- FreeSWITCH-users - FreeSWITCH Mailing Lists

turn off cdr
by Madovsky 26 Apr '11

26 Apr '11

is it possible to deactivate completly any cdr without any side effect ? Thanks

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variable direction
by Boris Kovalenko 26 Apr '11

26 Apr '11

Hello! I found that with gateways I may use <variable .... direction="..."/>. Is this possible with regular (directory) users too? If not, may somebody explain why? -- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005

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presence in profiles
by Madovsky 26 Apr '11

26 Apr '11

Is it a problem to turn off presence in a FS cluster environment ? Thanks

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config ramdisk on CentOS 5.5
by fieldpeak 26 Apr '11

26 Apr '11

i'm trying tuning the FS to max performance on centos 5.5, and referred to http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#Free… #1, i configure the DB of FS to ramdisk , when i run "tmpfs /opt/freeswitch/db tmpfs defaults 0 0", it output: "-bash: tmpfs: command not found" #2, i run "ethtool -g eth0", the output is below, what value i should config for RX and TX for max performance... Ring parameters for eth0: Pre-set maximums: RX: 4096 RX Mini: 0 RX Jumbo: 0 TX: 4096 Current hardware settings: RX: 256 RX Mini: 0 RX Jumbo: 0 TX: 256 Apprecited if anyone help how to configure it... thanks! Regards, Charles

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FreeTDM disconnect supervision
by Luis F Urrea 25 Apr '11

25 Apr '11

Hello, According to what I have found in regards the tones used for signaling on FreeTDM, it seems that anything set through DAHDI is ignored and specs from tones.conf are used instead. However I have not been able to properly detect a busy tone to be able to set an FXO back on hook once a busy tone is sniffed. tones.conf has the following references for busy tones [us]: generate-busy => v=-7;%(500,500,480,620) detect-busy=> 480,620 I would like to know if detect-busy takes into account the time frame that the frequencies are on (played) or off (not being played) and if there is somehow a way to specify this. Is there any default value for time on and time off? I appreciate your help! Regards

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Forwarding Remote-Party-ID from Callee to Caller
by Sam Oredoyin 25 Apr '11

25 Apr '11

Hi All, I would like to pass the Remote-Party-ID field of the called party (sent by the callee in the Ringing & OK messages when answering the call) back to the originator's phone. I use the dialplan action ["set" data="sip_callee_id_name] as below but this only gives static information, we want to use the information passed back by the actual callee phone. The destination sends the Remote-Party-ID in the Ringing and OK replies, but they are not relayed to the original caller ("Outbound call" is sent instead). Is it at all possible to keep the SIP headers from the B-Leg of a call so we can pick the value from there, although it will be better if FS forwards the value rather than replacing it with "Outbound call". I know someone asked this question a couple of months ago but didn't seem to get the resolution. I'm running :> FreeSWITCH Version 1.0.head (git-4c435ec 2011-03-14 11-54-08 -0500) Regards Sam

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Re: [Freeswitch-users] g729 annexa/annexb interoperability
by Javier Gallart 25 Apr '11

