search for: dropcount

...nf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw notransfer=yes trunk=no (I have also tried trunk=yes and nothing for trunk=) jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 I have tried with jitterbuffer=no, and then rather then one-way-audio I get high packet loss until the connection settles back down. Any ideas on other things I can try?

...estions: - What is the sugested codec for such setup? Now I'm using ULAW, but realizing it may not be the best choice. Sometimes I can hear broken audio. Maybe speex is better choice? - Jitter buffer, yes/no? What are the suggested values. Currently I'm using these values: jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=20 jittershrinkrate=2 - Trunking? Is it reliable enough? Thanks for any hints. -- David

...... disallow=lpc10 ; Icky sound quality... Mr. Roboto. allow=gsm ; Always allow GSM, it's cool :) ; ; You can adjust several parameters relating to the jitter buffer. ; The jitter buffer's function is to compensate for varying ; network delay. ; ; All the jitter buffer settings except dropcount are in milliseconds. ; The jitter buffer works for INCOMING audio - the outbound audio ; will be dejittered by the jitter buffer at the other end. ; ; jitterbuffer=yes|no: global default as to whether you want ; the jitter buffer at all. ; ; dropcount: the jitter buffer is sized such that no more t...

In my IAX config file I have: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) delayreject=yes disallow=all allow=ulaw allow=gsm jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 notransfer=yes allanrobertson- 209.23.224.97 (D) 255.255.255.255 1207 OK (33 ms) Why is it running on port 1207?

...sterisk Remote Unix connection disconnected" but cannot find anything I can recognize. I checked my iax.conf and I don't have anything with an IP defined: [general] port=5036 bandwidth=low disallow=lpc10 ; Icky sound quality... Mr. Roboto. allow=gsm ; Always allow GSM jitterbuffer=yes dropcount=6 tos=lowdelay [guest01] type=user context=default auth=md5,cleartext secret=xxxxxxxxx; [target] type=user context=default deny=0.0.0.0/0.0.0.0 ------------------ end of file --------------------------- has anybody faced this problem? is there a method to get around it? what else could be causing i...

...signalling conversion. sip.conf... --- [sip-router-1.gradwell.net] context=sip-inbound type=peer host=sip-router-1.gradwell.net [sip-router-2.gradwell.net] context=sip-inbound type=peer host=sip-router-2.gradwell.net --- iax.conf... [general] bandwidth=high disallow=lpc10 jitterbuffer=yes dropcount=2 maxjitterbuffer=500 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1 tos=lowdelay --- when a call comes in, we dial something like this, in our dial plan: -- Executing Goto("SIP/213.166.5.134-118f5310", "sip-users|7770002|1") in new stack -- Goto (sip-us...

...d with ILBC. The iax.conf files on each end start like this: > [general] > trunk=no > notransfer=yes > iaxcompat=no > > bandwidth=low > > disallow=all > allow=ilbc > > jitterbuffer=yes > dropcount=3 > maxjitterbuffer=500 > maxexcessbuffer=150 > minexcessbuffer=40 > jittershrinkrate=1 Of course, perhaps the jitter buffer isn't to blame, but given that one side of the call sounds perfect, I can't think of anything else obvious that would cause this....

...ICE[5126]: chan_sip.c:5860 handle_response: Peer '6666' is now REACHABLE! Iax.conf: [general] ;port=5036 ;I commented this out because I discovered it is hard coded anyway. bindaddr=10.20.30.10 amaflags=default accountcode=default bandwidth=low disallow=all allow=gsm jitterbuffer=no ;dropcount=3 ;maxjitterbuffer=500 ;maxexcessbuffer=100 ;trunkfreq=20 tos=lowdelay authdebug=no register => username:password@xxx.xxx.xxx.xxx ;these are real values that I changed for security reasons [arlington] type = friend context = local callerid = auth = plaintext secret = password ;inke...

...CLI, I got messages that indicate many iax channel hung and I cannot do soft hangup to them :(. Here is my iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) tos=0x68 ; bandwidth=low jitterbuffer=yes dropcount=2 disallow=all allow=ilbc ;allow=g723.1 ;allow=g729 ;allow=ulaw ;allow=alaw ;allow=gsm mailboxdetail=yes .... the other settings on iax.conf are just iax2 account for trunk and personal use. So I cut them in order to save spaces... Perhaps it's a bug? I've found this http://bugs.digium.c...

...max bandwidth). We've no E1/T1 links, everything 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 th...

Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

...oto(testdtmf|s|1) iax.conf file -------------------------------------------------------------------------------------- ; iax.conf [general] ${INCOMING-USR}=SECRET-USERNAME ${INCOMING-PWD}=SECRET-PWD ${LIVEVOIP-SVR}=217.160.244.186 bandwidth=high disallow=lpc10 jitterbuffer=yes dropcount=2 maxjitterbuffer=500 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1 register => ${INCOMING-USR}:${INCOMING-PWD}@${INCOMING-SVR} tos=lowdelay [incoming] ; this is the incoming IAX provider type=user secret=ITS-SECRET deny=0.0.0.0/0.0.0.0 permit=217.160.244.186/255.255.255.0 context=a...

