Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-23 Thread Holger Schurig
 And I would very much like feedback on those - are they useful?
 If they are, I'll backport to chan_sip.

I find them useful (for the HTML based block LED field display of DeStar). 
Today I even thought about writing a patch that sends the 
available/unavailable messages from the quality=... code to the Manager.

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Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-23 Thread Holger Schurig
 And I would very much like feedback on those - are they useful?

Oh, I just found out by looking at the source code that there are database 
entries SIP/Registry. I think the used database entries is something that 
is currently under-documented ...  :-)

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Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-23 Thread John Todd
At 9:41 AM +0200 on 7/23/04, Holger Schurig wrote:
  And I would very much like feedback on those - are they useful?
 If they are, I'll backport to chan_sip.
I find them useful (for the HTML based block LED field display of DeStar).
Today I even thought about writing a patch that sends the
available/unavailable messages from the quality=... code to the Manager.
Would it be just as useful to actually have the numeric values 
readable to the Manager, as well as the binary 
(available/unavailable) status of results from quality= statements?

You could (theoretically) color code an icon through some type of 
good-to-bad spectrum depending on how fast it was responding to the 
quality queries.

More eye candy for manager-types.
JT
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Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-23 Thread Philipp von Klitzing
Holger Schurig wrote:
And I would very much like feedback on those - are they useful?
Oh, I just found out by looking at the source code that there are database 
entries SIP/Registry. I think the used database entries is something that 
is currently under-documented ...  :-)
If you simply type database show (and nothing more) on the CLI you'll 
discover even more if you have IAX clients registered... :-)

Cheers, Philipp
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Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-23 Thread Holger Schurig
The patch tries to send the time as well, but it fails. There are some 
problems currently:

I applied the path from bug 2117. After this I got some events:

Event: SIPPeerStatus
Peer: weckhardt
Status: reachable
Time: 55

Event: SIPPeerRegistration
Peername: dnarotam
Status: Offline



I think we have several problems here. Once it's Peer:, the other time 
it's Peername. Also, I don't like the name of the event. It should just 
be PeerStatus and PeerRegistration, because we might add something to 
IAX2 as well. So I'd suggest to do it this way:

Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: reachable
Time: 55

Event: PeerRegistration
Peer: SIP/dnarotam
Channel: Offline



This way, other channels can send the same events, just with PeerStatus: 
IAX2/qtiax.

I guess I'll redo the path and re-submit it to bugs.digium.com.

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Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-23 Thread Holger Schurig
Okay, I have finished my patch. With qualify=yes in sip.conf it looks 
like this:

-
# telnet 127.0.0.1 5038
Trying 127.0.0.1...
Connected to localhost.localdomain (127.0.0.1).
Escape character is '^]'.
Asterisk Call Manager/1.0
Action: Login
Username: destar
Secret: secret

Response: Success
Message: Authentication accepted

Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Registered

Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Reachable
Time: 81

Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Registered

Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Unreachable
Time: -1

Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Unregistered
Cause: Expired
-

Without the quality, you still get the PeerStatus: Registered and 
PeerStatus: Unegistered events.


John, you can do your color-coding :-)




And while I was at this patch, I also changed the

Event: SIPRegistry
Domain: ...
Status: ...

to

Event: Register
Channel: SIP
Domain: ...
Status: ...

I don't have a register= line in my sip.conf, so I didn't see this event. 
But again I hope that this makes the event usable for other channel types 
as well.

Because I did not see the event, I am not sure if I need the Channel: 
SIP at all. If the Domain is actually an sip:// - URL, it would be 
superfluous.

I'll post the patch to bugs.digium.com when someone enlightens me on this 
point :-)

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Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-23 Thread Olle E. Johansson
Holger Schurig wrote:
The patch tries to send the time as well, but it fails. There are some 
problems currently:
Seems like you are getting Time - how does it fail? Please explain...
/O
I applied the path from bug 2117. After this I got some events:
Event: SIPPeerStatus
Peer: weckhardt
Status: reachable
Time: 55
Event: SIPPeerRegistration
Peername: dnarotam
Status: Offline

I think we have several problems here. Once it's Peer:, the other time 
it's Peername. Also, I don't like the name of the event. It should just 
be PeerStatus and PeerRegistration, because we might add something to 
IAX2 as well. So I'd suggest to do it this way:

Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: reachable
Time: 55
Event: PeerRegistration
Peer: SIP/dnarotam
Channel: Offline

This way, other channels can send the same events, just with PeerStatus: 
IAX2/qtiax.

