[Asterisk-Users] getting cid from spa3k pstn to *

[Randy Bush]

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:105 at my.asterisk.su 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:8081234567 at my.asterisk.su>;tag=42d678b4c352ea69o1
	    To: <sip:105 at my.asterisk.su>
	    Remote-Party-ID: CID Name    <sip:8081234567 at my.asterisk.su>;screen=yes;party=calling
	    Call-ID: b5c572d0-3664d6fd at 42.666.11.7
	    CSeq: 101 INVITE
	    Max-Forwards: 70
	    Contact: spa3k pstn <sip:spa3k at 42.666.11.7:5061>
	    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:105 at my.asterisk.su 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:8081234567 at my.asterisk.su>;tag=42d678b4c352ea69o1
	    To: <sip:105 at my.asterisk.su>;tag=as2741cf03
	    Call-ID: b5c572d0-3664d6fd at 42.666.11.7
	    CSeq: 101 ACK
	    Max-Forwards: 70
	    Contact: spa3k pstn <sip:spa3k at 42.666.11.7:5061>
	    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:105 at my.asterisk.su 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:spa3k at my.asterisk.su>;tag=8fc58211a0dc60f2o1
	    To: <sip:105 at my.asterisk.su>
	    Remote-Party-ID: spa3k pstn <sip:spa3k at my.asterisk.su>;screen=yes;party=calling
	    Call-ID: daed83bd-b2b66b36 at 42.666.11.7
	    CSeq: 101 INVITE
	    Max-Forwards: 70
	    Contact: spa3k pstn <sip:biwaa1 at 42.666.11.7:5061>
	    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