April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_nibblebill updates
by Rogelio Perez 26 Apr '11

26 Apr '11

Hello, I'm using mod_nibblebill to bill my calls only for leg b and heartbeats set to 'off'. After the hangup I see 3 different MySQL updates from nibblebill instead of just one. This is my dial plan: <extension name="TEST"> <condition field="destination_number" expression="^(1\d+)$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="{nibble_rate=5,nibble_account=13}sofia/gateway/xxxxxxx/$1"/> <action application="hangup"/> </condition> </extension> ...this is the MySQL log: 68080 Query select 1 68080 Query UPDATE web_account SET balance=balance-0.343372 WHERE id='13' 68080 Query select 1 68080 Query SELECT balance AS nibble_balance FROM web_account WHERE id='13' 68080 Query select 1 68080 Query UPDATE web_account SET balance=balance-0.001177 WHERE id='13' 68080 Query select 1 68080 Query SELECT balance AS nibble_balance FROM web_account WHERE id='13' 68080 Query select 1 68080 Query UPDATE web_account SET balance=balance-0.001197 WHERE id='13' 68080 Query select 1 68080 Query SELECT balance AS nibble_balance FROM web_account WHERE id='13' ...and here's the FS log: http://pastebin.freeswitch.org/16175 The xml_cdr is saved after the second update, so the nibble_total_billed is different than the real total value debited from the account balance. Is this normal? Shouldn't nibblebill just do one update after the hangup?

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call not connecting sometime
by Philippe Le Toquin 26 Apr '11

26 Apr '11

Hello, I had the problem a few weeks ago but after a reinstall (unrelated to that) the problem had gone so I put it done as my messing up my system at the time! But today I had the same issue again. When I call sometime the call is not going through completely The symptom on my side are that nothing happens (no ring tone) on the other they say that the phones rings but when they pick up the phone they can't hear anything. Below is a siptrace where I changed the number and IP so I hope I didn't mess it up too much. http://pastebin.freeswitch.org/16165 Can someone let me know if they see something wrong? I tried to understand but it is beyond me :( Regards /Philippe

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Re: [Freeswitch-users] how to pass arbitrary headers from A leg to B leg when bridging
by Dave Horton 26 Apr '11

26 Apr '11

So I'm guessing this isn't possible without hacking the source code, which I've already done to solve my problem for now. But I'd like to make sure there isn't a better way of doing things, and thus I'd like to revise and restate my question for clarity. First, though, let me describe what I am doing, because I think it's a not-uncommon scenario that I think would be something that others may want to do. I basically want to use FS as a simple transcoding server between two endpoints, call them A and B. Calls coming in from A will be using speex codec and I want to send them out to B using PCMU; calls coming in from B will be PCMU and I want to send them to A using speex. The FS server will be a B2BUA and will be doing transcoding only -- no authentication (and no registration). Simple, right? The only fly in the ointment is that A is authenticating calls with B by providing a Proxy-Authorization header. So I need to take the Proxy-Authorization header received on the A leg and include it on the B leg. So far, the only way I have found to do that is to hack the code to create a new channel variable. I've done this, and it works. However, this leads me to the following questions 1) Is there a better way to do this? If there is no way to do it as a dialplan out of the box, can it be done as a script? 2) sofia has parsed all of the sip headers on the incoming invite for us, and they're all available from mod_sofia. Shouldn't those all be available to us (i.e., application developers) by some means (i.e., channel variables)? Dave

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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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