[Asterisk-Users] Problem with Vegastream 50 BRI

[Dean Collins]

This was posted to me by Vegastream tech support in regards to your
earlier question (I emailed them your question last week), sorry I'm
just getting familiar with both boxes so I'm not able to help you at
this stage I have just signed a deal for distribution of the vegastream
here in Australia and for anyone here on the list who is interested it
will be shortly certified with Comindico for use on the new ecall
network.

Cheers,
Dean




Hi,

We don't have any particular hands-on experience with Asterisk
ourselves, although a number of customers use it. My main observation
would be:

- The asterisk server isn't finding the phone number 57228047 and is
  returning a 404 (obviously)
- As the follow-up poster observed, asterisk has "tlsgw" set as the
default
  context and has entered the phone number extension for tlsconf, but is
  using the "sip" context instead, hence doesn't find the number
- Quite possibly this is because the Vega doesn't do TLS (Transport
Layer
  Security), and therefore that context isn't applicable, though that's
  just a guess since I haven't used asterisk...

Cheers,
Bryan

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Michael
Devenijn
Sent: Sunday, 21 March 2004 2:37 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Problem with Vegastream 50 BRI

Here is a sip log from my vegastream 50BRI to my asterisk box and i
can't figure out why the call doesn't go trough ...


sip.conf extract : 

[gw001]
type=friend
host=dynamic
defaultip=192.168.0.12
nat=no
dtmfmode=rfc2833
canreinvite=yes
qualify=no
context=tlsgw



extensions.conf extract (from the contact [tlsgw]) :

exten => 57228047,Dial(SIP/cs001,40,tr) 
...




Sip read:
INVITE sip:57228047 at 192.168.0.15:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D
From: 478758923 <sip:478758923 at 192.168.0.12>;tag=0000-0002-3A81BAD9
To: <sip:57228047 at 192.168.0.15>
Max-Forwards: 70
Call-ID: 0001-0002-94F67DB7-0 at 192.168.0.12
CSeq: 83606 INVITE
Contact: <sip:478758923 at 192.168.0.12:5060;maddr=192.168.0.12>
Supported: replaces
Allow: INVITE,ACK,BYE,CANCEL,INFO,NOTIFY,OPTIONS,REFER
Accept-Language: en
Content-Type: application/sdp
Remote-Party-ID: 478758923
<sip:478758923 at 192.168.0.12>;party=calling;screen=no;privacy=off
Content-Length: 178
 
v=0
o=Vega50 3 1 IN IP4 192.168.0.12
s=Sip Call
t=0 0
m=audio 10004 RTP/AVP 8 0 18
c=IN IP4 192.168.0.12
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
 
14 headers, 9 lines
Using latest request as basis request
Sending to 192.168.0.12 : 5060 (non-NAT)
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMA
Found description format PCMU
Found description format G729
Capabilities: us - 524302, them - 268/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for 57228047 in sip
Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D
From: 478758923 <sip:478758923 at 192.168.0.12>;tag=0000-0002-3A81BAD9
To: <sip:57228047 at 192.168.0.15>;tag=as1fa83a23
Call-ID: 0001-0002-94F67DB7-0 at 192.168.0.12
CSeq: 83606 INVITE
ser-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:57228047 at 192.168.0.15>
Content-Length: 0
 
 
 to 192.168.0.12:5060
dkmapbx*CLI>
 
Sip read:
ACK sip:57228047 at 192.168.0.15:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D
From: 478758923 <sip:478758923 at 192.168.0.12>;tag=0000-0002-3A81BAD9
To: <sip:57228047 at 192.168.0.15>
Max-Forwards: 70
Call-ID: 0001-0002-94F67DB7-0 at 192.168.0.12
CSeq: 83606 ACK
Contact: <sip:478758923 at 192.168.0.12:5060;maddr=192.168.0.12>
Content-Length: 0
 
 
9 headers, 0 lines


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