uqVaxT] 6bfc05b FS-6887 Add always_auto_adjust parameter [Jira: http://ift.tt/1E2VGRD] 43733a6 FS-6886 #comment addition of ignoring unhold as well to mod_sofia [Jira: http://ift.tt/1uqVaxT] 10a3fa5 Add bypass_media_resume_on_hold and bypass_media_after_hold variables to be set to true to enable these functions on a per channel basis 0d1f5d0 Add way to globally disable system commands by setting global var disable_system_api_commands=true Improvements in cross platform build supports: 812f911 Fix reference to libressl in makefile on OpenBSD a39db86 FS-6870 #vs2012 and vs2010 make download of openssl dependent [Jira: http://ift.tt/1E2VGBl] d17f14e Make sure to pass along appropriate configure flags to sub-configure’s when cross compiling d52cb33 Fix trivial vs2010 build errors The complete list of commits can be found here: 2014_9_28-2014_10_5

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mod_verto unwanted profile shutdown
by Balázs Sándor 04 Nov '14

04 Nov '14

Hi all. I'm testing mod_verto. After a while (5-10 minutes) it always disconnecting. I do nothing with FS (its sleeping) It seems to be normal shutdown 2014-11-03 08:50:30.495346 [INFO] mod_verto.c:4130 Secure key and cert specified 2014-11-03 08:50:30.495370 [INFO] mod_verto.c:4218 mine Bound to 0.0.0.0:8081 2014-11-03 08:50:30.495377 [INFO] mod_verto.c:4218 mine Bound to 0.0.0.0:8082 -- 2014-11-03 09:00:29.211502 [INFO] mod_verto.c:4524 profile mine shutdown, Waiting for 0 threads 2014-11-03 09:00:29.211502 [INFO] mod_verto.c:4535 mine Thread ending I attached a detailed log Best regards, Balázs SÁNDOR Software Developer Virtual Call Center MUNICH <http://virtual-call-center-software.de/> | BUDAPEST <http://virtual-call-center-software.hu/> | WARSAW <http://virtual-call-center-software.pl/> Phone: +44 (0) 863 801 69 Web: www.virtual-call-center.eu <https://www.linkedin.com/profile/view?id=215233036>

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Embedded Freeswitch
by Guillermo Ruiz Camauer 04 Nov '14

04 Nov '14

Can anyone point me to any examples / source code of embedde Freeswitch? I have looked at the documentation, and the different books, but can find now examples of this. Guillermo -- Guillermo Ruiz Camauer

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To decypt webrtc audio
by Varghese Paul 04 Nov '14

04 Nov '14

Hi all, We are using freeswitch for handling webrtc calls and the negotiation in DTLS mode. We are looking for a tool to decrypt the traffic between the webrtc client and freeswitch server. I know there is already a tool pcaputils from PJSIP for decrypting the SRTP traffic. Since we are using in DTLS mode we can't use this. Can any one suggest any tool/method to decrypt the webrtc audio? Regards Varghese Paul

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Good News Everyone! We can finally use HTML5, Oh yeah we already were!
by Ken Rice 04 Nov '14

04 Nov '14

New Post on freeswitch.org from anthm check it out at http://ift.tt/1xVsqdv Good News Everyone! We can finally use HTML5, Oh yeah we already were! After almost 15 years, we can finally get started on HTML6! Maybe it’ll be done in time for FreeSWITCH 3.0 http://ift.tt/1zGJkBV

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