Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context
Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to the asterisk box. This seems to be quite useful software but it's frustratingly difficult to get running. Jon SIP debug shows following mrpenguin*CLI> -- Registered '2203' (AUTHENTICATED) at 192.168.0.2:5036 -- Accepting AUTHENTICATED call from 192.168.0.2, requested format = 2, actu al format = 2 -- Executing SetCallerID("[EMAIL PROTECTED]/9", "91184") in new stack -- Executing SetCIDName("[EMAIL PROTECTED]/9", "calisto") in new stack -- Executing Dial("[EMAIL PROTECTED]/9", "SIP/[EMAIL PROTECTED]") in new stack We're at 82.38.193.149 port 16612 Answering with preferred capability 4 Answering with preferred capability 2 Answering with non-codec capability 1 11 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 From: "calisto" <sip:[EMAIL PROTECTED]>;tag=as1f0e4544 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 214 v=0 o=root 15141 15141 IN IP4 82.38.193.149 s=session c=IN IP4 82.38.193.149 t=0 0 m=audio 16612 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (NAT) to 192.246.69.223:5060 -- Called [EMAIL PROTECTED] Sip read: CLI> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 From: "calisto" <sip:[EMAIL PROTECTED]>;tag=as1f0e4544 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: Free World Dialup (0.8.11rc3 (i386/linux)) Content-Length: 0 8 headers, 0 lines Sip read: CLI> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 From: "calisto" <sip:[EMAIL PROTECTED]>;tag=as1f0e4544 To: <sip:[EMAIL PROTECTED]>;tag=as63b4567c Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Contact: <sip:[EMAIL PROTECTED]:5028> Content-Length: 0 9 headers, 0 lines -- SIP/fwd.pulver.com-43fd is ringing Sip read: CLI> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 From: "calisto" <sip:[EMAIL PROTECTED]>;tag=as1f0e4544 To: <sip:[EMAIL PROTECTED]>;tag=as2046b5cb Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 10 headers, 0 lines -- SIP/fwd.pulver.com-43fd is ringing Sip read: CLI> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 Record-Route: <sip:[EMAIL PROTECTED];ftag=as1f0e4544;lr=on> From: "calisto" <sip:[EMAIL PROTECTED]>;tag=as1f0e4544 To: <sip:[EMAIL PROTECTED]>;tag=as2046b5cb Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 212 v=0 o=root 11472 11472 IN IP4 65.121.72.14 s=session c=IN IP4 65.121.72.14 t=0 0 m=audio 12268 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 12 headers, 10 lines Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format GSM Found description format PCMU Found description format telephone-event Capabilities: us - 6, them - 6/0, combined - 6 Non-codec capabilities: us - 1, them - 1, combined - 1 list_route: hop: <sip:[EMAIL PROTECTED];ftag=as1f0e4544;lr=on> list_route: hop: <sip:[EMAIL PROTECTED]> set_destination: Parsing <sip:[EMAIL PROTECTED];ftag=as1f0e4544;lr=on> for addr ess/port to send to set_destination: set destination to 192.246.69.223, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 Route: <sip:[EMAIL PROTECTED]> From: "calisto" <sip:[EMAIL PROTECTED]>;tag=as1f0e4544 To: <sip:[EMAIL PROTECTED]>;tag=as2046b5cb Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 192.246.69.223:5060 -- SIP/fwd.pulver.com-43fd answered [EMAIL PROTECTED]/9 set_destination: Parsing <sip:[EMAIL PROTECTED];ftag=as1f0e4544;lr=on> for addr ess/port to send to set_destination: set destination to 192.246.69.223, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 Route: <sip:[EMAIL PROTECTED]> From: "calisto" <sip:[EMAIL PROTECTED]>;tag=as1f0e4544 To: <sip:[EMAIL PROTECTED]>;tag=as2046b5cb Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 192.246.69.223:5060 == Spawn extension (default, 7612, 3) exited non-zero on '[EMAIL PROTECTED]/9' -- Hungup '[EMAIL PROTECTED]/9' Sip read: CLI> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 Record-Route: <sip:[EMAIL PROTECTED];ftag=as1f0e4544;lr=on> From: "calisto" <sip:[EMAIL PROTECTED]>;tag=as1f0e4544 To: <sip:[EMAIL PROTECTED]>;tag=as2046b5cb Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 11 headers, 0 lines mrpenguin*CLI> SIP.CONF mrpenguin:/etc/asterisk# more sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = [REMOVED] ; Address that we're going to put in SIP message s if we're behind a NAT context = default ; Default for incoming calls disallow=all ; Disallow all codecs allow=ulaw ;allow=alaw allow=gsm register => 91184:[EMAIL PROTECTED]/2203 ;[2203] ;type=friend ;username=2203 ;host=dynamic ;defaultip=192.168.0.2 ;dtmfmode=inband ;canreinvite=no [fwd.pulver.com] type=friend secret=[REMOVED] username=91184 host=fwd.pulver.com nat=yes canreinvite=no reinvite=no IAX.conf (for the DIAX softphone) [general] port=5036 bandwidth=low allow=ulaw allow=gsm jitterbuffer=no tos=lowdelay [guest] type=user context=default callerid="Guest IAX User" [iaxtel] type=user context=default auth=rsa inkeys=iaxtel [iaxtel2] type=user context=default deny=0.0.0.0/0.0.0.0 permit=216.207.245.47/255.255.255.255 [demo] type=peer username=asterisk secret=supersecret host=216.207.245.47 [2203] type=friend host=dynamic secret=mypassword context=default qualify=300 nat=yes _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users