similar to: system alias

* 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 --

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

Hi Kelvin, The audio bandpass setting is only configurable when the encoder is instantiated (eg: start of a Voip call ) , but you can change the bitrate anytime.For example if you can read incoming RTCP packets , you can check if there's reported packet loss , and then lower the bitrate accordingly.Yes, the app has to be aware of the packetloss ?percentage. Cheers,Dragos From: Kelvin

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, 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 Kelvin, You can use something like :opus_encoder_ctl(enc,OPUS_SET_BITRATE(bitrate));opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(bandpass)); bandpass is the audio bandpass?, eg: OPUS_BANDWIDTH_WIDEBAND . You will need to calculate the codec bitrate from the available network bitrate (by taking into account the size of the IP+UDP+RTP headers which is 40 bytes ) . By default the audio bandwidth

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---------------->

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 All! First, let me introduce myself, as this is my first post to the list (I've been lurking for quite some time now). My name is Matt Hardeman, and I work for a software development firm in Birmingham, AL. We are interested in the Asterisk PBX and it's various configurations first as an internal solution for our occasionally bizarre telephony needs, and eventually are interested

Hi all! I have a new Cisco 7960 that I am testing out with asterisk, but it is currently loaded with the CallManager firmware. The reseller has sent me the latest SIP firmware files to load onto the phone, but as is mentioned in the Cisco FAQ entry for converting a CallManager phone to SIP, there is a bug in the firmware presently on my phone which prevents it from being upgraded straight to

hi guys! i'm going to share a workaround for authentication from msn messenger, you have to change two lines in chan_sip.c msn messenger is known to look for the correct realm in authentication, therefore, change the realm in chan_sip.c, line 2061 and line 2910 (release 0.4.0) i hope the realm can be parsed from extensions.conf in the next release... ~kelvin =) -------------- next part

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, 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

Hello all We succesfully added a H323 Gateway to our CallManager 3.0 that resides in Mexico and were/are able to make calls from CallManager SCCP phones to the Asterisk Server phones in the U.S.; however, we have not been able to call from Asterisk server in U.S. to CallManager phones in Mexico Here is what we tried: 1. Adding a Gatekeeper into CallManager and then have Asterisk (and also

Hello everybody, im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager. We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux

I have a .csv table named mailing.csv as below. It consist a receiver, subject and sender. Receiver subject sender 1 Adrian Cole RE: [WHIRR-117] Composable services Tom White 2 Adrian Cole RE: [WHIRR-117] Composable services Tom White 3 Adrian Cole RE: [WHIRR-117] Composable services Adrian Cole 4 Adrian Cole RE: [WHIRR-117]

I have a Cisco Call Manager - and need to use the IVR Feature from Asterisk. My extension is 400 and I am calling 558 on Asterisk In my extension.conf I have these lines : exten => 558,1,Answer exten => 558,2,Playback(message.wav) exten => 558,3,Dial(SIP/439@CallManager) When I call 558 I heared the message then Asterisk tries to call 439 on CallManager but with this error :

given opus as a variable bitrate codec applied to voip rtp, i can verify that the bitrate really changes by a few kbps between max and min. as i understood, the bitrate variation is dependent on the audio source. are there any other factors which would affect this varying bitrate? like for example: packet losses, jitter, latency, etc. Will it automatically shift to lower bitrate / sampling rate

Hi guys, I have been reading a lot about the "adaptiveness" of opus and i quote: ... can still change, e.g. to adapt to changing network conditions. useinbandfec ... can somebody please enlighten me on this "adaptiveness"? whatever way I do our tests, it sticks to the same sampling rate and the same average bitrate, it would go up, down a bit but that's it. When we get