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On Tue, 11 Oct 2005 10:36:51 +0200, Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca> wrote: > Source code at: > http://people.xiph.org/~jm/speexclient/ I rewritten my program so it is more similar to yours, I grabbed your code for storing and retrieving echo, and I have better results, I think the echo is cancelled in about 50% but it still can be heared. I think there is a

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm

Concerning the VAD in SPEEX, I've been using "VoIPerized" a phone using SPEEX sampling at 16K, I use a Plantronics USB headset. I find that I have to set the VAD quite sensitive in order not to lose "S's" from the start of a word/sentence. the result is that I'm often sending packets unintentionally resulting in packet loss from the incoming stream(usually caused

Hi, I saw there is a new release of speex. What is this SPEEX_PLC_TUNING option for? And is there a more complete list of changes? Because I like to determine if it is interesting to update to 1.1.8 Greetings Jeroen de Kleijn (developer of VoIPerized)

Hi, I implemented the echo cancellation from speex in my Windows VOIP-client VoIPerized --> www.voiperized.com Greetings Jeroen de Kleijn -----Original Message----- From: speex-dev-bounces@xiph.org on behalf of hs Sent: Tue 10/11/2005 4:02 PM To: speex-dev@xiph.org Cc: Subject: Re: R: [Speex-dev] echo cancellation On Tue, 11 Oct 2005 10:36:51 ?, Jean-Marc Valin

Hi, Ok, what you suggest sound logical to me. Currently, I have done a small trick to prevent this problem. What I did is that whenever windows request a voice packet from me and if I do not have the voice packet, I repeat the previous packet. Hence, all the breaking portion is filled with previous packet. This trick seems to work so far. I am not sure what is the side effect. I think jitter

Hello fellows VOIPer, If you want to share with the rest of the VoIP & RTC community some news, interesting or breaking through ideas, or even more, some experience you had in terms of designing, integrating or operating various solutions or platform based on Open Source Softwares, then you should consider submitting a paper for the OpenSIPS Summit 2020 in May, Amsterdam.

Hi, When an audiopacket is received too late I could interpolate this frame. The problem is that I don't know if it is a true bufferloss or just the last audiopacket of a talkspurt. Now my question is if it's harmfull for the audioquality that at the end of the talkspurt one frame is interpolated? Or would this be almost inpossible to hear since the last audiopacket in the talkspurt

> Ok, what you suggest sound logical to me. Currently, I > have done a small trick to prevent this problem. What > I did is that whenever windows request a voice packet > from me and if I do not have the voice packet, I > repeat the previous packet. Hence, all the breaking > portion is filled with previous packet. This trick > seems to work so far. I am not sure what is the

I tried this http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html But I am NOT getting call in asterisk. SIP.conf file : _________________ [general] port = 5060 bindaddr = 0.0.0.0 context = default externhost=59.160.44.21 localnet=192.168.0.2/255.255.255.0 ; register SIP account on remote machine if using SIP trunks ; register => testSIPtrunk:test at 10.10.10.16:5060 ;

Hello all, Can I execute linux command like(ls, mkdir) through the Manager interface? I can't seem to access the manual at digium.com. I keep getting 'Forbidden' error. Looks like they are upgrading or something. CF

Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira