search for: tcpbindaddr

...an asterisk 16.2.1 on an ubuntu on AWS, which is experiencing a drop in call. It does not have a certain time, it is random. The audio 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....

I am playing with tcpenable... on 13.11.2 so in sip.conf I have tcpenable=yes tcpbindaddr=192.168.1.8:5070 but when I "telnet localhost 5070" I get no connect. iptables -L -n -v | grep 5070 0 0 ACCEPT tcp -- * * 0.0.0.0/0 0.0.0.0/0 state NEW tcp dpt:5070 firewall is good. Is my syntax not correct above to run on port 5070 for SIP over TC...

...It does not have a certain time, it is random. The > audio > 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 > <http://foo.dyndns.net>; refreshed p...

Hello List. I am puzzled and how asterisk listens to calls or connections from clients. When I do a netstat -nat I don't see asterisk listening on port 5060. Now, I'm testing a server with three network interfaces: two to the internet doing load balancing and the other to our LAN. I would like asterisk to only accept connections coming from our LAN but, can't find where to configure

...b8::1/ /; c) Listen on the IPv4 wildcard. Example: bindaddr=0.0.0.0/ /; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=::/ /; (You can choose independently for UDP, TCP, and TLS, by specifying different values for/ /; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".)/ /; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat./ /; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)/ /;/ /; You may optionally add a port number. (The default is port 5060 for UD...

...https://www.mail-archive.com/sr-users at lists.sip-router.org/msg18558.html What I don't know is how to configure sip.conf, so far I've just been making guesses based on online examples and documentation. My current sip.conf looks like this: [general] bindport = 5070 bindaddr = 127.0.0.1 tcpbindaddr = 127.0.0.1:5070 tcpenable = no limitonpeers = yes ;rtcachefriends = yes tos_sip=cs3 tos_audio=ef realm = testers.com I've tried defining realm and domain values, but I lack proper understanding of those. Can you guys help me out? Are there any other configurations I need to check? Respectful...

...x,n,Set(CALLERID(num)=1234567890) exten => xxx,n,Set(CALLERID(name)=) exten => xxx,n,Noop(CallerID is ${CALLERID(all)}) exten => xxx,n(dialout),Dial(SIP/POTS1,60,o) exten => xxx,n,Hangup And my general and section from sip.conf [general] allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=alaw limitonpeers=yes notifyringing=yes maxexpirery=180 defaultexpirey=180 [POTS1] type=friend secret=xxx context=pots_in host=dynamic dtmfmode=info disallow=all allow=ulaw allow=alaw canreinvite=no qualify=yes call-limit=4 rtptimeout=30 And her...

...r nothing at the peer's end. When one peer calls another, sound comes through just fine. So my hunch is that is something to do with the audio supplied by the server. Do I need to have alsa installed?? Any hint? sip.conf: [general] context = unauthenticated bindport = 5060 bindaddr = 0.0.0.0 tcpbindaddr = 0.0.0.0 tcpenable = yes videosupport = no textsupport=yes alwaysauthreject=yes allowguest=no [1001] ; grandstream 1 context = home type = friend callerid = One <1001> secret = XYZ host = dynamic mailbox = 1001 disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-t...

...m that it is an error, that asterisk sends the RTP stream via udp and this is the cause of the silence? Is there any way to tell asterisk to use tcp only? I'm aware of the drawbacks, but i still need to get this working. I'd appreciate any help. thanks adam sip.conf: tcpenable=yes tcpbindaddr=0.0.0.0 [ocs] type=friend host=192.168.1.1 context=ocs qualify=yes transport=tcp nat=no canreinvite=no disallow=all allow=alaw allow=ulaw [deverto4] type=friend host=172.18.200.4 context=deverto qualify=yes nat=no canreinvite=yes transport=tcp disallow=all allow=alaw allow=ulaw and the extension...

...t; 555,1,Dial(IAX2/111) exten => 555,n,Hangup() [special] exten => 111,1,Dial(IAX2/111) exten => 111,n,Hangup() [default] exten => 444,1,Dial(IAX2/444) exten => 444,n,Hangup() - Sip.conf (SIP server): [general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes --------- - Logs server: --------- -- Accepting AUTHENTICATED call from 10.0.100.238: > requested format = gsm, > requested prefs = (), > actual format = ulaw, > host prefs = (), > priority = mine -- Executing [111 a...

