search for: externrefresh

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 _________________________________________________________________ Windows Live?: Keep y...

...d 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; Another RFC1918 with CIDR notation       localnet = 169.254.0.0 / 255.255.0.0;...

....org. All works fine, until the host foo.dyndns.org for some reason change his ip, asterisk didn't resolve again the new ip until a "sip relolad" Actually, i use a cron with a bash script to track the ip and eventually reload the sip.conf. Any tips for Asterisk ? Something like externrefresh for a peer? Thanks, Alessandro

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

...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 >        localnet = 10.0.0.0 / 255.0.0.0; Also RFC1918 >        localnet = 172.16.0.0 / 12; Another RFC1918 with CIDR notation >...

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 with the only exception of both parties bei...

Hi All; I am using IAX Trunk and I used ddns (dyndns.org) with the host (host=xyz.dyndns.org), when I restart the router who has the hostname xyz.dyndns.org then its IP address change and updated, but at asterisk level still it keeps sending for the old IP address and sometimes this problem does not resolve until I restart asterisk. Any one faced this and has idea how to resolve it so Asterisk

...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 insecure=port,invite And t...

...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 dial=SIP/00491771111111 and fin...

...te 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 12.194.12.12:5060: INVITE sip:173242...

.../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] logger.c: -- Executing [s at...

...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; Another RFC1918 with CIDR notation > localnet...

I have an asterisk server with a dynamic public IP address. Once the IP changes, remote clients suddenly have one-way audio again. I can resolve the problem with a restart, but am thinking have adding a cron command which does this every night. Will a "reload" cause asterisk to respect the new IP address specified in sip.conf? Or do I have to restart? Thanks, MD --------------

...actually correct? sip_nat.conf # this is when i got the NAT -route, right? #gets all the dyndns-stuff #externip = home.mydomain.com (Enter your DynamicDNS domain name. Obviously it's just easier to get a static IP address and avoid using DynamicDNS altogether.) externhost = home.mydomain.com externrefresh = 5 (which means lookup hostname every 5 minutes to refresh ip adress) localnet = internal.network.address.0/255.255.255.0 thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090709/80ccd32a/attachment....

...0.0.0 allowexternaldomains = no allowoverlap = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = 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 dtm...

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 asterisk is restarted or sip reload. Or just everyone uses fix...

....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; Another RFC1918 with CIDR notation &...

...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 dial=SIP/00491771111111 and fin...

Hi to everyone! I have succerfully instaled my new Asterisk 1.4 on my debian etch. I have my users in sip.conf like this: [200] type=peer host=dynamic context=home secret=200 callerid= 200 dtmfmode=rfc2833 nat=yes mailbox=200 at home disallow=all allow=ulaw I can make calls in my LAN but i can?t ear comunications with another client trought Internet. My Asterisk is in my LAN and i not have a

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