Summary:ASTERISK-05801: AgentLogin/AgentCallbackLogin not work with SIP-channel
Reporter:Denis Smirnov (mithraen)Labels:
Date Opened:2005-12-08 06:22:23.000-0600Date Closed:2006-01-05 15:45:52.000-0600
Versions:Frequency of
Environment:Attachments:( 0) full
Description:AgentLogin/AgentCallbackLogin not work with SIP-channel

SCCP works well

When I dial to extension with AgentLogin asterisk hangup. Other applications works well.

Comments:By: Kevin P. Fleming (kpfleming) 2005-12-12 22:37:28.000-0600

The SIP phone clearly sent a BYE to Asterisk; what else do you expect us to do when the phone tells us to hang up?

By: Denis Smirnov (mithraen) 2005-12-13 02:07:37.000-0600

_Why_ it happend?

It happend only when using AgentLogin/AgentCallbackLogin.

- Echo/Miliwatt test works right;
- Calls from this phone to other SIP devices works fine;
- Calls from this phone to other asterisk server with IAX2 works fine;
- Calls to Queue application from this phone works fine;

All works fine, but with AgentLogin/AgentCallbackLogin in does not work.

It is happend only with last 1.2 snapshot, old spanshot from November also works fine with AgnetLogin/AgentCallbackLogin.

By: Kevin P. Fleming (kpfleming) 2005-12-13 10:10:15.000-0600

OK, I misread your log... that log does not even show a call being placed to an agent phone at all, it shows an incoming call to Asterisk.

If you want help in debugging this, please get a proper _complete_ trace showing the problem occurring.

By: Denis Smirnov (mithraen) 2005-12-13 10:24:43.000-0600

There is no call _to_ agent phone.

I call to number, AFAIR, '7', in extensions.conf:

exten => 7,1,AgentCallbackLogin

When I call from agent SIP device to '7' I see that phone hangup.

By: Kevin P. Fleming (kpfleming) 2005-12-13 10:33:36.000-0600

Your log trace shows the call going to AgentLogin, not AgentCallbackLogin.

By: Denis Smirnov (mithraen) 2005-12-13 11:19:07.000-0600


This bug identically reproduced with AgentLogin and AgentCallbackLogin.
I have forgotten which result from testings have sent here.

By: Kevin P. Fleming (kpfleming) 2005-12-13 11:27:09.000-0600

But it's still the same issue... the phone sends INVITE, we send OK (after the initial steps), the phone sends ACK and the call is up and running. Then the phone sends BYE, and the call is hung up. There is no other activity in the log between the ACK and the BYE. Asterisk is not causing this call to be hung up.

By: Denis Smirnov (mithraen) 2005-12-13 11:30:00.000-0600

Hmm. May be some intersting info in RTP stream?
I don't hanup. Phone don't hangup when call other application.
How I can debug this?

By: Kevin P. Fleming (kpfleming) 2005-12-13 16:59:08.000-0600

I don't know how you can debug it. You'll have to find out why the phone is sending BYE to Asterisk. Certainly the behavior would _not_ be the same with AgentCallbackLogin, which does not keep the channel open to the phone, but instead just records where you want the calls sent and then hangs up (but that is what it is supposed to do).

By: Denis Smirnov (mithraen) 2005-12-14 04:20:30.000-0600

It BYE after I dial to extension with AgentLogin/AgentCallbackLogin, not after I enter any password e.t.c. I Dial '7' and imeediatly SIP-phone hangups.

Can it be in some incompatibility with RTP support on this phone and Asterisk?

By: Tilghman Lesher (tilghman) 2006-01-05 15:45:52.000-0600

It could be, but this is clearly a problem with your SIP phone, not with Asterisk.  Please take it up with the manufacturer of your SIP phone.