...qualify=yes realm=mcts.org promiscredir=yes ;Some have suggested using canreinvite=no with Avaya- didn't try that yet ;canreinvite=no canreinvite=yes transport=tcp ;context=incoming context=from-internal ;username=10.90.0.103 fromdomain=mcts.org disallow=all allow=ulaw allow=alaw tcpenable=yes tcpbindaddr=0.0.0.0:5060 Nothing I tried seems to make it stop sending asterisk@(null) in the header. This is supposed to be a sip trunk, not an extension, so I think I should NOT be user a username or secret. I'm not even sure what promiscredir does, or if it's helping or harming me. There's vir...

...is experiencing a > drop in call. It does not have a certain time, it is random. The audio > 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 > > localne...

...provider 'sip_proxy'. Calls from this provider ; connect to local extension 1234 in extensions.conf, default context, ; unless you configure a [sip_proxy] section below, and configure a ; context. sip.conf: [general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes register => tjoen:mypasswd at sip_proxy/1234 qualify=yes externip=myipnr localnet=192.168.254.0/255.255.255.0 nat=yes [sip_proxy] type=peer host=ekiga.net extensions.conf: [default] include => demo exten => 1234,1,Dial(SIP/2133) Output of asterisk -vvv ... REGISTER...

...K address localmask = 255.255.255.0 ; Internal netmask ; The externip, localnet and localmask is used ; when registering and communication with other proxies ; that we're registered with tcpbindaddr=0.0.0.0 bindaddr = 0.0.0.0 Leif

...g to follow "The Asterisk Book" at http://www.the-asterisk-book.com/unstable/minimale-telefonanlage.html and created a VERY basic sip.conf; see http://min.us/my-asterisk#2 or http://min.us/my-asterisk#1 for the complete /etc/asterisk directory. [general] udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 srvlookup=yes realm=tel.skwar.me [2000] type=friend secret=1234 host=dynamic [2002] type=friend secret=0220 host=dynamic ; EOF Now I'm trying to connect to my Asterisk using the Siphon SIP softphone on iPhone; I've set username to 2000, password to 1234 and hostname to tel.skwar....

...e=/var/lib/asterisk/keys/asterisk.pem tlscafile=/var/lib/asterisk/keys/ca.crt tlscipher=ALL srtpcapable=yes ;tlsclientmethod=tlsv1 tlsdontverifyserver=yes and the phone is sourced by context=default ; Default context for incoming calls allowoverlap=no udpbindaddr=:: tcpenable=yes tcpbindaddr=:: srvlookup=yes and [IPV6](!,my-codecs) dtmfmode=rfc2833 context=sip-out type=friend host=dynamic transport=tls encryption=yes nat=no qualify=yes the phone it's self contains [200](IPV6) context=abc callerid=123 defaultuser=123 fromuser=123 secret=secret mailbox=123 at default The rtp p...

...ib/asterisk/keys/asterisk.pem tlscafile=/var/lib/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 tlsdontverifyserver=yes ;--------------------------Default---------------- context=default ; Default context for incoming calls allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 srvlookup=yes [my-codecs](!) ; a template for my preferred codecs disallow=all allow=g722 allow=alaw allow=ulaw allow=speex allow=g723 allow=ilbc allow=g729 allow=gsm [NAT](!,my-codecs) dtmfmode=rfc2833 context=sip-out type=friend host=dynamic transport=tls,tcp qualify=...

...It does not have a certain time, it is random. The audio >> 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 &...

...m the CLI to my SIP extension. What does this mean? Any help with troubleshooting this appreciated. Some info: /etc/asterisk/sip.conf context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes [authentication] [basic-options](!) dtmfmode=rfc2833 context=from-office type=friend [natted-phone](!,basic-options) nat=yes canreinvite=no host=dynamic [public-phone](!,basic-op...

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes limitonpeers=yes callcounter=yes vmexten=5199 nat=no ; SE registrations register => user1:password1 at sipgate.co.uk:5060/se2489 register => user2:password2 at sipgate.co.uk:5060...