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Alejandro Vargas

2006-Feb-20 01:44 UTC

[Asterisk-Users] spa3000

I'm trying to get working a spa3000 with asterisk. My problem is I
cant get wroking the incomming calls (I installed the lastest
firmware). My problem is asterisk is rejecting the authentication from
the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I
think I placed the username and password correctly...

Sip.conf says this:
[linea2]
username=linea2
type=peer
secret=1111
context=from-pstn
auth=md5

[PSTN_2]
username=line2
type=peer
secret=1111
qualify=yes
port=5061
nat=no
host=192.168.0.20
context=from-pstn
canreinvite=no
auth=md5

The sip debug says this:

<-- SIP read from 192.168.0.20:5061:
INVITE sip:s@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-fc209f89
From: <sip:987073366@192.168.0.254>;tag=be447a1af149c461o1
To: <sip:s@192.168.0.254:5060>
Call-ID: a459834e-816b2cbd@192.168.0.20
CSeq: 101 INVITE
Max-Forwards: 70
Contact: <sip:987073366@192.168.0.20:5061>
Expires: 240
User-Agent: Sipura/SPA3000-3.1.7(GWg)
Content-Length: 426
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 23156020 23156020 IN IP4 192.168.0.20
s=-
c=IN IP4 192.168.0.20
t=0 0
m=audio 16478 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

--- (14 headers 19 lines)---
Using INVITE request as basis request - a459834e-816b2cbd@192.168.0.20
Sending to 192.168.0.20 : 5061 (non-NAT)
Found peer 'PSTN_2'
Reliably Transmitting (no NAT) to 192.168.0.20:5061:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-fc209f89;received=192.168.0.20
From: <sip:987073366@192.168.0.254>;tag=be447a1af149c461o1
To: <sip:s@192.168.0.254:5060>;tag=as535b07db
Call-ID: a459834e-816b2cbd@192.168.0.20
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@192.168.0.254>
Proxy-Authenticate: Digest realm="asterisk",
nonce="5a2eee21"
Content-Length: 0


---
Scheduling destruction of call 'a459834e-816b2cbd@192.168.0.20' in 15000
ms
server*CLI>
<-- SIP read from 192.168.0.20:5061:
ACK sip:s@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-fc209f89
From: <sip:987073366@192.168.0.254>;tag=be447a1af149c461o1
To: <sip:s@192.168.0.254:5060>;tag=as535b07db
Call-ID: a459834e-816b2cbd@192.168.0.20
CSeq: 101 ACK
Max-Forwards: 70
Contact: <sip:987073366@192.168.0.20:5061>
User-Agent: Sipura/SPA3000-3.1.7(GWg)
Content-Length: 0


--- (10 headers 0 lines)---
server*CLI>
<-- SIP read from 192.168.0.20:5061:
INVITE sip:s@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-2e547cf5
From: <sip:987073366@192.168.0.254>;tag=be447a1af149c461o1
To: <sip:s@192.168.0.254:5060>
Call-ID: a459834e-816b2cbd@192.168.0.20
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="linea2",realm="asterisk",nonce="5a2eee21",uri="sip:s@192.168.0.254:5060",algorithm=MD5,response="f18750c7e09707b6e76e0c6c08f10b77"
Contact: <sip:987073366@192.168.0.20:5061>
Expires: 240
User-Agent: Sipura/SPA3000-3.1.7(GWg)
Content-Length: 426
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 23156020 23156020 IN IP4 192.168.0.20
s=-
c=IN IP4 192.168.0.20
t=0 0
m=audio 16478 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 INVITE request as basis request - a459834e-816b2cbd@192.168.0.20
Sending to 192.168.0.20 : 5061 (non-NAT)
Found peer 'PSTN_2'
Reliably Transmitting (no NAT) to 192.168.0.20:5061:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-2e547cf5;received=192.168.0.20
From: <sip:987073366@192.168.0.254>;tag=be447a1af149c461o1
To: <sip:s@192.168.0.254:5060>;tag=as535b07db
Call-ID: a459834e-816b2cbd@192.168.0.20
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@192.168.0.254>
Content-Length: 0


---
server*CLI>
<-- SIP read from 192.168.0.20:5061:
ACK sip:s@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-2e547cf5
From: <sip:987073366@192.168.0.254>;tag=be447a1af149c461o1
To: <sip:s@192.168.0.254:5060>;tag=as535b07db
Call-ID: a459834e-816b2cbd@192.168.0.20
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest
username="linea2",realm="asterisk",nonce="5a2eee21",uri="sip:s@192.168.0.254:5060",algorithm=MD5,response="104aa010d2f90b4a69c56b0ebf0991d3"
Contact: <sip:987073366@192.168.0.20:5061>
User-Agent: Sipura/SPA3000-3.1.7(GWg)
Content-Length: 0


--- (11 headers 0 lines)---
Destroying call 'a459834e-816b2cbd@192.168.0.20'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 80.239.235.200:5060:
OPTIONS sip:sip1.sipdiscount.com SIP/2.0
Via: SIP/2.0/UDP 81.172.52.3:5060;branch=z9hG4bK0596a5f2;rport
From: "Unknown" <sip:Unknown@81.172.52.3>;tag=as6c5807a2
To: <sip:sip1.sipdiscount.com>
Contact: <sip:Unknown@81.172.52.3>
Call-ID: 43c17608446ba56231766bb82c8e350e@81.172.52.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 20 Feb 2006 08:16:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0




--
Alejandro Vargas

Rich Adamson

2006-Feb-20 06:18 UTC

head link

[Asterisk-Users] spa3000

> I'm trying to get working a spa3000 with asterisk. My problem is I
> cant get wroking the incomming calls (I installed the lastest
> firmware). My problem is asterisk is rejecting the authentication from
> the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I
> think I placed the username and password correctly...
> 
> Sip.conf says this:
> [linea2]
> username=linea2
> type=peer
> secret=1111
> context=from-pstn
> auth=md5
> 
> [PSTN_2]
> username=line2
> type=peer
> secret=1111
> qualify=yes
> port=5061
> nat=no
> host=192.168.0.20
> context=from-pstn
> canreinvite=no
> auth=md5
Try something like this instead:
[linea2]
username=linea2
type=friend
secret=1111
context=from-pstn

[line2]
username=line2
type=friend
secret=1111
qualify=yes
nat=no
context=from-pstn
canreinvite=no

Note that I changed this to type=friend, removed the host=, and
removed the auth=md5. The above works just fine here with a spa3k.

In the [linea2] section, you want the fxs port to be able to place
calls as well as receive calls, therefore use type=friend. The same
with section [line2]. Note that I also changed the [PSTN_2] to [line2]
to match the username= line. Watch the upper/lower case matching to
your spa3k configuration.

After you get the above working correctly, then you can play around
with auth=md5, etc.

Chris Mason (Lists)

2006-Feb-20 07:56 UTC

head link

[Asterisk-Users] spa3000

I don't mess with configuring these, the wizard on voxilla.com does 
everything except set the right context. Try using default for 
everything to get it working then separate as needed.

-- 
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: netconcepts_anguilla@yahoo.com 


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