asterisk users - Nov 2004 - getting callerid from spa3k to asterisk

i am still dying on this one, and my critical user, my fiance'e,
is giving me hell over it on my home test environment; even my
daytime job, for which i am prototyping, is more patient. :-)

i can not get caller-id from a call coming in to the spa3k pstn
to asterisk.  fwiw, this used to work with older * and spa3k
versions, but of course it could be something i did to configs.

essentially, if i tell the spa3k to pass callerid to *, the sip
session gets rejected by *.  since no one seemed to like to see
ethereal output, i have posted * sip debug form of the sessions.

does anyone have their spa3k and * config working that i could
look at?  or, if you can shoot the bug, i'll pay you US$100 by
paypal or whatever.

the spa3k configiuration
<http://rip.psg.com/~randy/spa3k.html>

sip debug with spa3k config set to
PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = NO
call accepted ok, but no callerid received by asterisk
<http://rip.psg.com/~randy/debug-0.txt>

sip debug with spa3k config set to
PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = YES
call rejected by asterisk
<http://rip.psg.com/~randy/debug-1.txt>

sip.conf entry

    [spa3k-in]
    type=friend     ; user fails to register
    host=dynamic
    port=5061
    auth=md5
    secret=dontbesilly
    qualify=1000
    dtmfmode=rfc2833
    canreinvite=yes
    context=ext-in42

extensions.conf for the incoming

    [ext-in42]
    exten => _X.,1,NoOp("ext-in42 cid=${CALLERIDNUM}")
    exten => _X.,2,SetVar(areacode=206)
    exten => _X.,3,SetVar(mailbox=1)
    exten => _X.,4,GoTo(ext-common,s,1)

    [ext-common]
    exten => s,1,NoOp("ext-common cid=${CALLERIDNUM}")
    exten => s,2,Background(zz-who-common)
    exten => i,1,Hangup()
    exten => t,1,GoTo(ext-common,s,1)
    include => speeddials
    include => extensions
    include => conferences
    include => applications

randy

Randy Bush wrote:
> essentially, if i tell the spa3k to pass callerid to *, the sip
> session gets rejected by *.  since no one seemed to like to see
> ethereal output, i have posted * sip debug form of the sessions.
You could maybe look at the autocreatepeer option for sip.conf - just a 
wild guess so YMMV.

BTW: Sorry didn't click through to your traces so this might be pointless!

:-) Think that's enough disclaimers!

-- 
Cheers,

Matt Riddell
_______________________________________________

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

> You could maybe look at the autocreatepeer option for sip.conf
that level of vulnerability would not seem to be a good approach
to solving some sort of sip/config problem :-)

the problem is in the sip handshake between the spa3k and *.  i
have been hoping a sip geek would have a chance to look at it.

randy

You could try adding the line insecure=very to the relevant section of
the sip.conf this would force asterisk to only validate the IP address
and not the user name (possibly  but it is woth a shot)



Jason


On Mon, 8 Nov 2004 10:28:03 -0800, Randy Bush <randy@psg.com>
wrote:
> > You could maybe look at the autocreatepeer option for sip.conf
> 
> that level of vulnerability would not seem to be a good approach
> to solving some sort of sip/config problem :-)
> 
> the problem is in the sip handshake between the spa3k and *.  i
> have been hoping a sip geek would have a chance to look at it.
> 
> randy
> 
> 
> 
> _______________________________________________
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ok, with a good pointer from Chris Stenton <jacs@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@asterisk.bogus.com SIP/2.0
Via: SIP/2.0/UDP 198.180.150.195:5061;branch=z9hG4bK-69580ec1
From: CallerName 
<sip:2065551212@asterisk.bogus.com>;tag=25aee11517d597a1o1
To: <sip:105@asterisk.bogus.com>
Remote-Party-ID: CallerName 
<sip:2065551212@asterisk.bogus.com>;screen=yes;party=calling
Call-ID: 8816a525-dbaa22d1@198.180.150.195
CSeq: 101 INVITE
Max-Forwards: 70
Contact: biwa 0431 <sip:spa3k-in@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'

> if i have two sip contexts for my spa3k, on inbound and
> one outbound, e.g.
> 
>     [spa3k-in]
>     type=friend
>     host=dynamic
>     port=5061
>     auth=md5
>     secret=pfui
>     qualify=1000
>     canreinvite=yes
>     context=ext-in42
> 
>     [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
> 
> 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.
i found the problem, or at least a work-around.

if i reverse the order of the above two sip contexts, the incoming
call is properly routed to the spa3k-in sip context as opposed to
the wrong one, spa3k-out.

my guess is that * is traversing a list and taking the first
context which has the ip address and port it wants without
checking the context name against the name which was received
over the wire.  so it depends on what order the contexts are
inserted in the list.

aiiiiiiiiiiiiiiiiiiiiii!

randy