...this doesn't really matter as there's no buffer for SIP) as recorded by 'sip show peers' as about 70ms (but in reality ping I think is about half this). My current setup is : SIP and IAX clients --> my * --> providers via IAX (one 5ms away, one 80ms away) jitterbuffer=yes dropcount=5 maxjitterbuffer=100 maxexcessbuffer=45 If anyone can post any amplification on these settings apart from what's in iax.conf or their experiences or maybe some adjustments I should try that'd be really really helpful. Thanks!! Chris

...ace, the other is using ztdummy. IAX.conf on machine 1: [general] port=5036 ;iaxcompat=yes bandwidth=low disallow=ilbc disallow=lpc10 ; Icky sound quality... Mr. Roboto. allow=ulaw ;allow=gsm ; Always allow GSM, it's cool :) jitterbuffer=no trunkfreq=20 ;dropcount=3 ;maxjitterbuffer=500 ;maxexcessbuffer=100 ; tos=lowdelay register => lachnet@woodlane.lach.net register => lachnet@10.1.1.158 ; [woodlane] allow=ulaw ;allow=gsm type=friend jitterbuffer=no username=woodlane context=dialout host=dynamic trunk=yes trunkfreq=20 IAX.conf on machine2: [general...

...1.26 132040 %0 33.82 42.88 1904.79 36.62 131919 %0 0.00 47.04 1924.64 36.36 Using these numbers, and knowing that the max rtt will not happen very often how do the jitter settings below look? Does anyone have any recommendations to improve my call quality using the jitterbuffer? jitterbuffer=300 dropcount=1 maxjitterbuffer=500 maxexccessbuffer=20 -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

...s, for connecting to a isdn 30 bundle, is the Wildcard E100P the card to use ? Many Thanks John Pugh johnpugh@cambrian-printers.co.uk _SIP.conf_ [general] port = 5060 bindaddr = 0.0.0.0 disallow=all context = bogon-calls allow=ulaw allow=alaw allow=G729 jitterbuffer=yes dropcount=3 maxjitterbuffer=500 maxxcessjitterbuffer=100 callgroup=1 pickupgroup=1 [2000] type=friend username=2000 secret=2000 host=dynamic context=from-sip [2001] type=friend username=2001 secret=2001 host=dynamic context=from-sip...

...pting native bridge" line of the output) and am then able to talk both ways for 20 to 30 seconds and then the IAX phone appears off line. If I wait on the PSTN line for another 30-45 seconds, I get the Hungup lines in the trace. I've tried jitterbuffer=no and then jitterbuffer=yes with dropcount=3 maxjitterbuffer=500 maxexcessbuffer=100 (defaults in the iax.conf) and neither setting worked. What is wrong? TIA for any help!!!!!! Ben Here is the console output: -- Accepting AUTHENTICATED call from 192.168.1.103, requested format = 2, actual format = 2 -- Executing Dial(&quot...

...Name/Username Host Mask Port Status 2004/2004 192.168.199.69 (D) 255.255.255.255 4569 UNREACHABLE My iax.conf is: [general] bindaddr=0.0.0.0 disallow=all allow=ilbc allow=gsm allow=ulaw allow=alaw bandwidth=low tos=reliability jitterbuffer=yes dropcount=3 maxjitterbuffer=500 maxexcessjitterbuffer=100 [guest] type=user context=voipmenu [2004] type=friend secret=mysecret auth=md5 context=from-iax host=dynamic notransfer=yes trunk=no qualify=1000 canreinvite=no

...n FXO port through a zaptel channel. It wasn't obvious to me if this same process could be done on the IAX channels and if so how? For reference my iax.conf looks likes: [general] port=5036 bindaddr=0.0.0.0 bandwidth=low echocancel=yes jitterbuffer=yes maxjitterbuffer=500 maxexcessbuffer=100 dropcount=5 register => djohnson:12dookie@iaxtel.com tos=lowdelay #include /etc/asterisk/users/iax/iax_users [iaxfwd] type=user context=IAX_FWD deny=0.0.0.0/0.0.0.0 permit=65.39.205.0/255.255.255.0 Thanks for the help! dj

...d 'jitterbuffer=yes' but to fail. My system configuration is like this. PSTN<========>Asterisk(TDM/Fxo 4port*3)<=====LAN(IAX2)=====>Iaxclient library And iax.con is... ----------------------------------------------- [general] port=5036 disallow=all allow=gsm jitterbuffer=yes dropcount=8 maxjitterbuffer=500 maxexccessbuffer=100 tos=lowdelay [agent00] type=friend username=aaaaa secret=aaaaa context=default host=dynamic notransfer=yes callerid=22813333 ----------------------------------------------- Any idea to reduce call drop rate? How can i monitor or log dropped call? If log...