search for: stunaddr

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? >

ekiga registration fails. I've set nat = yes ( also blank ) and i've set externip. Anybody have a sip.conf that works? Here's the sip debug: Reliably Transmitting (NAT) to 86.64.162.35:5060: REGISTER sip:ekiga.net SIP/2.0 Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport Max-Forwards: 70 From: <sip:test at ekiga.net>;tag=as64618445 To: <sip:test at

...ncing a "606 not Acceptable" error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. I tried putting stunaddr=stun.ekiga.net into the sip.conf file under [ekiga]. I also tried externip=71.xxx.xxx.xxx as shown below. Neither works. My junction client set up is working behind the same firewall. Can anybody suggest a fix? Thanks, Bud Roth P.S. Debug output and sip.conf file pasted below: Sip set d...

...ddress change. This a lot less trivial, maybe unnecessary, and probably covered by the previous item. I would seriously consider this patch for addition to 1.4 and 1.2. The code is very little intrusive, and it would solve in a correct way the nat traversal problems for which externip/externaddr/stunaddr are only a partial and expensive workaround. __________________________________________________________________

Having some trouble with sip behind a nat. So tried: stunaddr = numb.viagenie.ca in sip.conf. Didn't help so tried stun debug: asterisk*CLI> stun set debug on STUN Debugging Enabled STUN Packet, msg Binding Response (0101), length: 36 Found STUN Attribute Mapped Address (0001), length 8 Ignoring STUN attribute Mapped Address (0001), length 8 Found S...

is there general documentation on how asterisk behaves as a stun client (besides res_stun_monitor.conf) ? e.g.,: * can asterisk use multiple stun servers ? (im interested in availability, not data parity) * what is the relationship between gtalk.conf's stunaddr and res_stun_monitor.conf ? will duplicate queries be sent ? * Does asterisk provide some call (through AMI, console, etc.) that shows the status of the stun interoperability? like 'stun show status'? -- Jeremy Kister http://jeremy.kister.net./

...00000000", "gtalk/andy-gtalk/+1xxxyyyzzzz at voice.google.com") in new stack [Apr 1 10:41:53] ERROR[2416]: chan_gtalk.c:1934 gtalk_request: No XMPP client to talk to, us (partial JID) : andy-gtalk gtalk.conf [general] context=google-in ; Context to dump call into allowguest=yes stunaddr = numb.viagenie.ca bindaddr=0.0.0.0 externip=aa.bb.cc.dd disallow=all allow=ulaw [andy-gtalk] username=<username>@gmail.com context=google-in connection=andy-jabber gtalk show settings Global Settings: ---------------- UDP Bindaddress: 0.0.0.0 Stun Address: 66...

Dear all, I have enabled stun module and configured it in asterisk , but asterisk not using stun returned public ip address for any of the sip requests going out of my network. i have done settings as below res_stun_monitor.conf settings: [general] stunaddr = stun.ideasip.com stunrefresh = 30 stun show status Hostname Port Period Retries Status ExternAddr externport stun.ideasip.com 3478 30 3 OK 61.12.17.171 39710 sip.conf localnet=192.168.0.0/255.255.255.0 register=>jai9999:123456:jai9999 at sip2...

...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 1u7hp86jdg67.invalid;transport=ws   RTP Engine   : asterisk   En...

...ins fixes since the last release candidate as reported by the community. A sampling of the changes in this release candidate include: * Additional fixups in chan_gtalk that allow outbound calls to both Google Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip and stunaddr. (Closes issue #13971. Patched by dvossel) * Resolve manager crash issue. (Closes issue #17994. Reported by vrban. Patchd by dvossel) * Documentation updates for sample configuration files. (Closes issues #18107, #18101. Reported, patched by lathama, lmadsen) * Resolve issue wh...

...ins fixes since the last release candidate as reported by the community. A sampling of the changes in this release candidate include: * Additional fixups in chan_gtalk that allow outbound calls to both Google Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip and stunaddr. (Closes issue #13971. Patched by dvossel) * Resolve manager crash issue. (Closes issue #17994. Reported by vrban. Patchd by dvossel) * Documentation updates for sample configuration files. (Closes issues #18107, #18101. Reported, patched by lathama, lmadsen) * Resolve issue wh...

...disallow=all allow=ulaw dtlsenable=yes dtlsverify=fingerprint dtlscertfile=/etc/asterisk/keys/asterisk.pem dtlscafile=/etc/asterisk/keys/ca.crt dtlssetup=actpass *extensions.conf:* [default] exten => _6XXX,1,Dial(SIP/${EXTEN}) *rtp.conf:* [general] rtpstart=10000 rtpend=20000 icesupport=yes stunaddr=stun.l.google.com:19302 2015-08-10 12:35 GMT-03:00 Joshua Colp <jcolp at digium.com>: > Marek Cervenka wrote: > >> hello, >> >> i'm facing strange problem >> >> asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 >> person1 to person3 are b...

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

...=yes [whitehat238] type=client serverhost=talk.google.com username=whitehat238 at gmail.com/Talk secret=password port=5222 usetls=yes usesasl=yes status=Available statusmessage="No Information Available" timeout=100 keepalive=yes ##gtalk.conf## [general] allowguest=yes context=googlein stunaddr=stun01.sipphone.com [guest] disallow=all allow=ulaw connection=whitehat238 context=googlein ##extensions_custom.conf## exten => whitehat238 at gmail.com ,1,Set(CALLERID(name)=${CUT(CALLERID(name),@,1)}) exten => whitehat238 at gmail.com,n,GotoIf($["${CALLERID(name):0:2}" != &quot...

...rfc2833 context=from-voipms srvlookup=yes register => myuserid:mypass at dallas.voip.ms:5060~600 session-timers=refuse session-expires=3600 session-minse=600 session-refresher=uas localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 localnet=169.254.0.0/255.255.0.0 stunaddr=stun01.sipphone.com allow=ulaw allow=gsm [voipms] canreinvite=yes context=from-voipms host=dallas.voip.ms secret=mypass type=friend username=myuserid disallow=all allow=gsm fromuser=myuserid trustrpid=yes sendrpid=yes insecure=port,invite I have complete logs and SIP debugging from the time that...

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,

Hi folks. Please don't flame me but I've been googling around for days, read a tremendous amount, tried everything, and still no go. This is most definitely a typical newbie question. - I sure hope there's somebody(s) out there who'll humble themselves to help me out. I've set up an 'out of the box' basic Asterisk server running on Slackware Linux. - It basically

....net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr = stun.exiga.net insecure=port,invite ; required for incoming ekiga.net calls /etc/asterisk/extensions.conf: [from-internal] ... exten => _8.,1,Dial(SIP/ekiga/${EXTEN:1},20,r)) I tried a echo test, dialing in my case to 8500, but in spite of seeing traffic towards Internet, nothing is heard...

...oot at myserver admin]# yum install uuid-devel libuuid-devel [root at myserver admin]# ./configure --libdir=/usr/lib64 [root at myserver admin]# make menuselect [root at myserver admin]# make && make install In my sip.conf I have : icesupport = yes In my rtp.conf I have : icesupport=yes stunaddr=stun.l.google.com:19302 My SIP peer definition for webRTC client (sipml5) : [770000wrtc] type=peer host=dynamic username=770000wrtc defaultuser=770000wrtc fromuser=770000wrtc secret=987654 disallow=all allow=alaw ;allow=gsm qualify=yes canreinvite=no dtmfmode=rfc2833 amaflags=billing context=test...