search for: rtptimeout

Hi Since switching from 1.6.x to 1.8.4 I have noticed the following 1. When you do a 'core show channel <channel name>' the resulting information only shows data for "Frames In" , "Frames out" is always 0. 2. The rtptimeout option in the sip.conf no longer seems to work. I have this set to 60 seconds but have had channels which have not timeout when the rtp stops. If I subsequently do a "channel request hangup" then the CLI reports that there was a rtptimeout but will be hugely over the set amount. For insta...

Dears; I am facing a problem in disconnecting the calls, it is related to the rtptimeout (disconnecting if there is no RTP packets from both sides). My Asterisk Box is behind NAT but there is a static real IP address at the ADSL router. We call from the Mobile to the PSTN analogue numbers which are connected to Asterisk Analogue card (the telephone lines are analoge), and then we dial...

Hi Folks, Since I upgraded my asterisk box from 1.2.x to 1.4.x (1.4.10.1 now) I noticed some dead calls "apparently" running for more than 8 hours. I'm using rtptimeout=60 and rtpholdtimeout=120 and found some log messages like this: chan_sip.c: 'SIP/XXX-085a9308' will not be disconnected in 61 seconds because it is directly bridged to another RTP stream I can kill that calls using 'soft hangup <channel>' but I'd like to know if its a n...

...r H323 log file ;Default - /var/log/tvcti/h323_log ;logfile=/var/log/tvcti/h323_log ;Following values apply to all users/peers/friends defined below, unless ;overridden within their client definition ;Sets default context all clients will be placed in. ;Default - default context=from-h323 ;Sets rtptimeout for all clients, unless overridden ;Default - 60 seconds rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;Type of Service ;Default - none (lowdelay, thoughput, reliability, mincost, none) tos=lowdelay ;amaflags = default amaflag...

...323 log file ;Default - /var/log/asterisk/h323_log logfile=/var/log/asterisk/h323_log ;Following values apply to all users/peers/friends defined below, unless ;overridden within their client definition ;Sets default context all clients will be placed in. ;Default - default context=default ;Sets rtptimeout for all clients, unless overridden ;Default - 60 seconds ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;Type of Service ;Default - none (lowdelay, thoughput, reliability, mincost, none) ;tos=none ;amaflags = default ;The acc...

...ening with me that from time to time, I find some DAHDI channels are stayed connected (stuck) for long time. I know how to write the extensions.conf in a way to handle the hangup properly, also I send the incoming calls to the voicemail to be sure it is hanged up properly. One more thing, I set the rtptimeout in case there is any problem in the sip phone and the network .. But, still after sometime, I am surprised that some channels are stuck and stayed connected and then I have to reset it manually !! This is happening only in the analoge channels. What other than the rtptimeout, the hangup in the ext...

...23 log file ;Default - /var/log/asterisk/h323_log ;logfile=/var/log/asterisk/h323_log ;Following values apply to all users/peers/friends defined below, unless ;overridden within their client definition ;Sets default context all clients will be placed in. ;Default - default context=default ;Sets rtptimeout for all clients, unless overridden ;Default - 60 seconds ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity &nb sp; ; when we're not on hold ;Type of Service ;Default - none (lowdelay, thoughput, reliability, mincost, none) ;tos=lowdelay ;amaflags = de...

...at the sip debug stuff, and all I can see is my asterisk sending the registration packets, but no answer is received. Here's the relevant parts of my sip.conf: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes tos=0x18 checkmwi=10 videosupport=yes allow=all relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 register => XXXXX:pass@sipgate.de/XXXXX ;XXXXX == sipgateid [XXXXX] type=friend insecure=very nat=yes username=XXXXX fromuser=XXXXX fromdomain=sipgate.de secret=pass host=sipgate.de qualify=yes -- Michiel van Baak michiel@vanbaak.info http://michiel.vanbaak.info GnuPG...

...es ;tos=184 ;tos=lowdelay ;maxexpiry=3600 ;defaultexpiry=120 ;notifymimetype=text/plain ;checkmwi=10 ;vmexten=voicemail ;videosupport=yes ;recordhistory=yes disallow=all allow=g729 allow=gsm allow=ulaw jitterbuffer=yes maxjitterbuffer=1500 ;allow=ilbc ;musicclass=default ;language=en ;relaxdtmf=yes rtptimeout=60 ;rtpholdtimeout=300 ;trustrpid = no ;sendrpid = yes ;progressinband=never ;useragent=Asterisk PBX ;promiscredir = no ;usereqphone = no dtmfmode = rfc2833 ;compactheaders = yes ;sipdebug = yes ;subscribecontext = default ;notifyringing = yes And these are the extensions: [xxxx] type=friend ho...