I guess I'll redo the path and re-submit it to bugs.digium.com.
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--
Olle E. Johansson, Edvina.net AB, [EMAIL PROTECTED]
- Phone +46 8 594 788 10, Cell phone: +46 70 593 68 51
- IP phone: sip:[EMAIL PROTECTED]
- Address: Runbovägen 10, SE-192 48 Sollentuna, Sweden
- Web: http://edvina.net
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Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-23 Thread Olle E. Johansson
Holger Schurig wrote:

I think we have several problems here. Once it's Peer:, the other time 
it's Peername. 
That's clearly a bug.
Also, I don't like the name of the event. It should just 
be PeerStatus and PeerRegistration, because we might add something to 
IAX2 as well. So I'd suggest to do it this way:
I used SIP as a prefix for testing in chan_sip2. But if it's possible to add
similar events in IAX, I agree.
Good to get feedback that it was a useful addition :-)
/O
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Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-23 Thread Olle E. Johansson
Holger Schurig wrote:

And while I was at this patch, I also changed the
Event: SIPRegistry
Domain: ...
Status: ...
to
Event: Register
Channel: SIP
Domain: ...
Status: ...
I still believe it would be better to call this Registry since that's a common
term across IAX and SIP for outbound registrations. The event reflects changes
in our registry, like SIP show registry and Iax2 show registry.
Because I did not see the event, I am not sure if I need the Channel: 
SIP at all. If the Domain is actually an sip:// - URL, it would be 
superfluous.
Domain is just the domain we register to. It might be a good thing
to add the host IP we resolve this to, since that may change.
Channel: makes sense, to separate registries.
/O
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Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-23 Thread John Todd
At 10:58 AM +0200 on 7/23/04, Holger Schurig wrote:
Okay, I have finished my patch. With qualify=yes in sip.conf it looks
like this:
-
# telnet 127.0.0.1 5038
Trying 127.0.0.1...
Connected to localhost.localdomain (127.0.0.1).
Escape character is '^]'.
Asterisk Call Manager/1.0
Action: Login
Username: destar
Secret: secret
Response: Success
Message: Authentication accepted
Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Registered
Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Reachable
Time: 81
Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Registered
Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Unreachable
Time: -1
Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Unregistered
Cause: Expired
-
Without the quality, you still get the PeerStatus: Registered and
PeerStatus: Unegistered events.
John, you can do your color-coding :-)
[snip]
Not me! :-)  I'd point a finger at Nicolás Gudiño 
and have him include it in the Asterisk Flash 
Operator panel, which seems to be one of the 
appropriate places that this could create a 
graphical representation of registration status 
and quality= response time.

Maybe a red-to-green spectrum of colors on the 
button background.  I'd expecet that each button 
would need to have probably independent 
configurations, since some devices may be very 
far away and thus have different numeric values 
mapped to different colors.  If the device falls 
out of registration, then perhaps have thin black 
lines diagonally through the button, and dim it 
slightly?

JT

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Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-23 Thread Nicolas Gudino
Hi John,
John Todd wrote:
At 10:58 AM +0200 on 7/23/04, Holger Schurig wrote:
Okay, I have finished my patch. With qualify=yes in sip.conf it looks
like this:
output snip
Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Reachable
Time: 81
some more snip
-
Without the quality, you still get the PeerStatus: Registered and
PeerStatus: Unegistered events.
John, you can do your color-coding :-)
[snip]
Not me! :-)  I'd point a finger at Nicolás Gudiño and have him include 
it in the Asterisk Flash Operator panel, which seems to be one of the 
appropriate places that this could create a graphical representation of 
registration status and quality= response time.

