I have a SIP voice server which I want to place an Asterisk server in
front of to handle call routing.
At the moment I can call the apps on the server fine, but it cannot
transfer to another extension via asterisk.
Even attempting a call from the voice server to an asterisk extension
results in a "407 Proxy Authorisation Required" error on Asterisk and
a
bridge_reject "noautho"
The server is registering fine with Asterisk.
Some debug stuff:
============================
sip.conf
[general]
context=internal
srvlookup=yes
[paul]
type=friend
secret=removed
qualify=no
nat=no
host=dynamic
canreinvite=yes
context=internal
[test]
type=friend
secret=removed
qualify=no
nat=no
host=dynamic
canreinvite=yes
context=internal
[vgp]
type=friend
secret=removed
qualify=no
nat=no
host=10.0.2.136
canreinvite=yes
context=internal
insecure=very
permit=10.0.2.136
extensions.conf
[internal]
exten => 101,1,Dial(SIP/paul)
exten => 103,1,Dial(SIP/test)
exten => 202,1,Dial(SIP/paulxfertest at middlesborough
<mailto:SIP/paulxfertest at middlesborough.kainos.com> )
SIP/paulxfertest at middlesborough
<mailto:SIP/paulxfertest at middlesborough.kainos.com> is a voice app
which simply attempts to transfer the call to: "103 at
asterisk-server".
SIP Error:
<--- Reliably Transmitting (no NAT) to 10.0.2.136:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.0.2.136:5060;branch=z9hG4bKb675eb1820a7c8;received=10.0.2.136
From: sip:paul at 10.0.4.147;tag=C1431100-8D08-7D26-6246-3AB1F318B851
To: <sip:103 at 10.0.4.147>;tag=as4c65adb2
Call-ID: C1431100-8D08-A124-575D-680811759D87-5060 at 10.0.2.136
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="6e23a069"
Content-Length: 0
I'd be very grateful for any suggestions.
Thanks,
Paul
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