search for: resyncthreshold

...X2, I have finally received almost acceptable quality. I am receiving 5-8 complaints a day of calls "breaking up" from both the customer and agent sides. What I have discovered is that in most of these cases, the new jitterbuffer performed a resync during the call. Currently, I have the resyncthreshold, and all other jb parameters at their default levels The traffic is running over a fairly high latency WAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping times run about 85ms. > I am interested to know why you are using ilbc, n why not g729 ot g723 or speex. What is the size of...

    Not these particular two servers. On 02/03/20 12:16, Doug Lytle wrote: >>>> I am trying to troubleshoot two Asterisk servers that have an IAX2 >>>> trunk between them. > Carlos, > > Had caller-id ever worked between these two systems? > > Doug > -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161

...g is IP based. My boss complains that many of the calls he holds with others has a bad quality. He also says that its not just him. My iax.conf file has: disallow=all allow=ulaw allow=alaw bandwidth=high jitterbuffer=yes dropcount=2 maxjitterbuffer=1000 maxjitterinterps=10 resyncthreshold=1000 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1 tos=lowdelay autokill=yes He complains of broken audio, muffled audio, and says compared to Skype its very poor, particularly during conference calls (zaptel meetme). Most of these would be SIP based within our server though, rather tha...

...callerid=yes disallow=all allow=alaw allow=ulaw allow=gsm host=my.super.duper.host username=my.super.duper.username secret=my.super.duper.secret context=sip qualify=500 qualifysmoothing=yes requirecalltoken=no trunk=yes jitterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=300 maxjitterinterps=100 resyncthreshold=1500 tos=ef cos=5 Doug

Hi, I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. The customer is connected via IAX2 to our softswitch. On the customer's end I have the following config in iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all

Ok this is my first post and I will try to keep it short. I have searched everywhere and haven't found an answer to my question I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the

So are there any plans for bridged line appearance support in Asterisk? The new Linksys SPA9000 supports it. A lot of other VoIP systems from Nortel, Sylantro etc. supposedly support it. Seems to me that Asterisk needs to get on the bandwagon or be relegated to call centers, specialized voicemail applications, and phone chat businesses. It's not needed for companies used to PBX's but