search for: af41

...L=51 Any suggestions or is this normal? Should I enable qos on my Cisco 3725 router and 2950 switch? Would I also need to enable the following in the sip.conf ;tos_sip=cs3 ; Sets TOS for SIP packets. ;tos_audio=ef ; Sets TOS for RTP audio packets. ;tos_video=af41 ; Sets TOS for RTP video packets. ;tos_text=af41 ; Sets TOS for RTP text packets. ;cos_sip=3 ; Sets 802.1p priority for SIP packets. ;cos_audio=5 ; Sets 802.1p priority for RTP audio packets. ;cos_video=4 ;...

...nux Uber Alles perl || die ...there are two types of command interfaces in the world of computing: good interfaces and user interfaces. - Dan Bernstein, Author of qmail PGP Fingerprint = 503D A72D AF41 2AD1 D433 C514 98D9 9A39 B25A 2405

...nux Uber Alles perl || die ...there are two types of command interfaces in the world of computing: good interfaces and user interfaces. - Dan Bernstein, Author of qmail PGP Fingerprint = 503D A72D AF41 2AD1 D433 C514 98D9 9A39 B25A 2405

...an Asterisk 1.2 system (current production machine/Asterisk as root): tos=0xB8 (Hex B8 = Decimal 184 = Binary 10111000) or if you are running Asterisk v1.4 or newer: tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos_video=af41 ; Sets TOS for RTP video packets. To match the current 1.2 machine would I set the Polycom's sip.cfg to the first or second QOS option? Option 1: ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ <QOS> <Ethernet> <RTP qos.et...

...8, 34.210.91.112/28, 34.226.36.32/28, 147.75.65.192/28 [flowroute] type = auth username = 12345678 password = XXZZXXZZXXZZ [flowroute] type = endpoint context = from-trunk dtmf_mode = rfc4733 allow = !all,ulaw direct_media = no from_domain = us-west-wa.sip.flowroute.com tos_audio = ef tos_video = af41 ; Note: "force_avp = yes" fixes issues with calls coming from MagicJack with no audio after a few seconds. force_avp = yes auth = flowroute outbound_auth = flowroute aors = flowroute t38_udptl = yes t38_udptl_ec = fec [anonymous] type=endpoint context = anonymous allow = !all,ulaw ------...

...link (ADSL 8bps connection) Router is preventing any traffic other than VoIP. for example we tried to pass HTTP requests via the internet link .. but did not go through. Asterisk Side: sip.conf sample: [GENERAL] notifyringing=yes notifyhold=yes limitonpeers=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes t38pt_udptl = yes bindport=5070 externip=SERVER_IP rtptimeout=60 session-timers=originate session-expires=600 session-minse=90 session-refresher=uas rtpholdtimeout=120 rtpkeepalive=20 allow=gsm t38pt_udptl=yes sendrpid=yes trustrpid=no directrtpsetup=yes [USERNAME] deny=0.0.0....

...type = friend username = 4111 disallow = all allow = alaw cancallforward = yes call-limit = 6 My general section of sip.conf : [general] qualify=yes context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=srvcentral.meudominio.com.br tos_sip=cs3 tos_audio=ef tos_video=af41 language=pt_BR rtptimeout=60 rtpholdtimeout=300 notifyringing = no notifyhold = no limitonpeers = yes nat=yes rtcachefriends=yes rtsavesysname=yes rtupdate=yes Registration is working fine, the only problem I can see is qualify. Anybody can help me ? Marcelo H. Terres mhterres at gmail.com ****...

...problem... i want to connect my asterisk server to a lancom 1722 device (ISDN/SIP) Gateway. sip.conf: [general] context=default allowguest=yes realm=10.1.1.209 bindport=5060 bindaddr=0.0.0.0 tos_sip=cs3 ; f?r SIP-Pakete (Kommunikationsaufbau) tos_audio=ef ; f?r RTP-Audio-Pakete tos_video=af41 ; f?r RTP-Video-Pakete allow=all dtmfmode=rfc2833 canreinvite=yes [3000] type=friend secret=3000 qualify=yes host=dynamic [lancom] type=friend context=fax-in secret=1000 username=1000 fromuser=1000 port=5060 i added a sip-line in my lancom, but it doesn't connect. does anybody know how to...

...nux Uber Alles perl || die ...there are two types of command interfaces in the world of computing: good interfaces and user interfaces. - Dan Bernstein, Author of qmail PGP Fingerprint = 503D A72D AF41 2AD1 D433 C514 98D9 9A39 B25A 2405

...son ************************what i have now************** HTB (6Mb) | GRED (1-12) using grio af11 -> gred vq 10 af12 -> gred vq 11 af13 -> gred vq 12 af21 -> gred vq 7 af23 -> gred vq 8 af23 -> gred vq 9 af31 -> gred vq 4 af32 -> gred vq 5 af33 -> gred vq 6 af41 -> gred vq 1 af42 -> gred vq 2 af43 -> gred vq 3 _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc

Hi,All I setup traffic control configuration with HTB this way: 1: root HTB qdisc | 1:1 HTB class rate 1024kbit | /-----+-----+-----+------+-----\ 1:10 1:20 1:30 1:40 1:50 1:60 EF AF41 AF31 AF21 AF11 BE and alloct different bandwidth to these PHBs(queues).So which tool would I use to generate these packets at the same to for testing?Thank you! Another question:I am studying sch_htb.c,but it's so tough for to understand,especially htb_dequeue().Would anyone pleas...

