Scheduling destruction of SIP dialog 'RN3QrX8Jgovfbr-hmrKh41I3nyFFZa2t' in 17984 ms (Method: INVITE) WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6024025a" Via: SIP/2.0/UDP 10.48.86.155:5060 branch=z9hG4bKPjbT.cxVVSfoiiUzWyTLpDVpUnKN5uzoWI received=my. rport=47943Īllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Using INVITE request as basis request - RN3QrX8Jgovfbr-hmrKh41I3nyFFZa2tįound peer '100' for '100' from my.:47943 Supported: replaces, 100rel, timer, norefersub Via: SIP/2.0/UDP 10.48.86.155:5060 rport branch=z9hG4bKPjbT.cxVVSfoiiUzWyTLpDVpUnKN5uzoWIįrom: tag=MRPQ-.7-6bNoWfiylMKQKuvdxdAQYepSĬall-ID: RN3QrX8Jgovfbr-hmrKh41I3nyFFZa2tĪllow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS No such command '#start' (type 'core show help #start' for other possible commands) My.=a public ip addressġ0.48.86.155~=~i dont know this ip? also not pingable! or telnet 5060 This is the sip debugging information: Where: = Spawn extension (macro-phone, NOANSWER, 1) exited non-zero on 'SIP/100-00000018' in macro 'phone' x=2, open writing: /opt/var/spool/asterisk/voicemail/default/300/tmp/nx78QP format: wav, 0x9e8dd4 x=1, open writing: /opt/var/spool/asterisk/voicemail/default/300/tmp/nx78QP format: gsm, 0x8587bc x=0, open writing: /opt/var/spool/asterisk/voicemail/default/300/tmp/nx78QP format: wav49, 0x9d782c Playing 'vm-isunavail.gsm' (language 'en') Playing 'vm-theperson.gsm' (language 'en') In this case I did not pickup the phone, so the voicemail has to pick it up. This is the verbose listing of a call from the NATted CSipSimple phone calling the Cisco301.
![csip simple android app csip simple android app](https://www.netelip.com/blog/wp-content/uploads/2012/11/csip-simple.jpg)
No such command '#HANG UP' (type 'core show help #HANG UP' for other possible commands) = Spawn extension (macro-phone, s, 1) exited non-zero on 'SIP/100-00000022' in macro 'phone' No such command '#PICKED UP' (type 'core show help #PICKED UP' for other possible commands) No such command '#RING' (type 'core show help #RING' for other possible commands) It is also not in one of the config files. Where is that comming from? I dont know this ip. One thing that attracts my attention is this ip address: 10.48.86.155. With less words: I can call the phone visaversa but I do not hear anything. With this line “With the Cisco301 (local) I can call the CSipSimple (outsite), and the Cisco301 also rings wenn I call it with the CSipSimple.”, I tried to explain what isn’t working. (!,basic-options) another template inheriting basic-options Nat = auto_force_rport Set the force_rport option if Asterisk detects NAT (default) Srvlookup=yes Enable DNS SRV lookups on outbound callsĮxternhost = my.:5060 Public address of my nat box.Įxternrefresh = 600 check hostip every 10min. The order determines the primary default transport. Transport=udp Set the default transports. Tcpbindaddr=0.0.0.0 IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) Tcpenable=no Enable server for incoming TCP connections (default is no) Udpbindaddr=0.0.0.0 IP address to bind UDP listen socket to (0.0.0.0 binds to all) Defaults to 'default'Īllowoverlap=no Disable overlap dialing support. # cat sip.conf|egrep -v "^ |^$|^\s* "Ĭontext=public Default context for incoming calls. Who can help me to find the configuration error? in sip.conf (see below, =CSipSimple, =Cisco301) I read a lot about NATting but I cannot seem to make it work. With the Cisco301 (local) I can call the CSipSimple (outsite), and the Cisco301 also rings wenn I call it with the CSipSimple.
![csip simple android app csip simple android app](https://miro.medium.com/max/480/1*JF1AN-nGfDuEgPoNNjIYAg.jpeg)
I have Asterisk 11 running on my qnap TS219.Īnd a CSipSimple installed on a Samsung S2.īut (as you can expect) when login with my CSipSimple from outsite, not all is working well I’ve been trying now for days no and I give up