In just about every combination of configurations I have tried (unless they were blatantly incorrect) the regular CLI say nothing (except when I tried to install AMP which gave me a permission error in the spooler).
My existing config I will put below. The debug says this: <-------------> --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 66.176.193.46 : 11214 (no NAT) <--- Transmitting (no NAT) to 66.176.193.46:11214 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46 From: <sip:[EMAIL PROTECTED]>;tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 949 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 <------------> <--- Transmitting (no NAT) to 66.176.193.46:11214 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46 From: <sip:[EMAIL PROTECTED]>;tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0 To: <sip:[EMAIL PROTECTED]>;tag=as0d27cf25 Call-ID: [EMAIL PROTECTED] CSeq: 949 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3721d6a7" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) <--- SIP read from 66.176.193.46:4024 ---> REGISTER sip:66.109.17.92 SIP/2.0 Via: SIP/2.0/UDP 66.176.193.46:11214 Max-Forwards: 70 From: <sip:[EMAIL PROTECTED]>;tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 950 REGISTER Contact: <sip:66.176.193.46:11214>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER" User-Agent: RTC/1.2.4949 Authorization: Digest username="UXMC", realm="asterisk", algorithm=MD5, uri="sip:66.109.17.92", nonce="3721d6a7", response="4d92865d351ad10e7f8ff0b4eabfbbe8" Event: registration Allow-Events: presence Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 66.176.193.46 : 11214 (no NAT) <--- Transmitting (no NAT) to 66.176.193.46:11214 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46 From: <sip:[EMAIL PROTECTED]>;tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 950 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 <------------> -- Saved useragent "RTC/1.2.4949" for peer UXMC <--- Transmitting (no NAT) to 66.176.193.46:11214 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46 From: <sip:[EMAIL PROTECTED]>;tag=5eef40c7be1b41e3a7e59df9ae0fc17a;epid=e4fa213bd0 To: <sip:[EMAIL PROTECTED]>;tag=as0d27cf25 Call-ID: [EMAIL PROTECTED] CSeq: 950 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: <sip:66.176.193.46:11214>;expires=120 Date: Wed, 30 May 2007 15:45:39 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) <--- SIP read from 66.176.193.46:4024 ---> INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 66.176.193.46:11214 Max-Forwards: 70 From: "UXMC" <sip:[EMAIL PROTECTED]>;tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: <sip:66.176.193.46:11214> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 448 v=0 o=- 0 0 IN IP4 66.176.193.46 s=session c=IN IP4 66.176.193.46 b=CT:1000 t=0 0 m=audio 32744 RTP/AVP 97 111 112 6 0 8 4 5 3 101 a=rtpmap:97 red/8000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:112 G7221/16000 a=fmtp:112 bitrate=24000 a=rtpmap:6 DVI4/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (11 headers 20 lines) --- Sending to 66.176.193.46 : 11214 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] <--- Reliably Transmitting (no NAT) to 66.176.193.46:11214 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46 From: "UXMC" <sip:[EMAIL PROTECTED]>;tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0 To: <sip:[EMAIL PROTECTED]>;tag=as55eebfec Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e7f413d" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) Found user 'UXMC' <--- SIP read from 66.176.193.46:4024 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 66.176.193.46:11214 Max-Forwards: 70 From: "UXMC" <sip:[EMAIL PROTECTED]>;tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0 To: <sip:[EMAIL PROTECTED]>;tag=as55eebfec Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK User-Agent: RTC/1.2 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 66.176.193.46:4024 ---> INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 66.176.193.46:11214 Max-Forwards: 70 From: "UXMC" <sip:[EMAIL PROTECTED]>;tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: <sip:66.176.193.46:11214> User-Agent: RTC/1.2 Proxy-Authorization: Digest