search for: tos_audio

Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using

...ol=udp bind=0.0.0.0 tos=af31 [pbxbeta] type=endpoint disallow=all allow=g722 allow=ulaw transport=transport-lan context=phone-level3 aors=pbxbeta send_rpid=no send_pai=yes trust_id_inbound=yes trust_id_outbound=yes direct_media=yes direct_media_glare_mitigation=outgoing ;direct_media_method=update tos_audio=46 tos_video=34 t38_udptl=no t38_udptl_nat=no [pbxbeta] type=aor contact=sip:{remote IP address}:5060 [pbxbeta] type=identify endpoint=pbxbeta match={remote IP address} The phones have the following set in pjsip.conf (snippet): type=endpoint disallow=all allow=g722 allow=ulaw transport=transpor...

...his out yet? Lots of places say to add the following to sip.conf of 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&g...

...not be! Please check UPGRADE.txt for correct configuration settings. Of course, i checked UPGRADE.txt, and lot of other resources, enabled few settings in sip.conf, but this still doesn't change. my sip.conf is: [general] port = 5060 bindaddr = 0.0.0.0 context = default-external tos_sip=0x18 tos_audio=0x18 callerid = Unknown dtmfmode=rfc2833 ignoreregexpire=yes limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes call-limit=1 and the corresponding realtime entry is: name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: device <21168> canreinvite: no contex...

...: bytes=32 time=36ms TTL=51 Reply from 209.251.157.93: bytes=32 time=28ms TTL=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 8...

Hello list, what can I do to minimalize the jitter in SIP-calls at server level ? If at local network level, there is a VoIP-router and their is a physical network dedicated to IP-phones, but there is still jitter. When using a Hosted Asterisk server, which settings on the Asterisk-server can minimalize the jitter between the VoIP-router and the Asterisk-server on the public internet ?? Kind

...[general] srvlookup=yes ;allows DNS lookups of server names naxexpirey=180 defaultexpirey=160 context=default ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) tos_sip=cs3 tos_audio=ef ; bindport is the local UDP port that Asterisk will ; listen on bindaddr=192.168.xx.xx ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls limitonpeers=yes notifyringing=yes rtupdate=yes[authentication] [104] type=peer context=p...

...sippasswd md5pwd rpid domain testers.com sippasswd2 and my sip.conf: [general] bindport = 5070 bindaddr = PU.BL.IC.IP udpbindaddr = PU.BL.IC.IP tcpenable = yes limitonpeers = yes rtcachefriends = no tos_sip=cs3 tos_audio=ef realm = testers.com autodomain=yes domain=PU.BL.IC.IP domain=testers.com transport=ws,wss,udp outboundproxy=PU.BL.IC.IP:5060 I'd appreciate Your advice. cheers, Olli -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/aster...

...ow is how to configure sip.conf, so far I've just been making guesses based on online examples and documentation. My current sip.conf looks like this: [general] bindport = 5070 bindaddr = 127.0.0.1 tcpbindaddr = 127.0.0.1:5070 tcpenable = no limitonpeers = yes ;rtcachefriends = yes tos_sip=cs3 tos_audio=ef realm = testers.com I've tried defining realm and domain values, but I lack proper understanding of those. Can you guys help me out? Are there any other configurations I need to check? Respectfully, Olli -------------- next part -------------- An HTML attachment was scrubbed... URL: <ht...

...ute match = 147.75.60.160/28, 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 al...

...ate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default nat = yes notifyringing = yes pedantic = no progressinband = never promiscredir = no realm = asterisk recordhistory = no registerattempts = 0 registertimeout = 20 relaxdtmf = no sendrpid = no sipdebug = no t1min = 100 t38pt_udptl = no tos_audio = none tos_sip = none tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = ulaw,gsm subscribecontext = device-hints register => 6087294351:<sip password>@sip.broadvoice.com [trunk_1] type=peer user=phone host=sip.broadvoice.co...

...n via another internet 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 [USE...

...atures.conf as follows: [parkinglot_a100] ; SSI context => a100-parking parkpos => 900-920 findslot => next [parkinglot_a10] ; EBC context => a10-parking parkpos => 800-820 findslot => next I then set the parkinglot parameters in sip.conf as follows: [general] context=incoming tos_audio=0xb0 tos_sip=0xb0 disallow=all allow=ulaw allow=alaw allow=speex allow=gsm canreinvite=nonat [common](!) type=friend host=dynamic [a10](!,common) ; EBC context=a10 parkinglot=parkinglot_a10 [a10o](!,common) ; EBC - authorized for outbound calls context=a10o [jintrabartola](a10o) callerid=Joe In...

...re? Here is my sip.conf: [general] subscribecontext=sip-blf context=default disallow=all allow=alaw allow=ulaw allowguest=no tcpenable=no tlsenable=no srvlookup=no localnet=192.168.56.0/255.255.255.0 localnet=192.168.57.0/255.255.255.0 tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. canreinvite=no dtmfmode = rfc2833 notifyringing=yes limitonpeers=yes callcounter=yes [basic-phone](!) type=friend context=from_internal_phones nat=no qualify=yes host=dynamic mohinterpret=default mohsuggest=default call-limit=20 callgroup...

On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks and visible to one another? If not the reinvite >>> may be passing an internal (nat'ed)

...elevant but I checked that Asterisk was successfully compiled with res_srtp module. Here's my sip.conf contents: bindport = 5070 ; using this since Kamailio is at 5060 bindaddr = PU.BL.IC.IP tcpenable = yes ;no limitonpeers = yes rtcachefriends = yes ; for realtime rtupdate=yes tos_sip=cs3 tos_audio=ef useragent=MyAsterisk realm = myrealm.com autodomain=no domain=PU.BL.IC.IP domain=testers.com allowexternaldomains=no allowguest=no avpf=yes encryption=yes transport=ws,udp icesupport=yes srvlookup=yes And here's an example of a ws client in my realtime peer table: id: 4...

...der for 1 phone number (2 channels). 8. 1 x IAX2 outgoing trunk (theoretically unlimited channels) to phone provider. 9. Internet connection used is ADSL 8mbs/350kbs. 10. All internet traffic goes through the Linux/Asterisk box (2 NIC's) and I use basic traffic shaping/priority using tos_sip/tos_audio/tos/cos directives in sip.conf and iax.conf. 11. The server doesn't have much load - some basic samba file sharing for 5 computers, some dovecot IMAP email for same machines. The setup has worked fine for it's first 8 months. After that, users started to report dropped calls. Most of t...

You have it right, for 1.2, use 'tos=', for 1.4 use 'tos_sip/tos_audio/tos_video'. ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Al lists Sent: Monday, July 23, 2007 10:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Fwd:...

...t i enabled everything necessary but something must be still missing. Here is my sip.conf (with changed IP-addresses of course): [general] port = 5060 bindaddr = 0.0.0.0 context = others ;nat specific stuff below externip = 2.3.4.5 canreinvite = no nat = yes qualify = yes ;qos stuff tos_sip = cs3 tos_audio = ef ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; ;template for voip-testing [testphones](!) context = voiptest type = friend host = dynamic nat = yes disallow = all ;disable all codecs except the ones defined below allow = ulaw allow = g723.1 allow = g729 ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;...

...y = yes secret = 123456 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...