chan_sip asking for CALLING party in sip.conf?

Hey all,

The conf below worked fine for 1.4.0 but breaks in yesterday’s SVN (dunno how to tell the SVN version offhand).

From what I can read in the debug output, chan_sip’s asking for there to be an entry in sip.conf for the CALLING party’s number. That can’t be right. Am I doing something wrong, or is something broken in the version I nabbed?

Thanks,

Brad Waite

;
; SIP Configuration for Asterisk
;
[general]
context=inbound ; Default for incoming calls
allowguest=no ; Refuse unauthenticated users
register=foo:bar@sipprovider-in2:5060

[sipprovider-in2]
insecure=port,invite
type=friend
allow=all
canreinvite=no
context=inbound
host=inbound2.sipprovider.net
;host=62.4.147.22
secret=bar
username=foo

[sipprovider-out]
type=friend
context=outbound
host=outbound1.sipprovider.net
;host=62.4.147.22
defaultip=62.4.147.22
insecure=port,invite
username=foo
fromuser=foo
secret=bar
trustrpid=yes
sendrpid=yes
allow=all
canreinvite=no

[100]
type=friend
username=100
secret=secret
host=dynamic
nat=yes
mailbox=Foo Bar@MyContext
context=MyContext
Subscribecontext=MyContext

[101]
type=friend
username=101
secret=secret
host=dynamic
nat=yes
mailbox=Foo Bar@MyContext
context=MyContext
Subscribecontext=MyContext

[102]
type=friend
username=102
secret=secret
host=dynamic
nat=yes
mailbox=Foo Bar@MyContext
context=MyContext
Subscribecontext=MyContext

[103]
type=friend
username=103
secret=secret
host=dynamic
nat=yes
mailbox=Foo Bar@MyContext
context=MyContext
Subscribecontext=MyContext

Inbound call from 8015551212:

<— SIP read from 62.4.147.22:5060 —>
INVITE sip:8011234567@216.220.238.150 SIP/2.0
Via: SIP/2.0/UDP 62.4.147.22:5060;branch=z9hG4bK278fbad0;rport
From: “8015551212” sip:8015551212@62.4.147.22;tag=as2662575a
To: sip:8011234567@216.220.238.150
Contact: sip:8015551212@62.4.147.22
Call-ID: 7f3d3a0120100f794b239bf00e39dfac@62.4.147.22
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 14 Feb 2007 04:05:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 308

v=0
o=root 28155 28155 IN IP4 62.4.147.22
s=session
c=IN IP4 62.4.147.22
t=0 0
m=audio 12284 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

<------------->
— (13 headers 14 lines) —
Sending to 62.4.147.22 : 5060 (NAT)
Using INVITE request as basis request - 7f3d3a0120100f794b239bf00e39dfac@62.4.147.22
No user ‘8015551212’ in SIP users list
No matching peer for ‘8015551212’ from ‘62.4.147.22:5060’
[Feb 13 21:00:24] NOTICE[62651]: chan_sip.c:13645 handle_request_invite: Failed to authenticate user “8015551212” <s
ip:8015551212@62.4.147.22>;tag=as2662575a

<— Reliably Transmitting (NAT) to 62.4.147.22:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 62.4.147.22:5060;branch=z9hG4bK278fbad0;received=64.2.142.27;rport=5060
From: “8015551212” sip:8015551212@62.4.147.22;tag=as2662575a
To: sip:8011234567@216.220.238.150;tag=as142fc13c
Call-ID: 7f3d3a0120100f794b239bf00e39dfac@62.4.147.22
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘7f3d3a0120100f794b239bf00e39dfac@62.4.147.22’ in 32000 ms (Method: INVITE)

<— SIP read from 62.4.147.22:5060 —>
ACK sip:8011234567@216.220.238.150 SIP/2.0
Via: SIP/2.0/UDP 62.4.147.22:5060;branch=z9hG4bK278fbad0;rport
From: “8015551212” sip:8015551212@62.4.147.22;tag=as2662575a
To: sip:3716267272@216.220.238.150;tag=as142fc13c
Contact: sip:8015551212@62.4.147.22
Call-ID: 7f3d3a0120100f794b239bf00e39dfac@62.4.147.22
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘7f3d3a0120100f794b239bf00e39dfac@62.4.147.22’ Method: ACK
Really destroying SIP dialog ‘028d145f49c027f42413216e71e92b03@216.220.238.150’ Method: REGISTER

*** UPDATE ***

If I add “allowguest=yes” in the general section I can now receive calls.

Isn’t “allowguest=yes” a bad thing? As I understand it, it allows anyone to use my server for both inbound and outbound calls, right?

BTW, if this is the wrong forum to ask about SVN code, just point me in the right direction.

*** UPDATE ***

Looks like a newer SVN snapshot fixes the issue.

I’d still like to get a definitive answer on the allowguest setting.