- FreeSWITCH-users - FreeSWITCH Mailing Lists

Help debug "Failed to write to Sangoma decoder RTP session"
by Antonio Silva 08 Feb '14

08 Feb '14

Hi, I have transconding card sangoma D100 and it works very well, but sometimes (1 or two times during the day on a server with 6calls/seconds) i get the error messages: " 2014-02-07 15:07:51.490715 [ERR] mod_sangoma_codec.c:517 Failed to write to Sangoma encoder RTP session. ae118c22-9d7a-4455-865e-1e9ea42ad487 2014-02-07 15:07:51.490715 [ERR] switch_core_io.c:1376 Codec Sangoma G729 encoder error! " What could be the cause for this problem? How can i debug it and try to fix it? Thanks, António

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BSD Maintainers needed...
by Brian West 08 Feb '14

08 Feb '14

FreeSWITCHers, Is anyone willing and able to maintain any of the BSD platforms? Currently you can not compile on Net/Open/FreeBSD with ease... OpenBSD and NetBSD are already well on their way to the unsupported due to no activity in the community to assist in maintaining these platforms. If you wish to assist in maintaining any or all of these platforms please let me know. Thoughts? Feedback? Feb. 12th we’ll be talking about this during our weekly call. Thanks, -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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Redirecting traffic - Freeswitch vs Kamailio
by Alex Lake 07 Feb '14

07 Feb '14

If I want to set up a SIP redirector for small call volumes (eg. <1/second), can Freeswitch do as good a job as Kamailio? The idea is that we get a call in from a PSTN-SIP gateway and then re-route to another box depending on the number being called. Just wondering if there's any difference in the signalling, which I presume is either a 302 or a SIP re-invite (but please correct me if I'm wrong!). At what kind of call volumes might one choose to switch to a more dedicated SIP proxy?

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sip_history_info value
by Giuseppe Cardone 07 Feb '14

07 Feb '14

Hi, I have FS 1.2.14 running and I'm testing it's behaviour in call forwarding scenarios. I'd like to include the SIP History-Info-Header as per RFC 4244 in forwarded calls, so I guess the sip_history_info channel variable should hold this data. However, when I show its contents in the fs_cli debugs by using the log action in the relevant part of the dialplan with: <action application="log" data="LOG SIP History info is '${sip_history_info}'"/> it appears to be empty: [DEBUG] mod_dptools.c:1595 Says SIP History info is '' Any clues as to how this variable should be used ? Thanks in advance, Giuseppe

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Generating RTP locally but timestamp passthru is configured, disabling....
by Carlo Dimaggio 07 Feb '14

07 Feb '14

Hi all, I have this warning on the fs_cli console. What could be the cause and what is the meaning? How I can disable timestamp passthru? Regards, Carlo Dimaggio

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Hangup by VAD timer
by Dmitriy Shumaev 07 Feb '14

07 Feb '14

Does anyone know how to use a VAD to hangup call on timer (To hangup call in which there is no sound/conversation for some time)? Thanks in advance, Shumaev D.A., porgrammer, KBR Ltd.

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Friday JIRA Work...
by Brian West 07 Feb '14

07 Feb '14

Remember we’ll be hanging out in 888 after the VUC call today, I want to see if we can do a sweep thru JIRA to clear any bugs/patches that have been sitting. Please join in… We have a lot of work to do. Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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a1-hash for reverse-auth-pass
by Yuriy Nasida 07 Feb '14

07 Feb '14

Guys, I think you all know there is perfect thing like a1-hash which allow us don't keep passwords as plain text.But what about reverse-auth-pass ? I really use this feature a lot. reverse-auth-pass should be equal a user's password as far as I know.How can I don't keep reverse-auth-pass as plain text as well ? Any thought?Thanks.

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FS <-> SKYPE with G729. how to set annexb=no ?
by Yuriy Nasida 07 Feb '14

07 Feb '14

Hi there! I know that it should be simple but when i add<action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/> nothing happens... I didn't see fmtp:18 annexb=no in SDR of B-led the callflow: Skype --> FS ----G729a---> VOIP Please advice Thanks.

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Skype <---> Freeswitch gateway. Skype <---> Freeswitch gateway. Where to take correct skype client at present ?
by Yuriy Nasida 07 Feb '14

07 Feb '14

Guys, I follow this manual http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#API_a… Script install.pl says : --2014-02-03 10:42:41-- http://mirrors.kernel.org/archlinux/community/os/i686/skype-oss-2.0.0.72-3-… request sent, awaiting response... 404 Not Found2014-02-03 10:42:42 ERROR 404: Not Found. Looks like this file was deleted :( Where I can take correct one for centos 5.10 and this manual ? I checked http://mirrors.kernel.org/archlinux/community/os/i686/ and just have found skype-4.2.0.13-1-i686.pkg.tar.xz But I am not sure that it is correct. Please advice.Thanks.

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