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Sipura SPA-3000 PSTN Outgoing - Asterisk problem



 
 
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  #1  
Old March 7th 06, 11:50 PM posted to uk.telecom.voip
Peter Watson
external usenet poster
 
Posts: 70
Default Sipura SPA-3000 PSTN Outgoing - Asterisk problem

I've had my SPA-3000 successfully working for both incoming and outgoing
calls to/from the PSTN but outgoing calls have stopped working!!

I'm using [email protected] and I don't think I've changed anything...

Here's the debug output of the Sipura (on 10.0.0.252) when I place a
call from an Asterisk extension:

---Log starts

INVITE :5061 SIP/2.0

Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK384c1f04;rport

From: "100" ;tag=as6ba389a0

To: :5061

Contact:

Call-ID:

CSeq: 102 INVITE

User-Agent: Pete's PBX

Max-Forwards: 70

Date: Tue, 07 Mar 2006 23:36:53 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Type: application/sdp

Content-Length: 197



v=0

o=root 2881 2881 IN IP4 10.0.0.1

s=session

c=IN IP4 10.0.0.1

t=0 0

m=audio 16322 RTP/AVP 0 3 8

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:8 PCMA/8000

a=silenceSuppff - - - -



SIP:Challenge INVITE
[1:5061]-10.0.0.1:5060
[1:5061]-10.0.0.1:5060
SIP/2.0 401 Unauthorized

To: :5061;tag=af3fa47e511 ea8bbi1

From: "100" ;tag=as6ba389a0

Call-ID:

CSeq: 102 INVITE

Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK384c1f04

Server: Sipura/SPA3000-3.1.7(GWg)

WWW-Authenticate: Digest realm="10.0.0.1", nonce="a3d355a9", qop="auth",
algorithm=md5

Content-Length: 0

---Log ends

and here's the section from sip_addition.conf

[sipura]
type=peer
secret=********
port=5061
insecure=very
host=10.0.0.252
fromuser=Asterisk
dtmfmode=inband
context=from-internal
canreinvite=no
auth=md5

[pstn-incoming]
type=friend
port=5061
insecure=very
host=10.0.0.252
host=dynamic
dtmfmode=inband
disallow=all
context=from-internal
canreinvite=no
allow=ulaw

I've defined Voip User 1 as 'Asterisk' and entered the correct password
on the SPA-3000. Screenshot at:

http://www.pwatson.org/advanced.htm

Any help most welcome!

Thanks,

Peter



  #2  
Old March 8th 06, 10:11 PM posted to uk.telecom.voip
alexd
external usenet poster
 
Posts: 331
Default Sipura SPA-3000 PSTN Outgoing - Asterisk problem

Peter Watson wrote:

host=10.0.0.252
host=dynamic


Hmm, looks a bit dubious specifying host= twice.

Any help most welcome!


Drop into an asterisk console with 'asterisk -Rc', type
'sip debug peer sipura' and see if that gets you anywhere.

--
http://ale.cx/ (AIM:troffasky) )
22:02:46 up 2:52, 1 user, load average: 0.04, 0.03, 0.00
This is my BOOOOOOOOOOOOOOOOOOOOOMSTICK

  #3  
Old March 8th 06, 10:58 PM posted to uk.telecom.voip
Peter Watson
external usenet poster
 
Posts: 70
Default Sipura SPA-3000 PSTN Outgoing - Asterisk problem

alexd wrote:
Peter Watson wrote:

host=10.0.0.252
host=dynamic


Hmm, looks a bit dubious specifying host= twice.

Any help most welcome!


Drop into an asterisk console with 'asterisk -Rc', type
'sip debug peer sipura' and see if that gets you anywhere.

Thanks Alex, I've tidied that bit up (removed the host=dynamic)
Asterisk debug messages mirror the Sipura logs btw

I've just tried adding username=Asterisk to [sipura] and it has started
working again. I don't recall removing this so I've no idea why it
stopped working.

I now need to read up on the difference between 'fromuser' and 'username'!!

Peter
  #4  
Old March 9th 06, 08:24 PM posted to uk.telecom.voip
alexd
external usenet poster
 
Posts: 331
Default Sipura SPA-3000 PSTN Outgoing - Asterisk problem

Peter Watson wrote:

Thanks Alex, I've tidied that bit up (removed the host=dynamic)
Asterisk debug messages mirror the Sipura logs btw

I've just tried adding username=Asterisk to [sipura] and it has started
working again. I don't recall removing this so I've no idea why it
stopped working.


It stopped working because hosts specified with an IP address [eg
host=10.0.0.252] must have username= specified. This would hopefully appear
in error logs somewhere.

I now need to read up on the difference between 'fromuser' and
'username'!!


I believe one is the username you use to authenticate with Asterisk with,
and the other is the username component of your SIP URI, eg
. So you would specify both if you wanted them to
be different.

Also, it's handy to use the 'username' to contact a peer before it's
registered with Asterisk, which would happen if your server rebooted in the
middle of an expiry.

--
http://ale.cx/ (AIM:troffasky) )
20:04:10 up 1 day, 54 min, 1 user, load average: 0.00, 0.02, 0.00
This is my BOOOOOOOOOOOOOOOOOOOOOMSTICK

 




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