search for: tcpenable

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...

...can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [10000] type=friend secret=test host=dynamic nat=yes -------------------------- ----- extensions.conf: ----- [common] exten => 501,1,MeetMe(12,MI) exten => 501,n,Hangup() exten => i,1,Hangup() exten => h,1,Hangup() exten => t,1,Hangup() ----------------...

Hello I have 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     ...

...ck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten this set-up (Asterisk11 with Snom870s using TLS) to work and if so could you provide the details? I have this in Asterisk sip.conf (loaded through FreePBXs sip_general_additional.conf). tcpenable=yes tlsenable=yes tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt tlscafile=/etc/pki/tls/certs/ca-bundle.crt tlsdontverifyserver=yes tlscipher=ALL tlsclientmethod=tlsv1 And I have this for the test device context: [41712] deny=0.0.0.0/0.0.0.0 secret=NearlyANastyThat dtmfmode=rfc...

...> 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 > <http://foo.d...

Hi all, I am trying to get iptables to work for me... I am running asterisk (11.23.0) on a C5 machine. Working fine on port 5060 udp. I have need to tcpenable=yes SIP and run that on port 5068. Since port 5060 is already running I was going to redirect 5068 to 5060. So I thought I could use iptables to do that - but does not seem to be working. 192.168.10.201 is my machine, 192.168.1.3 is the other machine. 1.3 should connect to 10.201 on port 5068. so...

...l, * The fax from Asterisk cmd SendFax to PSTN fax works well, * However, the fax from SPA112 to PSTN fax doesn't work. Using udptl debug, I can see packets between Asterisk and both sides (SPA112 and PSTN fax) but it seems that faxes can't agree how to send image. == sip.conf: [general] tcpenable=yes videosupport=yes transport=udp,tcp dtmfmode=rfc2833 qualify=yes directmedia=no allowguest=no alwaysauthreject=yes rtcachefriends=yes rtupdate=no callcounter=yes t38pt_udptl=yes,redundancy,maxdatagram=200 t38pt_rtp=no t38pt_tcp=no ignoresdpversion=yes disallow=all allow=alaw allow=ulaw externip=...

On Tue, March 3, 2015 13:37, James Cloos wrote: >>>>>> "JBB" == James B Byrne <byrnejb at harte-lyne.ca> writes: > > JBB> tcpenable=yes > JBB> tlsenable=yes > JBB> tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt > JBB> tlscafile=/etc/pki/tls/certs/ca-bundle.crt > JBB> tlsdontverifyserver=yes > JBB> tlscipher=ALL > JBB> tlsclientmethod=tlsv1 > > You are missing the tls...

...sk/keys/asterisk.pem dtlssetup actpass sippasswd md5pwd rpid domain testers.com sippasswd2 and my sip.conf: [general] bindport = 5070 bindaddr = PU.BL.IC.IP udpbindaddr = PU.BL.IC.IP tcpenable = yes limitonpeers = yes rtcachefriends = no tos_sip=cs3 tos_audio=ef realm = testers.com autodomain=yes domain=PU.BL.IC.IP domain=testers.com transport=ws,wss,udp outboundproxy=PU.BL.IC.IP:5060 I'd appreciate Your advice. cheers, Olli -------------- next part -------------- An HTML attachme...

.../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? Respectfully, Olli -------------- nex...

...xten => xxx,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 rtptime...

...#39;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-tones from the device...

...please confirm 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 a...

...ne] exten => 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 -- E...

...es exceeded to host", I can't use our IAX outgoing trunks and the only way to get things working again is to restart Asterisk. Am I missing something silly here? Here is my sip.conf: [general] subscribecontext=sip-blf context=default disallow=all allow=alaw allow=ulaw allowguest=no tcpenable=no tlsenable=no srvlookup=no localnet=192.168.56.0/255.255.255.0 localnet=192.168.57.0/255.255.255.0 tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. canreinvite=no dtmfmode = rfc2833 notifyringing=yes limitonpeers=yes ca...

...eed to enable or change in the configs or change my peer configurations? I'm not sure if this is relevant but I checked that Asterisk was successfully compiled with res_srtp module. Here's my sip.conf contents: bindport = 5070 ; using this since Kamailio is at 5060 bindaddr = PU.BL.IC.IP tcpenable = yes ;no limitonpeers = yes rtcachefriends = yes ; for realtime rtupdate=yes tos_sip=cs3 tos_audio=ef useragent=MyAsterisk realm = myrealm.com autodomain=no domain=PU.BL.IC.IP domain=testers.com allowexternaldomains=no allowguest=no avpf=yes encryption=yes transport=ws,udp icesupport=yes srv...

...allowguest=yes 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....

I do not want set allowguest=yes. The problem is, there is no official list with ip addresses of Telekom Germany. But I think all ip addresses comes from the ip range 217.0.0.0/13. I have now the following addition to sip.conf. I think it is the only safe option. Or what would you say? [telekom](!) context=from-trunk type=peer defaultuser= authuser= remotesecret= fromdomain=tel.t-online.de

>>>>> "JBB" == James B Byrne <byrnejb at harte-lyne.ca> writes: JBB> tcpenable=yes JBB> tlsenable=yes JBB> tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt JBB> tlscafile=/etc/pki/tls/certs/ca-bundle.crt JBB> tlsdontverifyserver=yes JBB> tlscipher=ALL JBB> tlsclientmethod=tlsv1 You are missing the tls key. The config name is tlsprivatekey;...

...buntu 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 &g...