Maybe this will be of help.
http://voip-forum.tmcnet.com/voip-forum/forum/forum_posts.asp?TID=1707&PN=0&TPN=1
----- Original Message ----- From: "Randy Bush" <[EMAIL PROTECTED]>
To: "splatters" <[EMAIL PROTECTED]>
Sent: Friday, November 12, 2004 9:13 PM
Subject: [Asterisk-Users] getting callerid from spa3k to asterisk
ok, with a good pointer from Chris Stenton <[EMAIL PROTECTED]>, 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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 198.180.150.195:5061;branch=z9hG4bK-69580ec1
From: CallerName <sip:[EMAIL PROTECTED]>;tag=25aee11517d597a1o1
To: <sip:[EMAIL PROTECTED]>
Remote-Party-ID: CallerName <sip:[EMAIL PROTECTED]>;screen=yes;party=calling
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: biwa 0431 <sip:[EMAIL PROTECTED]: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'
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