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mod_h323 and mod_opal
by freeswitch.opencode＠spamgourmet.com 02 Nov '16

02 Nov '16

I'm working on packaging FreeSWITCH (including all modules) for Arch Linux, and I'm trying to figure out how to handle mod_h323. Following the pointers from src/mod/endpoints/mod_h323/compiling.txt, I see that it consumes code from this project: http://www.h323plus.org/ However, that project appears to be inactive; it hasn't had a release for multiple years (and forum/mailing list activity has nearly ceased as well). According to Wikipedia, the parent OpenH323 project (https://en.wikipedia.org/wiki/OpenH323) forked into H323Plus and Open Phone Abstraction Library (OPAL). >From the FreeSWITCH documentation, it looks like mod_opal consumes the code from one fork and mod_h323 from the other. The documentation also currently says mod_opal is beta quality, but that statement may be outdated (per the bugs file in its module directory). I'm packaging up FreeSWITCH for Arch Linux, which currently doesn't provide H323Plus. I could package up H323Plus for Arch, but if the project is inactive, I'm not sure if that's worthwhile. Is it fair to see the H323Plus project as defunct? If so, do you think FreeSWITCH will end up deprecating/removing mod_h323 in favor of mod_opal? Or is mod_h323 currently favored over mod_opal for production-quality H.323 support and thus important for a complete FreeSWITCH package to provide? Also, I thought this was a more dev-focused question, so I tried posting to freeswitch-dev over a week ago. The auto-reply said it was a moderated list and the message was being held pending moderator approval. It said that either the message would get posted or I would be notified of the moderator's decision, but nothing ever happened. Did this message get stuck in the pending moderator approval list? Thanks, David

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RTP timing issue
by Jan Blom 01 Nov '16

01 Nov '16

Hi list, I bridge a call with ptime 10 on one side and ptime 20 on the other. PCM codec on both sides, so no transcoding. The issue I face is that when a 20 ms packet arrives it is immediately sent out as two 10ms packets. And then nothing for 20ms until next packet arrives. Delta time between sent our packets are either 0 or 20 ms, never 10ms as one would expect. The result is choppy audio since we introduce jitter. Could this be avoided somehow? I am using Freeswitch 1.6.12 on CentOS 7.2. Best regards, Jan Blom

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New Chrome Extension for mod_verto
by Vitaly Kovalyshyn 31 Oct '16

31 Oct '16

Hi there! We've made the Chrome Extension & Application for FreeSWITCH mod-verto. It supports: Audio & Video calls via mod_verto Screen sharing Click-to-call in any web-page Address book Multiline Be free to download and test it: https://chrome.google.com/webstore/search/webitel <https://chrome.google.com/webstore/search/webitel> Best regards, Vitaly Kovalyshyn http://kovalyshyn.pp.ua/ Twitter: @kovalyshyn http://вебітел.укр/

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record DID called in DB
by jungle boogie 31 Oct '16

31 Oct '16

Hi All, With the example CDR sql here: https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/scripts/sq… Inbound destination calls will show the endpoints/call groups that rings, like this (destination_number) field: "2003" "2002" "2600" Is it possible to show the DID that was called as a separate record? example: "2003","12134123000" "2002","12134123000" "2600","12134123000" In this case, 12134123000 is a DID setup on the freeswitch server that's called and it rings extensions/call groups 2003, 2002 and 2600. I see destination_number is used on outbound calls to record what extension and/or DID was dialed. Thanks, sean

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testing again
by Ken Rice 30 Oct '16

30 Oct '16

Is it working?

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stash down?
by David Villasmil 30 Oct '16

30 Oct '16

Hello guys, I'm trying to push to stash but i keep on getting Service Temporarily Unavailable The server is temporarily unable to service your request due to maintenance downtime or capacity problems. Please try again later. ------------------------------ Apache/2.2.22 (Debian) Server at freeswitch.org Port 443 ᐧ

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What concepts in FreeSWITCH did you struggle with when you first started?
by Brian West 27 Oct '16

27 Oct '16

I would like to get community input on what concepts caused you the most headache and any input/ideas that would help me design some tutorials / videos that would help new users coming to FreeSWITCH. Your feedback will be greatly appreciated. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Can't build 1.6 on Jessie w/ SCTP
by Josh Jore 27 Oct '16

27 Oct '16

Hi, I just downloaded a fresh Jessie64 and failed building Freeswitch v1.6 with SCTP. Seems maybe SCTP just isn't supported right now or ...? Are there any configurations where this works? $ vagrant init debian/jessie64 $ vagrant up $ vagrant ssh Follow https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie ... $ git checkout v1.6 # https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/96eca8abb… $ ./bootstrap.sh -j $ ./configure --enable-sctp --prefix=... ... Making all in tport make[8]: Entering directory '/home/vagrant/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' make all-am make[9]: Entering directory '/home/vagrant/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' LTCOMPILE tport.lo LTCOMPILE tport_logging.lo LTCOMPILE tport_stub_sigcomp.lo LTCOMPILE tport_type_udp.lo LTCOMPILE tport_type_tcp.lo LTCOMPILE tport_type_sctp.lo tport_type_sctp.c: In function 'tport_sctp_init_socket': tport_type_sctp.c:206:10: error: variable 'initmsg' has initializer but incomplete type struct sctp_initmsg initmsg = { 0 }; ^ tport_type_sctp.c:206:10: error: excess elements in struct initializer [-Werror] tport_type_sctp.c:206:10: error: (near initialization for 'initmsg') [-Werror] tport_type_sctp.c:206:23: error: storage size of 'initmsg' isn't known struct sctp_initmsg initmsg = { 0 }; ^ tport_type_sctp.c:211:36: error: 'SCTP_INITMSG' undeclared (first use in this function) if (setsockopt(socket, SOL_SCTP, SCTP_INITMSG, &initmsg, sizeof initmsg) < 0) ^ tport_type_sctp.c:211:36: note: each undeclared identifier is reported only once for each function it appears in tport_type_sctp.c:206:23: error: unused variable 'initmsg' [-Werror=unused-variable] struct sctp_initmsg initmsg = { 0 }; ^ cc1: all warnings being treated as errors

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Running a script on a completed transfer
by Walter Smith 27 Oct '16

27 Oct '16

I need to record calls if/when they touch a certain set of extensions. I have most of the logic working but am baffled by one thing: how do I cause an application (it’s a lua script) to run when a channel is bridged as the result of a transfer? My usual bag of tricks isn't working because the bridge happens "magically" in sofia in response to the REFER. For example, execute_on_answer doesn’t help because the answer happens during the attended transfer -- I need to run AFTER the transfer (that is, when the original A leg is bridged to the attended transfer’s B leg). Example scenario: 1. External call comes in (channel A = PSTN to switch, channel B = switch to phone with extension 1000) 2. Ext 1000 does an attended transfer to ext 1001 (channel C = 1000 to switch, channel D = switch to 1001) 3. Transfer completes (sofia bridges channel A to channel D) I want my script to run on either channel A or D after 3 happens. What’s the best way to do this? Thanks, Walter

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ClueCon Weekly - Ken's mic volume
by Valter Nogueira 27 Oct '16

27 Oct '16

I feel that Ken's mic is a little low comparing to other's ones. It could not be a problem for native speakers, but it is a little hard for me to get what he is talking. Regards, Valter

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