[Asterisk-Users] getting callerid from spa3k to asterisk

[Randy Bush]

ok, with a good pointer from Chris Stenton <jacs at gnome.co.uk>,
i found the problem.

if i have two sip contexts for my spa3k, on inbound and
one outbound, e.g.

    [spa3k-out]
    type=peer
    auth=md5
    secret=pfui
    username=outpass
    fromuser=outpass
    host=spa3k.bogus.com
    port=5061
    nat=no
    canreinvite=yes
    context=ext-in42

    [spa3k-in]
    type=friend
    host=dynamic
    port=5061
    auth=md5
    secret=pfui
    qualify=1000
    canreinvite=yes
    context=ext-in42

and the spa3k's PSTN / Subscriber Information / User ID: = spack-in,

the incoming connection from spa3k to * is being routed to the
spa3k-out context, not the spa3-in context.  see appended.

i suspect this is a bug in * 1.0.1.

so, until the problem is diagnosed, how do i work around it.
as the spa3k is registered, i tried to remove the spa3k-out
context entirely.  callerid now works.  yes!

but ...  if i try to place an outbound call using the spa3k-in
context, the call is sent to the spa3k, but it just gives me
the pstn's dialtone, and does not dial the number.  my spa3k
config is in <http://rip.psg.com/~randy/spa3k.html>.

so how do i place a call out the spa3k pstn without a separate
outbound context?

randy

---

Sip read: 
INVITE sip:105 at asterisk.bogus.com SIP/2.0
Via: SIP/2.0/UDP 198.180.150.195:5061;branch=z9hG4bK-69580ec1
From: CallerName  <sip:2065551212 at asterisk.bogus.com>;tag=25aee11517d597a1o1
To: <sip:105 at asterisk.bogus.com>
Remote-Party-ID: CallerName  <sip:2065551212 at asterisk.bogus.com>;screen=yes;party=calling
Call-ID: 8816a525-dbaa22d1 at 198.180.150.195
CSeq: 101 INVITE
Max-Forwards: 70
Contact: biwa 0431 <sip:spa3k-in at 198.180.150.195:5061>
Expires: 240
User-Agent: Sipura/SPA3000-2.0.11(GWa)
Content-Length: 428
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 8805171 8805171 IN IP4 198.180.150.195
s=-
c=IN IP4 198.180.150.195
t=0 0
m=audio 16396 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

15 headers, 19 lines
Using latest request as basis request
Sending to 198.180.150.195 : 5061 (non-NAT)
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 198.180.150.195:16396
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0xe(GSM|ULAW|ALAW), peer - audio=0x51d(G723|ULAW|ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Found peer 'spa3k-out'