I’ve googled and googled and I’m stumped. I am a newbie to asterisk running v. 1.0.9 on host 192.168.2.144, and have SIP softphone sipXphone running on host 192.168.2.101, both hosts on same subnet with no firewall between. sipXphone is set up as ext 200 with secret ‘blah’, and to point at the asterisk server. When dialing an asterisk extension the phone gives message [quote] Your call could not be completed. SIP code: Proxy Authentication Required (407). JTAPI code: Network not allowed (1001)[/quote]
The asterisk console outputs the following while I am attempting the call:[code]asterisk1*CLI>
Sip read:
INVITE sip:*43@192.168.2.144 SIP/2.0
From: sip:200@192.168.2.144;tag=1c13713
To: sip:*43@192.168.2.144
Call-Id: call-1130773851-3@192.168.2.101
Cseq: 1 INVITE
Contact: sip:200@192.168.2.101:5060
Content-Type: application/sdp
Content-Length: 196
Date: Mon, 31 Oct 2005 15:51:01 GMT
Max-Forwards: 20
User-Agent: sipX/2.5.2 (WinNT)
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE
Supported: sip-cc, sip-cc-01, timer, replaces
Via: SIP/2.0/UDP 192.168.2.101;branch=z9hG4bK-762846f324560aa928bb3b831b637668;rport
v=0
o=sipX 5 5 IN IP4 192.168.2.101
s=phone-call
c=IN IP4 192.168.2.101
t=0 0
m=audio 8766 RTP/AVP 0 8 96
a=rtpmap:0 pcmu/8000/1
a=rtpmap:8 pcma/8000/1
a=rtpmap:96 telephone-event/8000/1
15 headers, 9 lines
Using latest request as basis request
Sending to 192.168.2.101 : 5060 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.101;branch=z9hG4bK-762846f324560aa928bb3b831b637668
From: sip:200@192.168.2.144;tag=1c13713
To: sip:*43@192.168.2.144;tag=as7c17f5f4
Call-ID: call-1130773851-3@192.168.2.101
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:*43@192.168.2.144
Proxy-Authenticate: Digest realm=“asterisk”, nonce="036a8067"
Content-Length: 0
to 192.168.2.101:5060
Scheduling destruction of call ‘call-1130773851-3@192.168.2.101’ in 15000 ms
Found user '200’
asterisk1*CLI>
Sip read:
ACK sip:*43@192.168.2.144 SIP/2.0
Contact: sip:200@192.168.2.101:5060
From: sip:200@192.168.2.144;tag=1c13713
To: sip:*43@192.168.2.144;tag=as7c17f5f4
Call-Id: call-1130773851-3@192.168.2.101
Cseq: 1 ACK
Max-Forwards: 20
Via: SIP/2.0/UDP 192.168.2.101;branch=z9hG4bK-762846f324560aa928bb3b831b637668;rport
Content-Length: 0
9 headers, 0 lines
– Registered to ‘65.39.205.121’, who sees us as 207.154.100.17:4569
Destroying call ‘call-1130773851-3@192.168.2.101’
[/code]
My sip.conf file looks like:
[code]; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn’t, try adding “nat=1” to each peer definition to
; solve translation problems.
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
[/code]
and my sip_additional.conf looks like:
[code][200]
username=200
type=friend
secret=blah
record_out=On-Demand
record_in=On-Demand
qualify=no
port=5060
nat=never
mailbox=200@default
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=“Christopher Wrather” <200>
[201]
username=201
type=friend
secret=blah
record_out=On-Demand
record_in=On-Demand
qualify=no
port=5060
nat=never
mailbox=201@default
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=“Christopher Wrather” <201>
[/code]
The context [from-internal] looks like
[from-internal]
;allow phones to use applications
include => app-directory
include => app-dnd
include => app-callforward
include => app-callwaiting
include => app-messagecenter
include => app-calltrace
include => parkedcalls
include => from-internal-custom
;allow phones to dial other extensions
include => ext-fax
include => ext-local
include => ext-group
include => ext-queues
include => ext-zapbarge
include => ext-meetme
include => ext-record
include => ext-test
;allow phones to access trunks
include => outbound-allroutes
exten => s,1,Macro(hangupcall)
exten => h,1,Macro(hangupcall)
I would really appreciate any help you can give.