[Asterisk-Users] getting callerid from spa3k to asterisk

[Steve Totaro]

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" <randy at psg.com>
To: "splatters" <asterisk-users at lists.digium.com>
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 <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'
>
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