search for: udpbindaddr

...Asterisk 13 /usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip I assume I run it from /etc/asterisk with the input and output file as arguments however there's no instructions and I don't Grok python. Unfortunately it's not working, Despite what the below error states I do have a udpbindaddr set to 0.0.0.0 in my configuration. root at kiniston01:/etc/asterisk# /usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py sip.conf pjsip.conf Traceback (most recent call last): File "/usr/src/asterisk-13.0.0/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py", line 1158, i...

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....

...to make calls (in today's case to Cuba). Here's a copy (slightly modified) of my sip.conf: [general] context=default ; Default context for incoming calls videosupport=yes rtcachefriends=yes autocreatepeer=no t38pt_udptl=yes allowoverlap=no udpbindaddr=0.0.0.0 srvlookup=yes ;pedantic=yes disallow=all allow=alaw allow=ulaw allow=speex [1001] type=friend username=1001 secret=blah subscribecontext=default regexten=1001 callerid="blah" <XXXXXXXXXX> host=dynamic nat=yes canreinvite=no mailbox=1001 at default registertrying=yes [tes...

...,AccountID="sip:venu at 192.168.0.35",Sess ionID="0x337bf68",LocalAddress="IPV4/UDP/10.10.1.3/5060",RemoteAddress="IPV4 /UDP/192.168.1.90/5060",Challenge="41cdcd16" ^^^ The other networks confuse me, and perhaps asterisk. Perhaps serverA:sip.conf udpbindaddr=192.168.0.35 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) serverB:sip.conf udpbindaddr=192.168.0.36 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)...

...hich 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....

...uld add SIP users in here. However, I currently don?t understand whether this realtime dynamic configuration table is meant to replace or just supplement sip.conf. This is because the sippeers table does not offer certain fields for entries in the [general] section of my sip.conf file, such as the ?udpbindaddr? variable. So, am I supposed to put all that in the database by adding appropriate table columns, or can I leave this in the sip.conf file and chan_sip.so will read both the file and MySQL table once loaded? Also, is there anyway that I could use templates, so that I don?t have to redefine everythi...

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

...mple: bindaddr=2001:db8::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 defaul...

...tlsprivatekey /etc/asterisk/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 -------...

...p on one and pjsip on the other one. One is running at home and another at a VPS. The first one will 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=fr...

Hi Friends, I encountered a strange issue. I am running Asterisk 11.8.1 on Cent OS with no firewall running. It has 3 NIC interfaces. in my sip.conf I have allowguest=yes bindaddr=0.0.0.0 udpbindaddr = 0.0.0.0 But my Asterisk instance does not pick the call at all. When I check the listening apps using lsof -i I get asterisk 3046 asterisk 7u IPv4 1191172 0t0 TCP *:5038 (LISTEN) asterisk 3046 asterisk 10u IPv4 1191186 0t0 UDP *:sip asterisk 3046 asterisk 11u IPv...

...SIP_HEADER(From)}) exten => 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 ca...

...- [general] [externe] 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 = (), > priori...

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

...1 AM, Raj Roy Ghandhi <roy.gandhi at gmail.com> wrote: > Hi Friends, > I encountered a strange issue. > I am running Asterisk 11.8.1 on Cent OS with no firewall running. > It has 3 NIC interfaces. > > in my sip.conf I have > > allowguest=yes > bindaddr=0.0.0.0 > udpbindaddr = 0.0.0.0 > > But my Asterisk instance does not pick the call at all. > > When I check the listening apps using lsof -i I get > > asterisk 3046 asterisk 7u IPv4 1191172 0t0 TCP *:5038 (LISTEN) > asterisk 3046 asterisk 10u IPv4 1191186 0t0 UDP *:sip >...

...2102 with two phone numbers installed. I have NAT Mapping and NAT Keepalive enabled. No STUN server. Both are using 5060. This is behind an ADSL through a WRT54GL with no special port handling. The server is 11.15.1. My sip.conf includes this: [general] context=unauthenticated allowguest=yes udpbindaddr=0.0.0.0 nat=force_rport,comedia srvlookup=yes qualify=yes All of this works fine. It also works fine with the few clients that I have connected. However, certain changes cause failures, usually one way audio suggestion NAT issues. First experience - I add a softphone on my laptop and assign a t...

...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 &g...

...y/1234 ; ; Register 2345 at sip 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) Out...

...9;m new to Asterisk, I'm trying 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...

...yes tlsbindaddr=:: tlscertfile=/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 mailb...