Maybe a red-to-green spectrum of colors on the button background.  I'd 
expecet that each button would need to have probably independent 
configurations, since some devices may be very far away and thus have 
different numeric values mapped to different colors.  If the device 
falls out of registration, then perhaps have thin black lines diagonally 
through the button, and dim it slightly?
Thats me... :) Well, we already have in the panel dimmed buttons for SIP 
peers that are unreachable, and really dimmed ones for not registered 
ones. Now I will have code the color shift based on the round trip time. 
Maybe I can zoom out the button instead of color coding? If the latency 
is high display the button far far away :)

--
Nicolás Gudiño
House Internet S.R.L.
Buenos Aires - Argentina
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RE: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-22 Thread Matthias Endler
  Nicolas Gudino wrote:
   On Fri, 2004-07-16 at 18:28, Matthias Endler wrote:
  is it possible to receive SIP/IAX register and unregister
 events via the
  manager API (like in CLI)? I do receive all kinds of call events
  (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).
  
  
   chan_sip2 supports manager notifications:
  
   http://bugs.digium.com/bug_view_page.php?bug_id=759
  And I would very much like feedback on those - are they useful?
  If they are, I'll backport to chan_sip.

 first, sorry to answer so late.

 I find them very useful, especially for my current project
 written in Java,
 which heavily makes use of the Asterisk Call Manager Interface. I
 started to
 backport to chan_sip.c already, it will be done by tomorrow. If you are
 interested I can send you a diff (stable version 0.9.1).

As promised yesterday:

Anybody interrested can download the patch for Asterisk 0.9.1 at
http://matthiasendler.net/asterisk/patch/.

Best regards
Matt

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Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-22 Thread Olle E. Johansson
Matthias Endler wrote:

As promised yesterday:
Anybody interrested can download the patch for Asterisk 0.9.1 at
http://matthiasendler.net/asterisk/patch/.
Great!
Please add it to the bugtracker in a .txt file created with
cvs diff -u channels/chan_sip.c
The diff has to be for CVS HEAD, that is 1.0rc1
/O
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RE: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-21 Thread Matthias Endler
Hi Olle,

 Nicolas Gudino wrote:
  On Fri, 2004-07-16 at 18:28, Matthias Endler wrote:
 is it possible to receive SIP/IAX register and unregister events via the
 manager API (like in CLI)? I do receive all kinds of call events
 (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).
 
 
  chan_sip2 supports manager notifications:
 
  http://bugs.digium.com/bug_view_page.php?bug_id=759
 And I would very much like feedback on those - are they useful?
 If they are, I'll backport to chan_sip.

first, sorry to answer so late.

I find them very useful, especially for my current project written in Java,
which heavily makes use of the Asterisk Call Manager Interface. I started to
backport to chan_sip.c already, it will be done by tomorrow. If you are
interested I can send you a diff (stable version 0.9.1).

Best regards
Matt

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Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-17 Thread Olle E. Johansson
Nicolas Gudino wrote:
On Fri, 2004-07-16 at 18:28, Matthias Endler wrote:
is it possible to receive SIP/IAX register and unregister events via the
manager API (like in CLI)? I do receive all kinds of call events
(Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).

chan_sip2 supports manager notifications:
http://bugs.digium.com/bug_view_page.php?bug_id=759
And I would very much like feedback on those - are they useful?
If they are, I'll backport to chan_sip.
/O
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[Asterisk-Users] SIP register and unregister events via Manager API

2004-07-16 Thread Matthias Endler
Hi all,

is it possible to receive SIP/IAX register and unregister events via the
manager API (like in CLI)? I do receive all kinds of call events
(Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).

My manager.conf looks like this:

[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0

[manage]
secret = mysecret
deny=0.0.0.0/0.0.0.0
permit=192.168.1.0/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

Thanks for any hints in advance.

Matt

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Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-16 Thread Nicolas Gudino
Hi Matthias,

On Fri, 2004-07-16 at 18:28, Matthias Endler wrote:
 Hi all,
 
 is it possible to receive SIP/IAX register and unregister events via the
 manager API (like in CLI)? I do receive all kinds of call events
 (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).

chan_sip2 supports manager notifications:

http://bugs.digium.com/bug_view_page.php?bug_id=759

Best regards,

-- 
Nicolas Gudino [EMAIL PROTECTED]
House Internet S.R.L.

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