in order to get the cid from the spa3k to *, i need to turn on
PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = YES
this produces a sip invite as follows:
Frame 1 (1092 bytes on wire, 1092 bytes captured)
Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72
Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr: 666.42.7.11
(666.42.7.11)
User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060)
Session Initiation Protocol
Request-Line: INVITE sip:[EMAIL PROTECTED] SIP/2.0
Method: INVITE
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-388cd0e7
From: CID Name <sip:[EMAIL PROTECTED]>;tag=42d678b4c352ea69o1
To: <sip:[EMAIL PROTECTED]>
Remote-Party-ID: CID Name <sip:[EMAIL
PROTECTED]>;screen=yes;party=calling
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: spa3k pstn <sip:[EMAIL PROTECTED]:5061>
Expires: 240
User-Agent: Sipura/SPA3000-2.0.11(GWa)
Content-Length: 430
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
Message body
Session Description Protocol
note that the From: has the cid, as does the Remote-Party-ID:. and the
Contact: has the spa3k's id and display name. as the sip.conf entry looks
like
[spa3k]
type=friend
host=dynamic
port=5061
auth=md5
secret=hidden
qualify=1000
dtmfmode=rfc2833
canreinvite=yes
context=spa3k-ext
the From: and/or Remote-Party-ID: cause asterisk respond with 407 Proxy
Authentication Required, to which the spa3k responds
Frame 3 (450 bytes on wire, 450 bytes captured)
Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72
Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr: 666.42.7.11
(666.42.7.11)
User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060)
Session Initiation Protocol
Request-Line: ACK sip:[EMAIL PROTECTED] SIP/2.0
Method: ACK
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-388cd0e7
From: CID Name <sip:[EMAIL PROTECTED]>;tag=42d678b4c352ea69o1
To: <sip:[EMAIL PROTECTED]>;tag=as2741cf03
Call-ID: [EMAIL PROTECTED]
CSeq: 101 ACK
Max-Forwards: 70
Contact: spa3k pstn <sip:[EMAIL PROTECTED]:5061>
User-Agent: Sipura/SPA3000-2.0.11(GWa)
Content-Length: 0
and it all goes to hell from there.
if i set the spa3k config to have
PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = NO
Frame 1 (1072 bytes on wire, 1072 bytes captured)
Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72
Internet Protocol, Src Addr: 42.666.11.7 (42.666.11.7), Dst Addr: 666.42.7.11
(666.42.7.11)
User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060)
Session Initiation Protocol
Request-Line: INVITE sip:[EMAIL PROTECTED] SIP/2.0
Method: INVITE
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 42.666.11.7:5061;branch=z9hG4bK-f5998d8a
From: spa3k pstn <sip:[EMAIL PROTECTED]>;tag=8fc58211a0dc60f2o1
To: <sip:[EMAIL PROTECTED]>
Remote-Party-ID: spa3k pstn <sip:[EMAIL
PROTECTED]>;screen=yes;party=calling
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: spa3k pstn <sip:[EMAIL PROTECTED]:5061>
Expires: 240
User-Agent: Sipura/SPA3000-2.0.11(GWa)
Content-Length: 430
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
Message body
Session Description Protocol
the connection completes, but asterisk does not have the pstn caller id.
randy
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
