search for: externhost

externhost is monitoring for ip changes with an interval of externrefresh, so far so good. Wouldnt it be handy if asterisk would do an sip reregister if it detects an ip change? My SIP provider has sometimes very high intervals of 1 hour and if ip changes, the registration doesnt work until it expires or...

how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring. Edit sip_nat.conf for proper NAT: localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here) externrefresh=10 fromdomain=DOMAIN.com (Set your external domain name here) nat=yes qualify=yes canreinvite=no Add extra codecs to /etc/asterisk/sip_custom.conf allow=gsm allow=h261 allow=h263 allow=h263p videosupport=yes ___________________...

My Asterisk server is behind a NAT and I have set: ---------------------------------------------------------------------------- externhost="my.server.address" externrefresh=180 localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 nat=yes --------------------------------------------------------------------------- in [general] section of sip.conf. I can make perfect conversation with my friend...

...ll be the client (with dynamic ip) and the 2nd the server. The client uses sip and the server pjsip. This is the client's sip.conf [general] context = default allowguest = no realm = myrealm.com udpbindaddr = 0.0.0.0 qualify = yes subscribecontext = default localnet=192.168.1.0/255.255.255.0 externhost=myhost.com externrefresh=30 dtmfmode = auto canreinvite = no jbenable = no sendrpid = yes trustrpid = no disallow=all allow=ulaw allow=alaw register => myuser:mypass at vpsserver [vpsserver] type=friend secret=myuser defaultuser=mypass host=vpsserver.domain.com context=inbound canreinvite=no in...

...he LAN behind the router, in the same network asterisk is running at, takes the call. but we can't hear / talk with each other. Ay ideas to get this thing solved?! My general section in sip.conf: [general] port=5060 bindaddr=0.0.0.0 localnet=192.168.1.0/255.255.255.0 externip=85.183.112.3 externhost=voipfax.higazi-it.com allowtransfer=yes qualify=yes nat=yes [2006] type=friend secret=frank host=dynamic context=nurintern nat=no [2007] type=friend secret=jochen host=192.168.1.2 context=nurintern nat=yes

...o is flowing normally and the call is dropped. Has anyone ever experienced this? My settings changed below: allowoverlap = no udpbindaddr = 0.0.0.0 tcpenable = no tcpbindaddr = 0.0.0.0 transport = udp, ws, wss srvlookup = yes directmedia = no rtcachefriends = yes externaddr = my ip address externhost = my domain address ;   foo.dyndns.net; refreshed periodically externrefresh = 180       localnet = 172.31.40.21 / 255.255.240.0; AWS NETWORK       localnet = 192.168.0.0 / 255.255.0.0; RFC 1918 addresses       localnet = 10.0.0.0 / 255.0.0.0; Also RFC1918       localnet = 172.16.0.0 / 12; Ano...

...Content-Type: application/sdp^M Content-Length: 315^M ^M v=0^M o=root 1021147583 1021147583 IN IP4 Z.Z.247.106^M s=Asterisk PBX 1.6.0.17^M c=IN IP4 Z.Z.247.106^M t=0 0^M m=audio 18702 RTP/AVP 0 8 3 101^M I have the following in the sip_nat.conf localnet=Y.Y.47.149/255.255.0.0 externhost=Z.Z.247.106 externrefresh=10 fromdomain=att.com nat=yes qualify=yes canreinvite=no I think the SDP should have give the Y.Y.47.149 IP on the local net side to the phone. But I am unable to figure how make it do that. The Asterisk log shows this. [Feb 25 11:06:30] VERBOSE[1449] logge...

...one way audio problem, as far as i know it's a nating problem, my hosts inside my lan can call to outside internet hosts, but can't listen a thing, i read a lot about sip and rtp and protocols and the problem it seems to be with NAT, this is the config i put on my sip.conf file about nat: externhost=sip.server.com.ar > my server name on the internet localnet=192.168.5.0/255.255.0.0 > my LAN nat=yes canreinvite=yes And this are the ports i opened on my firewall script iptables -A INPUT -p udp -m udp --dport 8766:35000 -j ACCEPT iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCE...

I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the localnet, and a trunk to Sipphone/Gizmo/Google Voice. The externhost and localnet parameters are all set correctly in sip.conf. An inbound call from Sipphone works great until the local channel places the call on hold. During hold, the Sipphone user cannot hear music, only silence. The silence continues after the hold, though the local phone can hear the Sipphone...

Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 10000-20000 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 10000-20000. The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra

...; allowoverlap = no > udpbindaddr = 0.0.0.0 > tcpenable = no > tcpbindaddr = 0.0.0.0 > > transport = udp, ws, wss > > srvlookup = yes > > directmedia = no > > rtcachefriends = yes > > externaddr = my ip address > > externhost = my domain address ; foo.dyndns.net > <http://foo.dyndns.net>; refreshed periodically > externrefresh = 180 > >        localnet = 172.31.40.21 / 255.255.240.0; AWS NETWORK >        localnet = 192.168.0.0 / 255.255.0.0; RFC 1918 addresses >        localne...

...n my internal network in a NAT, connected to my sip-provider using a trunk. Firewall settings on the router are: forward UDP port 5060,5004,10000-20000 to asterisk server Firewall on asterisk server is switched off (for better testing) [sip.conf] [general] port=5060 bindaddr=0.0.0.0 language = de externhost=intakt-musik.dyndns.org externip=intakt-musik.dyndns.org localnet=192.168.0.0/255.255.255.0 nat=yes register => USER:PASS at sipconnect.sipgate.de/USER [sipconnect.sipgate.de] type = friend host = sipconnect.sipgate.de outboundproxy = sipconnect.live.sipgate.de port = 5060 username = USER fromu...

...; accepts the registration ; Default is 0 tries, continue forever ;callevents=no ; generate manager events when sip ua performs events (e.g. hold) ;----------------------------------------- NAT SUPPORT ------------------------ ; The externip, externhost and localnet settings are used if you use Asterisk ; behind a NAT device to communicate with services on the outside. ;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages ; if we're behind a NAT...

...e new ip address, the peers are connected with the old address. Waiting doesn?t help, only a ?sip reload? update the ip address of the peer. What is the solution for this problem? How can asterisk update the peer? The Asterisk is local behind a NAT with a firewall, following settings are used: externhost with DynDNS stun with stun.t-online.de <http://stun.t-online.de/> nat=yes srvlookup=yes allowguest=no trustrpid=no insecure=invite qualify=yes Thank you! Daniel -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-user...

Good Day All I recently implemnetd asterisk outside our LAN (external network).It works well in a NAT settings. But on external network with PROXY setting ASTERISK DID NOT WORK. My question are 1 Can ASTERISK work in a PROXY setting . 2 If it can work how can i implement it . Expecting your reply Thanks Paul --------------------------------- Yahoo! Messenger

...t it is a problem with the ports since the client registers itself at some time. Whatever happens, I'm allowing connections for the remote IP to the 5060 tcp/UDP port and 10000:20000 UDP in the firewall. The router that it is ahead has these ports redirected to the firewall. Also I'm using externhost, externip and localnet in /etc/asterisk/sip.conf Which can be the problem? Thanks in advance for your reply. Regards, Daniel -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvBAFQACgkQZpa/GxTmHTe0mgCcCmDNhkMm3DMc/Ckd7AAzZneF 4ngAn0SL/IC58kNDktcRsxJOaKPoAuCL =Ve4J...

...I can register my Asterisk on an external provider, I'd like to add a new feature and allow my mobile phone to connect to my Asterisk and manage calls. Well, first of all, my Asterisk is NOT direct on Internet available, but behind a NAT. So I configured my sip.conf: localnet=192.168.200.0/24 externhost=myhost.noip.com externrefresh=180 Then I added the peer in my users.con: [00491771111111] fullname = 00491771111111 secret = MYVERYSECRET type=peer nat=yes qualify=yes qualifyfreq=60 hassip = yes dahdichan = 1 transport=udp,tcp callwaiting = no context = default host = dynamic dtmfmode=rfc2833 di...

...i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13 externrefresh=10 fromdomain=test.localhost.com nat=yes qualify=yes canreinvite=no NAT on device end i.e. my softphone (extension) has already set to yes with canreinvite=no but still unable to resolve this issue. SIP traces are listed below; Reliably Transmitting (NAT) to 12.194.12...

...srvlookup=yes domain=proxy.myhostname disallow=all allow=alaw sipdebug = yes recordhistory=yes dumphistory=yes register => <authstuff>@sip.externalpeer.com externhost=proxy.myhostname 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 nat=never canreinvite=no [authentication] auth = <authstuff>@sip.externalpeer.com [provider] type=peer username=&lt...

...) > ;registerattempts=10 ; Number of registration attempts before we give up > callevents=no ; generate manager events when sip ua performs events (e.g. hold) > externip=The_IP_of_my_router ; Address that we're going to put in outbound SIP messages > ;externhost=foo.dyndns.net ; Alternatively you can specify an > ;externrefresh=10 ; How often to refresh externhost if > localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks > localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 > localnet=172.16.0.0/12...