...iced that there''s always an ingress and egress of which these are 2 different dev. Is it possible for before ingress and egress be the same dev?? I tried the script below but doesn''t seem to work... How to control the bandwidth in this case?? I want to show that AF11, AF21, AF31, AF41, EF, BE traffic has different throughput if 6 fullstreams of traffic is push out from server to client (with 6 different ports). eg: #! /bin/sh -x # device="eth1" client0="192.168.1.1/32" client2="192.168.1.20/32" /sbin/ipchains -A output -i $device -p tcp -s $clien...

...us versions. */ #define DEFAULT_TOS_AUDIO 0 /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */ #define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */ @@ -1106,10 +1103,6 @@ static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */ static char global_regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */ static char global_useragent[...

...om modifications, details at: http://freepbx.org/configuration_files ; ;--------------------------------------------------------------------------------; ; vmexten=*97 faxdetect=yes context=from-sip-external callerid=Unknown notifyringing=yes notifyhold=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes useragent=FPBX-2.8.1(1.8.9.2) disallow=all allow=gsm allow=alaw allow=ulaw allow=g729 allow=g723 allow=g722 allow=speex I am using the originate command through the Asterisk console to test this. With plain SIP/1064, codec negotiation works as expected: elx2*CLI> channel o...

Hello list, I have in sip.conf : /maxexpiry=60 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) defaultexpiry=120 ; Default length of incoming/outgoing registration ;-----------------------------------------

...############################## And now my extensions.conf and sip.conf [general] allowoverlap=no allowguest=no bindport=5060 bindaddr=0.0.0.0 externip=189.38.242.109 localnet=192.168.20.0/255.255.255.0 srvlookup=yes disallow=all ;allow=g729 allow=ulaw allow=alaw tos_sip=cs3 tos_audio=ef tos_video=af41 regcontext=incoming_calls register=> 7977529 at sip.tellfree.net:PASSWD:7977529 at sip.tellfree.net/7977529 [tellfree] type=friend context=incoming_calls host=sip.tellfree.net username=7977529 authuser=7977529 authname=7977529 secret=PASSWD Fromdomain=sip.tellfree.net fromuser=7977529 insecure=...

Greetings, I am porting the openssh-portable 6.8 release to the HP NonStop (NSE) platform. Prior versions were no real problem, with minor tweeks. However, with the inclusion of regress/netcat.c, which depends on arpa/telnet.h, we have an issue. Unfortunately, the platform does not have this file, nor anything like it - telnet is done rather differently. We do have a version of netcat (0.7.1

...table-47.0.2526.111-1.x86_64) SIP.js 0.7.2 I set up my SIP configuration to have two SIP accounts. Account 1000 is the Linphone and 1001 is the webrtc: [general] faxdetect=no vmexten=*97 context=from-sip-external callerid=Unknown notifyringing=yes notifyhold=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes useragent=FPBX-2.11.0(11.20.0) disallow=all allow=g723 allow=ulaw allow=gsm allow=alaw allow=g729 allow=speex allow=g722 allow=h264 allow=h263p allow=h263 allow=h261 tlsenable=yes tlsbindaddr=0.0.0.0 tlscipher=ALL tlsclientmethod=tlsv1 tlscertfile=/etc/asterisk/keys/asterisk.pe...

...d dev eth0 parent 2:0 classid 2:1 htb rate 1Mbit ceil 1Mbit #create 13 gred''s tc qdisc add dev eth0 parent 2:1 gred setup DPs 13 default 13 grio #ef tc qdisc change dev eth0 parent 2:1 gred limit 512000 min 24000 max 32000 \ avpkt 1000 burst 40 probability 0.01 bandwidth 1024 DP 1 prio 1 #af41 tc qdisc change dev eth0 parent 2:1 gred limit 512000 min 24000 max 32000 \ avpkt 1000 burst 40 probability 0.04 bandwidth 1024 DP 2 prio 2 #af42 tc qdisc change dev eth0 parent 2:1 gred limit 512000 min 24000 max 32000 \ avpkt 1000 burst 40 probability 0.06 bandwidth 1024 DP 3 prio 3 #af43 tc qdis...

https://bugzilla.mindrot.org/show_bug.cgi?id=1964 Bug #: 1964 Summary: QoS/DSCP names false translated to ToS hex value Classification: Unclassified Product: Portable OpenSSH Version: 5.9p1 Platform: amd64 OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: ssh