[Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN

[Steve]

OK trying this again armed with more info and examples I have still not
been able to make a successful outbound call.

I'm getting "Forbidden - wrong password on authentication for INVITE"

I KNOW 100$ for sure I am typing in the correct username and password.

The system still registers (same username and password above) and accepts
incoming calls from stanaphone.

I only get the error if I try an outbound (to the PSTN) call

For simpicity's sake (again) I made ONE context called sip and am only
working from there to attempt an outbound call.

Here is my sip.conf and extensions.conf [sip] section
and to follow that is the error line as seen in asterisk and some of the
sip debug output in case it helps.

I tried both user=
and

authuser =

and get the same result with either

Thanks!

Steve


-----------------------[ sip.conf ]--------------------

[general]
   port = 5060
    bindaddr = 0.0.0.0
     allow=ulaw

       externip = 68.X.X.X ;Outside address
       localnet = 10.73.73.133 ;Inside address
       localmask = 255.255.255.0 ;Inside subnet

          context = sip ; Default context for incoming calls
           register => secretnumber:**secret**@sip.stanaphone.com/77


[stanaphone-out]

type = friend
username = secretnumber
authuser = secretnumber
secret = **secret**
host = sip.stanaphone.com
nat = yes
canreinvite=no
insecure=very

;internal extensions are in my file but not shown here to save space
--------------------------[ sip.conf ]-----------------------------



--------------------------[ extensions.conf sip context ]---------------

[sip]

        ; include => default




exten =>_1XXXXXXXXXX,1,Dial(SIP/${EXTEN}@stanaphone-out)




exten => 78,1,Dial(SIP/78,20)
exten => 77,1,Dial(SIP/77,20)

--------------------------[ extensions.conf sip context ]---------------



Any further pointers would be greatly appreciated....


I have tried only user=

and only authuser=

as well

Get the same error either way.....




Here's some sip debug output when a stanaphone outgoing call is attempted:


Jun  4 18:19:52 WARNING[8422]: chan_sip.c:6829 handle_response: Forbidden 
- wrong password on authentication for INVITE to '"Steve 5.8Ghz Cordless" 
<sip:77 at PUBLIC_IP_HIDDEN>;tag=as3c1bd32f'
Reliably Transmitting:
CANCEL sip:15168372973 at sip.stanaphone.com SIP/2.0
Via: SIP/2.0/UDP PUBLIC_IP_HIDDEN:5060;branch=z9hG4bK5b7ba68e;rport
From: "Steve 5.8Ghz Cordless" <sip:77 at PUBLIC_IP_HIDDEN>;tag=as3c1bd32f
To: <sip:15168372973 at sip.stanaphone.com>
Contact: <sip:77 at PUBLIC_IP_HIDDEN>
Call-ID: 70d9ff1f223f93045a64edfd5d790beb at PUBLIC_IP_HIDDEN
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="STANAPHONE_NUMBER", 
realm="sip.stanaphone.com", algorithm=MD5, 
uri="sip:15168372973 at sip.stanaphone.com", 
nonce="42a22a1e3a06c8ca75eebf1325f5fa5dbd36d1a5", 
response="cd2e9744557c5720313017d6e4220737", opaque=""
Content-Length: 0

  (NAT) to 204.147.183.18:5060
Scheduling destruction of call 
'70d9ff1f223f93045a64edfd5d790beb at PUBLIC_IP_HIDDEN' in 15000 ms


Sip read:
SIP/2.0 481 Call Leg Does Not Exist
Via: SIP/2.0/UDP PUBLIC_IP_HIDDEN:5060;branch=z9hG4bK5b7ba68e
From: "Steve 5.8Ghz Cordless" <sip:77 at PUBLIC_IP_HIDDEN>;tag=as3c1bd32f
To: <sip:15168372973 at sip.stanaphone.com>;tag=as666beff7
Call-ID: 70d9ff1f223f93045a64edfd5d790beb at PUBLIC_IP_HIDDEN
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


10 headers, 0 lines
Destroying call '70d9ff1f223f93045a64edfd5d790beb at PUBLIC_IP_HIDDEN'


Sip read:


0 headers, 0 lines


Sip read:
CANCEL sip:15168372973 at 10.73.73.133 SIP/2.0
Via: SIP/2.0/UDP 10.73.73.111:5061;branch=z9hG4bK-d1780320
From: Steve 5.8Ghz Cordless <sip:77 at 10.73.73.133>;tag=9dc5f7ac1021ed2co1
To: <sip:15168372973 at 10.73.73.133>
Call-ID: bb4ca9ac-6ffae32c at 10.73.73.111
CSeq: 102 CANCEL
Max-Forwards: 70
Proxy-Authorization: Digest 
username="77",realm="asterisk",nonce="78df2dec",uri="sip:15168372973 at 10.73.73.133",algorithm=MD5,response="c58fbcfc34fe4535733650ca24045e47"
User-Agent: Linksys/PAP2-2.0.12(LS)
Content-Length: 0


10 headers, 0 lines
Sending to 10.73.73.111 : 5061 (NAT)
Reliably Transmitting (NAT):
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 
10.73.73.111:5061;branch=z9hG4bK-d1780320;received=10.73.73.111;rport=5061
From: Steve 5.8Ghz Cordless <sip:77 at 10.73.73.133>;tag=9dc5f7ac1021ed2co1
To: <sip:15168372973 at 10.73.73.133>;tag=as0688b73d
Call-ID: bb4ca9ac-6ffae32c at 10.73.73.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:15168372973 at PUBLIC_IP_HIDDEN>
Content-Length: 0


  to 10.73.73.111:5061
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
10.73.73.111:5061;branch=z9hG4bK-d1780320;received=10.73.73.111;rport=5061
From: Steve 5.8Ghz Cordless <sip:77 at 10.73.73.133>;tag=9dc5f7ac1021ed2co1
To: <sip:15168372973 at 10.73.73.133>;tag=as0688b73d
Call-ID: bb4ca9ac-6ffae32c at 10.73.73.111
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:15168372973 at PUBLIC_IP_HIDDEN>
Content-Length: 0


  to 10.73.73.111:5061


Sip read:
ACK sip:15168372973 at 10.73.73.133 SIP/2.0
Via: SIP/2.0/UDP 10.73.73.111:5061;branch=z9hG4bK-d1780320
From: Steve 5.8Ghz Cordless <sip:77 at 10.73.73.133>;tag=9dc5f7ac1021ed2co1
To: <sip:15168372973 at 10.73.73.133>;tag=as0688b73d
Call-ID: bb4ca9ac-6ffae32c at 10.73.73.111
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest 
username="77",realm="asterisk",nonce="78df2dec",uri="sip:15168372973 at 10.73.73.133",algorithm=MD5,response="1a015a2cddf9bbd76024aa37a7766816"
Contact: Steve 5.8Ghz Cordless <sip:77 at 10.73.73.111:5061>
User-Agent: Linksys/PAP2-2.0.12(LS)
Content-Length: 0


11 headers, 0 lines
Destroying call 'bb4ca9ac-6ffae32c at 10.73.73.111'


Sip read:


0 headers, 0 lines


Sip read:


0 headers, 0 lines