I am having trouble getting incoming calls to work with my SIP provider.
I have tried any number of settings in the “incoming” section of my trunk definition, all to no avail.
I have tried to set up an inbound route with the DID, I have tried adding the DID to the extension, all to no avail.
The Asterisk box is has its own official external IP address, so there should be no NAT issues.
I get absolutely NO output in the CLI (despite verbosity=110) unless I turn SIP debugging on; then I get the following:
pbx*CLI>
<— SIP read from 196.17.242.198:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 196.17.242.198:5060;branch=z9hG4bK6fa65e89;rport
From: “06649662860” sip:[email protected];tag=as090ae269
To: sip:[email protected]
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 23 Jul 2008 12:26:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 2854 2854 IN IP4 196.17.242.198
s=session
c=IN IP4 196.17.242.198
t=0 0
m=audio 17534 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<------------->
— (13 headers 10 lines) —
Sending to 196.17.242.198 : 5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘voipco-vcs’
<— Reliably Transmitting (NAT) to 196.17.242.198:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 196.17.242.198:5060;branch=z9hG4bK6fa65e89;received=196.17.242.198;rport=5060
From: “06649662860” sip:[email protected];tag=as090ae269
To: sip:[email protected];tag=as5b68b468
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5e5b4a4e"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)
pbx*CLI>
<— SIP read from 196.17.242.198:5060 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 196.17.242.198:5060;branch=z9hG4bK6fa65e89;rport
From: “06649662860” sip:[email protected];tag=as090ae269
To: sip:[email protected];tag=as5b68b468
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
<------------->
— (10 headers 0 lines) —
pbx*CLI>
<— SIP read from 196.17.242.198:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 196.17.242.198:5060;branch=z9hG4bK7f227b73;rport
From: “06649662860” sip:[email protected];tag=as090ae269
To: sip:[email protected]
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username=“7682253”, realm=“asterisk”, algorithm=MD5, uri="sip:[email protected]", nonce=“5e5b4a4e”, response=“7d62c9c9c86f02fb14dfe41e43258f51”, opaque=""
Date: Wed, 23 Jul 2008 12:26:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 2854 2855 IN IP4 196.17.242.198
s=session
c=IN IP4 196.17.242.198
t=0 0
m=audio 17534 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<------------->
— (14 headers 10 lines) —
Sending to 196.17.242.198 : 5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘voipco-vcs’
<— Reliably Transmitting (NAT) to 196.17.242.198:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 196.17.242.198:5060;branch=z9hG4bK7f227b73;received=196.17.242.198;rport=5060
From: “06649662860” sip:[email protected];tag=as090ae269
To: sip:[email protected];tag=as5b68b468
Call-ID: [email protected]
CSeq: 103 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 6400 ms (Method: INVITE)
pbx*CLI>
<— SIP read from 196.17.242.198:5060 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 196.17.242.198:5060;branch=z9hG4bK7f227b73;rport
From: “06649662860” sip:[email protected];tag=as090ae269
To: sip:[email protected];tag=as5b68b468
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
<------------->
— (10 headers 0 lines) —
pbx*CLI> Reliably Transmitting (NAT) to 196.17.242.198:5060:
OPTIONS sip:bsip02.voipco.net SIP/2.0
Via: SIP/2.0/UDP 196.17.247.227:5060;branch=z9hG4bK7691ac01;rport
From: “Unknown” sip:[email protected];tag=as1caddd31
To: sip:bsip02.voipco.net
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 23 Jul 2008 12:24:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
pbx*CLI>
<— SIP read from 196.17.242.198:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 196.17.247.227:5060;branch=z9hG4bK7691ac01;received=196.17.247.227;rport=5060
From: “Unknown” sip:[email protected];tag=as1caddd31
To: sip:bsip02.voipco.net;tag=as137178c8
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0
<------------->
— (10 headers 0 lines) —
pbxCLI> Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
pbxCLI>