username="UXMC", realm="asterisk", algorithm=MD5, uri="sip:[EMAIL PROTECTED]", nonce="5e7f413d", response="a42405bdc4b7273e954ebcf9d26851d7" Content-Type: application/sdp Content-Length: 448 v=0 o=- 0 0 IN IP4 66.176.193.46 s=session c=IN IP4 66.176.193.46 b=CT:1000 t=0 0 m=audio 32744 RTP/AVP 97 111 112 6 0 8 4 5 3 101 a=rtpmap:97 red/8000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:112 G7221/16000 a=fmtp:112 bitrate=24000 a=rtpmap:6 DVI4/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (12 headers 20 lines) --- Sending to 66.176.193.46 : 11214 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found user 'UXMC' Found RTP audio format 97 Found RTP audio format 111 Found RTP audio format 112 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 5 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 66.176.193.46:32744 Found description format red for ID 97 Found description format SIREN for ID 111 Found description format G7221 for ID 112 Found description format DVI4 for ID 6 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 Found description format DVI4 for ID 5 Found description format GSM for ID 3 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc3f (g723|gsm|ulaw|alaw|g726|adpcm|ilbc|g726aal2)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 66.176.193.46:32744 Looking for 19544790554 in internal (domain 66.109.17.92) <--- Reliably Transmitting (no NAT) to 66.176.193.46:11214 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 66.176.193.46:11214;received=66.176.193.46 From: "UXMC" <sip:[EMAIL PROTECTED]>;tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0 To: <sip:[EMAIL PROTECTED]>;tag=as55eebfec Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) <--- SIP read from 66.176.193.46:4024 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 66.176.193.46:11214 Max-Forwards: 70 From: "UXMC" <sip:[EMAIL PROTECTED]>;tag=eae0276709cf472b97ca728728f23809;epid=e4fa213bd0 To: <sip:[EMAIL PROTECTED]>;tag=as55eebfec Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK User-Agent: RTC/1.2 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Existing config Present "non-working" config extensions.conf [internal] exten => _1XXXXXXXXXX,1,DIAL,(IAX2/UXMC,30,tr) exten => s,1,Answer() IAX.conf [general] port=4569 bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no tos=lowdelay autokill=yes register => UXMC:[EMAIL PROTECTED] [teliax] context=default type=friend host=voip-co3.teliax.com auth=md5 user=UXMC secret=xxxxxxxxxxxx disallow=all allow=ulaw allow=alaw allow=gsm with: *CLI> iax2 show registry Host dnsmgr Username Perceived Refresh State 207.174.202.4:4569 N UXMC xx.xxx.xx.xx:4569 60 Registe sip.conf [general] context=internal srvlookup=yes allowguest=yes allowoverlap=no [UXMC] user=UXMC context=internal type=friend qualify=yes nat=no secret=xxxxxxx canreinvite=no host=dynamic nat=no -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder Singh Sent: Wednesday, May 30, 2007 5:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Still cannot make a single call from asteriskvia softphone to pstn!!!!!!! Can you post some output from asterisk cli output while you make call ? On 30/05/07, BSumrall <[EMAIL PROTECTED]> wrote: > > > > > after 18 hours, over 200 pages of reading, a complete reinstall of asterisk > I am down to this. > > extensions.conf > > [globals] > CONSOLE=Console/dsp > IAXINFO=guest > TRUNK=Zap/g2 > TRUNKMSD=1 > > [default] > exten => 8005181896,1,Dial,(IAX2/UXMC) > exten => s,1,Answer() > > (I tried) > exten => _1XXXXXXXXXX,1,DIAL,(IAX2/teliax/${EXTEN},30,tr) > (as well) > > iax.conf > > [general] > port=4569 > bandwidth=low > disallow=lpc10 > jitterbuffer=no > forcejitterbuffer=no > tos=lowdelay > autokill=yes > > register => xxxx:[EMAIL PROTECTED] > > [teliax] > context=default > type=friend > host=voip-co3.teliax.com > auth=md5 > user=xxxx > secret=xxxxxxxxx > disallow=all > allow=ulaw > allow=alaw > allow=gsm > > sip.conf > > [UXMC] > user=xxxxxxx > context=internal > type=friend > qualify=yes > nat=no > secret=xxxxxxxx > canreinvite=no > host=dynamic > nat=no > > If I put back previous config, I can call into the 1800 number and here > that silly chick heckle me from my server! > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users