Displaying 3 results from an estimated 3 matches for "c1431100".
2007 Oct 26
1
Still more auth problems
Firstly can I ask when the documentation site will be online again? I'm
struggling here without it.
Further to my recent post I have tried to simplify things a little.
I have used a VoiceXML app to simple call an asterisk extension. EG:
<form id="transfer">
<block>
<call name="xfer" dest="sip:101 at 10.0.4.147:5060"/>
2007 Oct 30
0
Bad Request
..., port 5060
Reliably Transmitting (no NAT) to 10.0.2.136:5060:
BYE sip:vgp at 10.0.2.136:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.4.147:5060;branch=z9hG4bK4c02bc4f;rport
From: "Paul" <sip:paul at 10.0.4.147>;tag=as1ed7b694
To:
<sip:paulxfertest at middlesborough.kainos.com>;tag=C1431100-8D08-9097-2513
-8F10A9A44C48
Call-ID: 5b3c1b093ca08aa83b9a99cc128033c9 at 10.0.4.147
CSeq: 299 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of SIP dialog
'5b3c1b093ca08aa83b9a99cc128033c9 at 10.0.4.147' in 32000 ms (Method: BYE)
vm...
2007 Oct 25
0
Transfer and 407's
...nsfer 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: Dig...