[asterisk-dev] Start testing with res_sip

[Ron Arts]

Hi,

I cannot get authentication to work with res_sip. I get a not found.
Is there a way to enable
debugging in res_sip? The SIP trace below appears automatically, and I
don't know how to
stop that, but OTOH chan_sip has debugging that shows where it's
looking for and why it can't
find the peer. I include my res_sip.conf below. It's very short.

Thanks,
Ron

<--- Received SIP request (906 bytes) from UDP:10.211.55.2:29754 --->
INVITE sip:1000 at 10.211.55.78;transport=udp SIP/2.0
Via: SIP/2.0/UDP
10.211.55.2:29754;branch=z9hG4bK-d8754z-5b1e7e3cb5437903-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:101 at 10.211.55.2:29754;transport=udp>
To: <sip:1000 at 10.211.55.78>
From: <sip:101 at 10.211.55.78>;tag=1a04960b
Call-ID: OWQxOWQ4NjI5NjZlMzk0ZTNjMzUzY2UzZmJkNWIzYWI
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria 3 release 3.5.2 stamp 70365
Content-Length: 345

v=0
o=- 1371710347513784 1 IN IP4 10.211.55.2
s=Bria 3 release 3.5.2 stamp 70365
c=IN IP4 10.211.55.2
t=0 0
m=audio 65276 RTP/AVP 122 120 9 8 0 18 101
a=rtpmap:122 opus/48000/2
a=fmtp:122 useinbandfec=1
a=rtpmap:120 SILK/16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Transmitting SIP response (338 bytes) to UDP:10.211.55.2:29754 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
10.211.55.2:29754;rport;received=10.211.55.2;branch=z9hG4bK-d8754z-5b1e7e3cb5437903-1---d8754z-
Call-ID: OWQxOWQ4NjI5NjZlMzk0ZTNjMzUzY2UzZmJkNWIzYWI
From: <sip:101 at 10.211.55.78>;tag=1a04960b
To: <sip:1000 at 10.211.55.78>;tag=26g1d2865MPHiX2eiJPPNYLgAPR831Ei
CSeq: 1 INVITE
Content-Length:  0


<--- Received SIP request (357 bytes) from UDP:10.211.55.2:29754 --->
ACK sip:1000 at 10.211.55.78;transport=udp SIP/2.0
Via: SIP/2.0/UDP
10.211.55.2:29754;branch=z9hG4bK-d8754z-5b1e7e3cb5437903-1---d8754z-;rport
Max-Forwards: 70
To: <sip:1000 at 10.211.55.78>;tag=26g1d2865MPHiX2eiJPPNYLgAPR831Ei
From: <sip:101 at 10.211.55.78>;tag=1a04960b
Call-ID: OWQxOWQ4NjI5NjZlMzk0ZTNjMzUzY2UzZmJkNWIzYWI
CSeq: 1 ACK
Content-Length: 0


res_sip.conf:

[localnetwork]
type=transport
protocol=udp
bind=0.0.0.0:5060

[endpointtemplate](!)
callerid_privacy=allowed_not_screened
context=testing
disallow=all
allow=g722
allow=alaw
dtmfmode=rfc4733
transport=localnetwork
direct_media=yes
send_pai=yes

[101](endpointtemplate)
type=endpoint
aors=101
auth=101
callerid="Ron Arts" <101>

[101]
type=aor
max_contacts=10
remove_existing=yes
mailboxes=101 at default

[101]
type=auth
auth_type=userpass
password=101
username=101


On Wed, Jun 19, 2013 at 5:01 PM, Ron Arts <ron.arts at oneip.nl> wrote:
> Thanks! I already got pjsip installed, just didn't know how to go from there.
> This will definitely get me started!
>
> Ron