search for: icesupport

...chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:13 CN/8000... OK in sip.conf I have : icesupport = yes in rtp.conf I have : icesupport=true stunaddr=stun.ekiga.net sip peer has everything set for webrtc :   Realtime peer: Yes, cached   Prim.Transp. : WS   Allowed.Trsp : WSS   Codecs       : (alaw|g729|gsm)   Useragent    : SIP.js/0.10.0   Reg. Contact : sip:u79mer6v at 1u7hp86jdg...

...system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF for this peer icesupport=yes ; Tell Asterisk to use ICE for this peer ignorecryptolifetime=yes context=sameer ; Tell Asterisk which context to use when this peer is dialing ;directmedia=yes ; Asterisk will relay media for this peer transport=udp,ws ;Asterisk will allow this peer to register on UDP or WebSockets ;disallow=a...

...out authorization. For this reason, the previously working functionality of sending and receiving SMS from gateway GOIP had to rewrite their internal Protocol. - found hardphones and software phones that don't accept "long nonce" and refuse to register when using res_pjsip - enable icesupport also leads to problems of registration and cannot be "common solution" - issue tracker now contains multiple error messages that arise every day and reboot my server (which cannot be called a production) - And watchdog logs SegFaults and Hangs including other stacks that are not yet do...

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a): > Vinicius Fontes wrote: >> I'm having the same issue! The difference in my case is Asterisk server >> has a public IPv4 and the browser is behind a single NAT. >> >> I'm forwarding my configuration below (which I posted previously on >> asterisk-users). >> >> How can we debug ICE negotiation? >

...11] fullname = luca secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = myproxy host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgroup= dial=SIP/00493511111111 [00493512222222] fullname = fax secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = myproxy host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=forc...

...t of the "nonce" field. When using a longer nonce (pjsip) this phone simply does not respond to the request packet authorization (as do many hardware and software encountering something incomprehensible). The same behavior was on the built-in nokia 95 SIP client. > >> - enable icesupport also leads to problems of registration and cannot be >> "common solution" > > icesupport is only applied to calls, what happens for registration? Sorry. Not registration, but INVITE. The client software encounters an unfamiliar SDP headers and simply not responding to SIP pa...

...RTC - the browser can be a black box > at time and when things go wrong (like this) it's hard to dig and figure > out what is up. > Here is more information from the browser about the session: https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/af36iLlljtbYkbF On Asterisk I have icesupport=true in rtp.conf and ice_support=yes on the endpoint. I have configured a STUN server in both rtp.conf and res_stun_monitor.conf -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)8116-9161

On 10-08-16 08:52, Ludovic Gasc wrote: > > For WebRTC, I recommend you to use Asterisk 13+. > > Have a nice day. > > Ludovic Gasc (GMLudo) > http://www.gmludo.eu/ > > > Hello then why is there an option in sip.conf and rtp.conf " icesupport=yes" ?? This is no answer to my question. So again : what am I missing to get ICE support on my Asterisk 11.23.0 ?? Kind regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160810/eb1...

...riend nat=force_rport,comedia canrevinvite=no qualify=yes dtmfmode=rfc2833 context=home disallow=all allow=ulaw;h263 Can someone tell me if these settings are correct? I have no idea but it works now. I also made sure port 5060 and 5222 was open in iptables I also had to change rtp.conf to add icesupport=yes. I use my own rtp port range that is opened on the firewall. [general] icesupport=yes rtpstart=15000 rtpend=20000 ;rtpchecksums=no ;dtmftimeout=3000 ;rtcpinterval = 5000 ; Milliseconds between rtcp reports ; strictrtp=yes I also had to add icesupport=no in sip.conf[general]section to stop &...

...; hash323 = no > hasmanager = no > callwaiting = no > context = myproxy > host = dynamic > dtmfmode=rfc2833 > canreinvite=no > sendrpid=pai > type=friend > nat=force_rport,comedia > qualify=yes > qualifyfreq=60 > transport=Auto > avpf=no > force_avp=no > icesupport=no > encryption=no > callgroup= > pickupgroup= > dial=SIP/00493511111111 > > [00493512222222] > fullname = fax > secret = MYSECRET > dahdichan = 1 > hassip = yes > hasiax = no > hash323 = no > hasmanager = no > callwaiting = no > context = myproxy > h...

...06 ;replace with your Asterisk server public IP address or host transport=udp,ws,wss tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 [6000] host=dynamic secret=mysecret context=default type=friend icesupport=yes directmedia=no disallow=all allow=ulaw qualify=yes [6001] host=dynamic secret=mysecret context=default type=friend encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=no disallow=all allow=ulaw dtlsenable=yes dtlsverify=fingerprint dtlscertfile=/etc/asterisk/keys/asterisk.pem dtls...

hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side

...me = 00493511111111 secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = default host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup= pickupgroup= dial=SIP/00493511111111 [00493512222222] fullname = 00493512222222 secret = MYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = default host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=frie...

...v4 only just in case Asterisk doesn't like to see IPv6 ICE candidates. I try clicking to make an audio-only call from Jitsi. In the Asterisk logging (xmpp set debug on) I see the incoming "session-initiate" XML stanza but Asterisk does not send any XML back. I definitely have "icesupport=yes" in rtp.conf and I've tried it with and without specifying a TURN server from each end. Is this working for anybody?

...nd it's always used as outbound proxy bindaddr = PU.BL.IC.IP tcpenable = yes limitonpeers = yes rtcachefriends = yes rtupdate=yes tos_sip=cs3 tos_audio=ef realm = testers.com autodomain=no domain=testers.com allowexternaldomains=no allowguest=no ;avpf=yes ; encryption=yes transport=ws,wss,udp icesupport=yes srvlookup=yes nat=force_rport,comedia videosupport=yes directmedia=no And here's the way I've defined my websocket peer to my sippeers table: id: 4 name: 660 ipaddr: PU.BL.IC.IP port: 5060 regseconds: 1407744248 defaultuser: 660 fullc...

I need to get hooked up with this class, I could have students doing projects for homework :) Interested in RTCP? j On 6/26/23 7:45 PM, TTT wrote: > > I’m in training, so I have to demonstrate something SIP related.  I > figure it would be cool to hack a call, hanging it up while in > progress from outside Asterisk.  Doing so will demonstrate > use/knowledge of ARI, AMI, SIP,

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

...11.11.0 Here is my client sip config: [1060] type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=fee50 ; The SIP Password for SIP.js ;encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF for this peer icesupport=yes ; Tell Asterisk to use ICE for this peer context=default ; Tell Asterisk which context to use when this peer is dialing directmedia=no ; Asterisk will relay media for this peer transport=ws ; Asterisk will allow this peer to register on UDP or WebSockets force_avp=yes ; Force Asterisk to use av...

...----- -------------------- id 4 type friend name 660 host dynamic secret encryption yes avpf yes icesupport yes <---- ICE is enabled ipaddr PU.BL.IC.IP port 5060 regseconds 1410185500 defaultuser 660 fullcontact sip:660 at PU.BL.IC.IP:5060 lastms 0...

Am 23.06.2020 16:22, schrieb Marek Greško: > It seems your problems lie in something other. Most probably it is not > mtu problem. All my suspections are contradicted. If it is true you > have inter vlan voice quality problems, it is definitely something > different. Formerly I assumed you were trying only LTE vs LAN using > internet. I'm not sure what you mean with the last