25 Apr '11

Thanks David this is how Cisco sees the g729 issue: http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00800b67… IIRC the cng packets have 2 bytes payload, if an endpoint does not support annexb, what should it do about those packets? Regards Javi ---------- Forwarded message ---------- > From: David Ponzone <david.ponzone(a)ipeva.fr> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 25 Apr 2011 16:19:39 +0200 > Subject: Re: [Freeswitch-users] g729 annexa/annexb interoperability > Javier, > > I dont think there is anything to transcode. > G729 and G729B are the same, except the latter uses VAD and CNG. > > David Ponzone Direction Technique > email: david.ponzone(a)ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > > Service Client IPeva > tel: 0811 46 26 26 > www.ipeva.fr - www.ipeva-studio.com > > *Ce message et toutes les pièces jointes sont confidentiels et établis à > l'intention exclusive de ses destinataires. Toute utilisation ou diffusion > non autorisée est interdite. Tout message électronique est susceptible > d'altération. **IPeva** décline toute responsabilité au titre de ce > message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas > destinataire de ce message, merci de le détruire immédiatement et d'avertir > l'expéditeur.* > * > * > > > > Le 25/04/2011 à 13:44, Javier Gallart a écrit : > > Hello all > > we're using a Sangoma D100 card for transcoding. Our configuration, as far > as the codec policy is concerned, basically does this: > > -In the first offer to the b-leg we use the same codec list we receive > (<action application="export" > data="absolute_codec_string=${ep_codec_string}"/>) > -If the call fails with status 488, we repeat the call using all the codecs > available. This includes changing from annexb to annexa and viceversa. > > Let's suppose a-leg only supports g729 annexA, and B-leg only supports G729 > annexB. My understanding is that those different codec flavours don't > interoperate; in fact I've experienced many times audio problems when trying > to set up a call between endpoints supporting different g729 variants. > > In the described case, FS sends initially annexb=no, and B-leg rejects it > with cause 488. We rebuild the offer with annexb=yes, and then the offer is > accepted by the B-leg. Our concern is that no transcoding resources are used > in this case, and we might run into audio problems because of that. The > other concern is that in the answer to the A-leg (in 183 and 200), > annexb=yes is included. I'm not sure if all devices would support different > fmtp parameters in the offer and the answer (the RFC, as usual, won't be > explicit about this). > > Is there any way to force the transcoding in a situation like the one I > described? > > Thanks in advance > > Javi > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > >

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Community Thank Yous
by Michael Collins 25 Apr '11

25 Apr '11

Happy Monday to you all! I just wanted to take a moment to say thank you to the many folks who have been stepping up to help out with the not-so-glamorous side of running an open source project: documentation and community support. We all know that hacking code is the fun part, but writing documentation and helping out with IRC and mailing list questions are often thankless tasks. I wanted to personally say thank you to all those who've been stepping up. Regarding documentation, I've seen dozens of new users on the wiki, each of whom is adding something. This is indicative of people "paying the wiki tax" when a question is answered via ML or IRC. So to those who have been adding knowledge to the wiki: thank you! It is much appreciated. Please keep up the good work. In the IRC channel we've been having a good number of community members come in and make themselves available for answering questions. This can be very time-consuming, so again we say thank you. The same thing goes for the mailing list. It takes a lot of time to read through all the posts and respond. The following users have been especially active in helping others of late and I feel it is only fair to acknowledge their fine efforts. In no particular order... Ken Rice Steven Ayre Jeff Link Avi Marcus Kristian Kielhofner Steve Underwood Peter Olsson Stefan Knobloch Giovanni Maruzzelli Christopher Rienzo Every time a community member helps another community member we grow stronger. Please keep up the good work! -Michael

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conference_set_auto_outcall No Audio
by Chris Monkey 25 Apr '11

25 Apr '11

Hi, I am am relatively new to FS and have setup a home PBX with about 5 Snom 320s with TLS. Everything works fine, except when I try to use the mad_boss example, or any auto_outcall conference, the phones ring and will auto answer, but there is no audio. However, if I dial in to the conference, everything works fine. I've tried adding <action application="set" data="sip_secure_media=true"/> to the conference extension, but to no avail. I don't see any errors in the log, and I just did a make clean this afternoon. What am I missing? Thanks

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mod_enum compilation error
by mazilo 25 Apr '11

25 Apr '11

I have a local FS git pulled to adffe07f7512b00f7093091778965a119ca81513 and failed to compile mod_enum for OpenWRT as shown below: making all mod_enum make[7]: Entering directory `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/mod/applications/mod_enum' make[7]: *** No rule to make target `Makefile.am', needed by `Makefile.in'. Stop. make[7]: Leaving directory `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/mod/applications/mod_enum' make[6]: *** [mod_enum-all] Error 1 make[6]: Leaving directory `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/mod' make[5]: *** [all-recursive] Error 1 make[5]: Leaving directory `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src' make[4]: *** [all-recursive] Error 1 make[4]: Leaving directory `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git' make[3]: *** [all] Error 2 make[3]: Leaving directory `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git' make[2]: *** [/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/.built] Error 2 make[2]: Leaving directory `/opt/tmp/OpenWRT/feeds/packages/net/freeswitch_git' make[1]: *** [package/feeds/local/freeswitch_git/compile] Error 2 make[1]: Leaving directory `/opt/tmp/openwrt-svn-trunk' make: *** [package/freeswitch_git/compile] Error 2 ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-enum-compilation-error-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

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