search for: tos_video

....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=transport-lan send_rp...

...ip.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> <Ethernet> <RTP q...

...me=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 802.1p priority for RTP audio packets. ;cos_video=4...

....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 allow = !all,ulaw...

...piry = 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.com fromdomain=sip.broadvoice.com...

...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 [USERNAME] deny=0...

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: Asterisk a...

...t = 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 at gmail.com...

...ave a big 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 ho...

...s = no canreinvite = no checkmwi = 10 compactheaders = no defaultexpiry = 120 domain=sop-korniychuk domain=172.30.8.13 domain=172.30.8.13:5060 dumphistory = no externrefresh = 10 g726nonstandard = no notifyringing = yes srvlookup = yes 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 = alaw type = friend host=dynamic context = noop-context dtmfmode=rfc2833 nat=no rtcachefriends=yes qualify=10000 deny=0.0.0.0/0.0.0.0 permit=172.30.8.0/255.255.255.0 regards, Yaroslav

Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:

...; ; custom 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> chan...

...######################################## 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 inse...

HI ALL got small question i use call-limit=1 on peers but call limit is not working if user is not registered on PBX and making calls so the main question is -- how to Disallow CALLS without registering on PBX -- Best regards Antony tel. +380669197533 tel2. +380636564340 Paypal http://paypal.me/Satskiy

...l_group= t38_udptl_maxdatagram=0 media_encryption_optimistic=false aors=DEADDEADBEEF rpid_immediate=false outbound_proxy= identify_by=username inband_progress=false rtp_symmetric=false transport=transport-udp rtp_keepalive=0 t38_udptl_ec=none fax_detect=false t38_udptl_nat=false allow_transfer=true tos_video=0 srtp_tag_32=false timers_min_se=90 call_group= sub_min_expiry=0 100rel=yes direct_media=true rtp_timeout_hold=0 g726_non_standard=false dtmf_mode=rfc4733 voicemail_extension= rtp_timeout=0 dtls_cert_file= media_encryption=no media_use_received_transport=false direct_media_glare_mitigation=none tr...

...e-chrome-stable-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/asteri...

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)

...load and the local Mitel vendor can't seem to get his hands on anything newer than 6.0.0.something, though there is supposedly 7.1.x available. These phones are running 06.00.00.19. The Asterisk server has a pretty standard sip.conf, bindaddr=0.0.0.0 pedantic=no; bindport=5060 srvlookup=no tos_video=af41 notifyringing=yes notifyhold=yes allowsubscribe=yes limitonpeer=yes localnet=172.16.1.0/255.255.255.0 Polycom phones on this same asterisk server do not display this behavior. I'm wondering if there is a workaround for this apparent Mitel issue in Asterisk's configuration. Anyone u...