...ee if is logged and can be called. Overall i want only lagged users to reregister and users with good response time to be check from time to time. defaultexpiry = 900 defaultexpirey = 900 maxexpiry = 300 maxexpirey = 300 minexpiry = 60 registerattempts = 5 registertimeout = 5 rtpholdtimeout = 900 rtptimeout = 60 jbmaxsize = 60 jbresyncthreshold = 200 qualify = yes qualify = 600 qualifyfreq = 60 Thank you. P.S. If you consider that i use too much options you can tell me what to drop. I use asterisk 1.8.6.0. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lis...

Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [10000] type=friend secret=test host=dynamic nat=yes -------------------------- ----- extensions.conf: ----- [common] exten => 501,1,MeetMe(12,MI) exten => 501,n,Hangup() exten => i,1,Hangu...

...pickupgroup | varchar(10) | YES | | NULL | | | port | varchar(5) | | | | | | qualify | varchar(4) | YES | | NULL | | | restrictcid | char(1) | YES | | NULL | | | rtptimeout | char(3) | YES | | NULL | | | rtpholdtimeout | char(3) | YES | | NULL | | | secret | varchar(30) | YES | | NULL | | | type | varchar(6) | | | | | | username...

...xt=incoming-bogus-calls bindport=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) maxexpirey=3600 ; Must be larger than the re-register timeout on the router defaultexpirey=3600 notifymimetype=text/plain rtptimeout=60 rtpholdtimeout=300 disallow=all allow=ulaw ; ; This section is because i'm behind nat ; register=>6477235412:<mypassword>@sip.unlimitel.ca/6477235412 externip=<mystaticIPaddress> ;Outside address localnet=192.168.0.148/255.255.255.0 ;Inside Network ; ;************************...

There is one question about re-Invite. Is it possible to carry out operation corresponding to draft-ietf-sip-session-timer -14? Ichiro Nakata i-nakata@nttpc.co.jp

Does the System() dialplan application have a limit on how long it can run? Either a time limit, or server load limit? I'm trying to pipe the output of Sphinx2 into Text2Wave, but Asterisk just runs by it to the next extension priority, with no errors. If I run the same command via the system shell, all is good, though it does take a few seconds, probably about 5 seconds to run. Yes,

I have a couple asterisk boxes, running sip between both boxes. 1.4.43 on both. both are installed from source, both have default settings, My config for one box is: [devgeis] type=friend defaultname=devgeis username=devgeis secret=yes disallow=all allow=ulaw allow=alaw allow=gsm rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 host=192.168.1.8 context=panel The other box is the same. There are times when "sip show peers" has Unspecified like: devgeis/devgeis (Unspecified) D a 0 Unmonitored So the registration is lost. But...

...323 log file ;Default - /var/log/asterisk/h323_log ;logfile=/var/log/asterisk/h323_log ;Following values apply to all users/peers/friends defined below, unless ;overridden within their client definition ;Sets default context all clients will be placed in. ;Default - default context=default ;Sets rtptimeout for all clients, unless overridden ;Default - 60 seconds ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;Type of Service ;Default - none (lowdelay, thoughput, reliability, mincost, none) ;tos=lowdelay ;amaflags = default ;The...

I have a problem with calls not hanging up if for some reason the physical phone dies or gets unplugged I can demonstrate this in practice by making a call from a handset, then unplugging the handset from the power. The call remains active and asterisk never seems to disconnect it. More annoyingly when power is re-applied the handset comes back to life, won't receive incoming calls

...ing based on the DID dialed and am hoping someone on the list can assist me. Here's the relevant info: Ingress SIP trunk: IP: 123.45.45.3456 DID's XXX-XXX-XX00-XX10 sip.conf: [general] useragent=Asterisk port=5060 context=default tos=lowdelay disallow=all allow=ulaw allow=alaw allow=gsm rtptimeout=300 rtpholdtimeout=600 // // My thought in this context is I will grab any incoming SIP call from the IP address of my SIP trunk and pass it to my sip-defaul-in context // // in extensions.conf [sip-default-in] type=friend defaultip=123.45.3456 host=123.45.3456 nat=no insecure=very context=sip-de...

We have had 20 calls over the last month where the SIP channel has not identified that the person on the receiving end has hung up. Is there a way of fixing this ? TIA Julian