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Jan 6, 2009 2:36 AM by
gouriprasad 
>SIP Phone Error - 401 Unauthorized".
I assume you are using the SIP plugin for openfire to setup the SIP profile for the user and you that you have first confirmed that this works with Spark. The Admin console web page for the SIP user profile show show "registered" or whatever the status is.
>I also noted that my sip peers status on asterisk is Unreachable rather than ok
Another bug in MjSIP that has been identified. Set Qualify=no in Asterisk as a temp solution
>When I enter sparkweb, why should I fill in the server information??
There is thread in the Spark community right now about changes that you want to see. I suggest you register this suggestion there.
>I wish all communication would go through standard http, even voice... is it possible?
Flash by default, makes an RTMP connection on port 1935. All media (audio/video) is transmitted both ways over mulitple streams. Flash will also tunnel RTMP over HTTP. That is called RTMPT. Change your Red5 URL to rtmpt://your_server:8000/sip instead if rtmp:/sip. If port 8000 does do work for you look for the file plugins/red5/WEB-INF/classes/red5-rtmpt.xml and edit it accordingly. You can't use the HTTP-Bind port.
>exit sparkweb" does not work?
If you look at SparkWeb source code you will see "not implemented yet"
>but dtmf does not work
It is working in latest version. No audio feedback but it works for RFC2833 type DTMF which is the default for Asterisk. DTMF with INFO messages is not yet implemented. It works on my Asterisk and public phone numbers in the UK. I have not added this to SparkWeb yet.
>Thanks for your help and attention, please keep up this good work.
You are welcome. 
It is however a challenge to juggle all this with a full time day job.
I assume you are using the SIP plugin for openfire to setup the SIP profile for the user and you that you have first confirmed that this works with Spark. The Admin console web page for the SIP user profile show show "registered" or whatever the status is.
Indeed I am. Spark used to work fine with these settings. I am still having some trouble with remote connection, where voice is not passing through, but that is probably a port forwarding problem and I shouldn't bother you with that. The thing is, all my communication is working quite well locally, but red5 is still not connecting. As I got a "registered" when using spark, when I try with red5/sparkweb/ I get a "registration failed". When I use red5 only (the test page), I manage to have the phone working.
What I understood from you is that the test page uses the parameters set in the flash form : (here's where the image would be, in case I screw the formatting)
cid:5061B13F-7087-4D60-B16C-C3435AF1CDEE
On the other hand, the /red5/sparkweb/ section uses the parameters set both on the red5 (the red5 URL being that initially set with rtmp:/somewierdaddress which I have changed to rtmpt://220.110.24.198:8000/sip) admin section and the sip phone section. That's where I get confused, because if I'm right and since my sip phone plugin works correctly with Spark, I can't understand why sparkweb is not working at all. I'm not a pro with asterisk, but I debugged sip on the Asterisk CLI and here's what I got :
<--- SIP read from 220.110.24.198:5064 --->
REGISTER sip:220.110.24.198 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5064;rport;branch=z9hG4bK33285
Max-Forwards: 70
To: "3010" <sip:3010@220.110.24.198>
From: "3010" <sip:3010@220.110.24.198>;tag=z9hG4bK96257610
Call-ID: 582777981678@127.0.0.1<mailto:582777981678@127.0.0.1>
CSeq: 1 REGISTER
Contact: <sip:3010@127.0.0.1:5064>
Expires: 3600
User-Agent: mjsip stack 1.6
Content-Length: 0
<----
>
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 220.110.24.198 : 5064 (NAT)
<--- Transmitting (NAT) to 220.110.24.198:5064 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5064;branch=z9hG4bK33285;received=220.110.24.198;rport=5064
From: "3010" <sip:3010@220.110.24.198>;tag=z9hG4bK96257610
To: "3010" <sip:3010@220.110.24.198>
Call-ID: 582777981678@127.0.0.1
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:3010@220.110.24.198>
Content-Length: 0
<----
>
<--- Transmitting (NAT) to 220.110.24.198:5064 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.0.1:5064;branch=z9hG4bK33285;received=220.110.24.198;rport=5064
From: "3010" <sip:3010@220.110.24.198>;tag=z9hG4bK96257610
To: "3010" <sip:3010@220.110.24.198>;tag=as6a68790d
Call-ID: 582777981678@127.0.0.1
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4fb442d8"
Content-Length: 0
<----
>
Scheduling destruction of SIP dialog '582777981678@127.0.0.1' in 32000 ms (Method: REGISTER)
As you can see 220.110.24.198 is my pbx internal IP and since I'm running my openfire server in the same machine, it sometimes uses 127.0.0.1 and sometimes that IP. Another thing I noticed is that my Asterisk server first gives a Unauthorized answer to all SIP peers (for exemple those connecting from x-lite, I never knew why) and a 200OK response on the second attempt, which the mjsip stack doesn't make. Last, but not least, I've changed my red5 "enter" and "end" SIP and RTP ports into 5060-5099, 10000-20000 respectively, to compile with my company's defaults... I now think it was a stupid move, since all of those connections (I think) are going to be made from localhost only, so I shouldn't bother with opening those ports to the WAN (is that reasoning right?).
Well, I still got loads of doubts and I'm really sorry to waste your time with my newbie questions, specially knowing you have a hard schedule. If on the other hand you manage to take a look at my post, I would be really grateful.
Thanks in advance.
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SafariScreenSnapz 003.jpg (27.0 K)
well an update since my last post on this issue
Well it appears that Gizmo SIP settings for Registering SIP in Spark is now successful which all of you may know already.
Here is the settings I used:
Gizmo Settings:
server: proxy01.sipphone.com
port: 5060
**described setting on gizmo site: proxy01.sipphone.com:5060
stun: stun01.sipphone.com
stun port: 3478
**described setting on gizmo site: stun01.sipphone.com:3478
SIP Phone Mapping:
SIP Username: {YOUR GIZMO USER NAME}
Authorization Name: {YOUR GIZMO USER NAME}
Display Phone Number: {YOUR GIZMO SIP PHONE NUMBER}
Outbound Proxy: {LEAVE THIS BLANK}
When you add these Settings TEST DOES NOT WORK, however this will register when you start spark with SIP plugin enabled. Don't be discouraged by the failed TEST. Launch Spark and this should work, as always make sure you have the appropriate ports open on your firewall.
Problems are:
What does prompt for user credentials do?
Is this only for Asterisk SIP?
DELE:***ACCOUNT DOES NOT REGISTER IN SPARK WEB ~~~ DELE IF YOUR READING THIS WHAT DO YOU SUGGEST ON THIS PROBLEM?
Hope this helps anyone, since this has been a thorn in my side for a while.
here is a snapshot of my openfire startup
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opstartup.gif (28.4 K)
