similar to: msn authentication

hi guys, do you have any suggestions on how to connect 2 TE410P via E1? (for simulation and testing purposes) asterisk1 --> TE410P ----> ? ---------> ? ---->TE410P -->asterisk2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030907/698cd499/attachment.htm

hi guys, have any of you guys managed to use msn messenger to authenticate with asterisk using its DNS name? based on my experience with other sip proxies, msn will not authenticate if it sees a different realm than the realm of the client. one workaround i did was to edit the chan_sip.c to send a pre-defined realm, and also edit the Contact: field. after this, asterisk would send a 401 to the

Thanks Dragos, I assume I will be setting those parameters during initialization of encoder right? Question is, if connection gets too lossy, how will opus adapt to it? Can it automatically shift bitrate down to minimize impact? Mark from IRC suggests that the app has to be aware of the losses and change it on the fly. Has anybody on the list tried this? Kelvin Chua On Wed, Mar 4, 2015 at 5:53

I am using libopus for my implementation. I wonder if anybody in the list have any experience on how to make libopus dynamically adjust its bitrate? On Mar 3, 2015 10:42 PM, "Benjamin Schwartz" <benjamin.m.schwartz at gmail.com> wrote: > It sounds like your software isn't adjusting the opus bitrate in response > to network conditions. For example, many WebRTC

hi guys, have anybody tried using audiocodes sip fxs against asterisk? how's the device fairing? ~kelvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030725/ae4b2f25/attachment.htm

hi guys, i'm using sept 30 cvs and oh323 5.5 i'm having 5 second latecy(on only 1 audio path) when a call is transferred.... the scenario is this: sip--------->asterisk----->h323:operator (who then transfers the call) ---------------->h323:destination ------------------audio path 5-second latency---------------->

* compiled from cvs, i am trying call ip phones in callmanager 3.2 10.17.0.2 is my callmanager i noticed from network dumps that instead of sending rtp to the ip phone, * sends it to 10.17.0.2! thereby causing no audio from * to ip phone. audio from ip phone to * is ok. only callmanager calls fail. netmeeting works ok... here is the debug, thanks for any info ~kelvin H323 debug enabled --

hi guys, i'm encountering one way audio on cvs using netmeeting and chan_h323.so is there a quick fix or workaround for this? compiled using openh323 1.12 pwlib 1.5 i also saw this in earlier version of openh323 and pwlib.... thanks for any info ~kelvin -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi michael, here are the core dumps. only kphone works when 0.5.5 and * cvs. audiocodes and msn messenger all cause seg faults when calling ccm thru * (or vice-versa) ~kelvin [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper

hi guys, i'm in need of several 24port or higher fxs device which supports sip, aside from mediatrix and audiocodes (cisco's vg248 doesn't support sip), do you have any idea who else manufactures such device? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030801/67eb12dd/attachment.htm

On 11.9, trying to set up a callcentric peer: sip debug: > <--- SIP read from UDP:204.11.192.161:5060 ---> > INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0 > v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d > f: <sip:<calling number>@66.193.176.35>;tag=3606475083-968127 > t:

hey, I bought a license for 729 but I can't use it this is the message. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs!

Can I say that RPART is a modified algo of CART and PARTY a modified of CHAID? Thanks. ---- Chua Siang Li Consultant - Operations Research Acceval Pte Ltd Tel: 6297 8740 Email: siang.li.chua at acceval-intl.com Website: www.acceval-intl.com This message and any attachments (the "message"...{{dropped:12}}

Hello there. I have fitted a rpart model. > rpartModel <- rpart(y~., data=data.frame(y=y,x=x),method="class", ....) and can use rpart$where to find out the terminal nodes that each observations belongs. Now, I have a set of new data and used predict.rpart which seems to give only the predicted value with no information similar to rpart$where. May I know how

The time has come to review the current Contributor and Core contributor grants for ZFS. Since all of the ZFS core contributors grants are set to expire on 02-24-2009 we need to renew the members that are still contributing at core contributor levels. We should also add some new members to both Contributor and Core contributor levels. First the current list of Core contributors: Bill

Hello there. Is there any function in R that can do cluster on a set of data that has both categorical and numerical variables? thanks. siangli

- This avoids problematic "reloc'ed while mapped" messages and some associated corruption as well. Signed-off-by: Maarten Maathuis <madman2003 at gmail.com> --- src/gallium/drivers/nouveau/nouveau_screen.c | 21 +++++++++++++++++++++ src/gallium/drivers/nouveau/nouveau_screen.h | 3 +++ src/gallium/drivers/nouveau/nouveau_stateobj.h | 13 +++++++++++++

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Hi. I need some advises on how to use R to find pi (i is the index) with the following objective function and constraint: max (sum i)[ f(ai, bi, pi) * g(ci, di, pi) * Di ] s.t. (sum i)[ f(ai, bi, pi) * Di * pi] / (sum i)[ f(ai, bi, pi) * Di ] <= constant f and g are diffentiable. So, I am thinking of optim with method = "BFGS"? But wonder how to include the

Hello there. I uses the following codes for the purpose of variable selection. > lmModel <- lm(y~.,data.frame(y=y, x=x)) > step <- stepAIC(lmModel, direction="both") > step$anova Stepwise Model Path Analysis of Deviance Table Initial Model: y ~ x.Market.Price + x.Quantity + x.Country + x.Incoterm + x.Channel + x.PaymentTerm Final