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Summary:ASTERISK-10656: hint does now work with the calling SIP channel
Reporter:A.R. Nasir Qureshi (nasirq)Labels:
Date Opened:2007-10-31 09:20:59Date Closed:2011-06-07 14:03:04
Priority:MajorRegression?No
Status:Closed/CompleteComponents:
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) Bug_11128_Log.txt
Description:I have tried with Polycom 650 and Cisco ATA 186.

When ever a SIP device makes the call, its state remains Idle.

The hint state only works when the SIP device is called.

****** STEPS TO REPRODUCE ******

1. Register any sip device
2. Make a hint extension like "exten => 4302,hint,SIP/138"
3. Make a call from the registered sip device
4. Use "core show hints" to see the the device State
Comments:By: Joshua C. Colp (jcolp) 2007-10-31 09:24:48

This is a common configuration issue. Make sure that a call-limit is set on the sip.conf entries in question or in general, and also make limitonpeersonly is set to yes.
By: A.R. Nasir Qureshi (nasirq) 2007-10-31 09:44:39

I have tried it again, both with Polycom and Cisco ATA 186.

My sip.conf is:

[138] ; Polycom
type=friend
username=138
password=
host=dynamic
dtmfmode=inband
mailbox=138
progressinband=no
call-limit=8
callerid=302
mailbox=302
limitonpeers = yes

[7777] ; Cisco
type=friend
secret=
host=dynamic
mailbox=7777
username=7777
qualify=yes
context = extensions
nat = 1
canreinvite=no
callerid="A.R. Nasir Qureshi" <302>
callgroup = 1
pickupgroup = 1
call-limit = 1
limitonpeers = yes


But still the state remain idle when the channel makes the call.
By: Joshua C. Colp (jcolp) 2007-10-31 09:48:47

Ah, the option is limitonpeer - not limitonpeers. Change that and try.
By: A.R. Nasir Qureshi (nasirq) 2007-10-31 10:03:18

Still not working.

[138]
type=friend
username=138
password=
host=dynamic
dtmfmode=inband
mailbox=138
progressinband=no
call-limit=8
callerid=302
mailbox=302
limitonpeer = yes

[7777]
type=friend
secret=
host=dynamic
mailbox=7777
username=7777
qualify=yes
context = extensions
nat = 1
canreinvite=no
callerid="A.R. Nasir Qureshi" <302>
callgroup = 1
pickupgroup = 1
call-limit = 1
limitonpeer = yes
By: Joshua C. Colp (jcolp) 2007-10-31 10:08:32

Then we will need complete console output with debug enabled (debug in logger.conf plus core set debug 9) of a call that should modify the call count. As well as sip show subscriptions and core show hints. Please attach them as attachments.
By: A.R. Nasir Qureshi (nasirq) 2007-10-31 10:38:55

File Uploaded
By: Sam Deller (samdell3) 2007-10-31 14:53:28

Hints only work correctly with:

type=peer
call-limit=1

Dont worry about type=peer. You can still make and recieve calls. I spent hours trying to work this problem out also.....
By: pj (pj) 2007-11-07 03:19:01.000-0600

sip hints work fine for me with this options:

type=friend
call-limit=2
busy-level=1

[general]
limitonpeer=yes
By: A.R. Nasir Qureshi (nasirq) 2007-11-12 02:28:08.000-0600

Using the following, fixed it, Thanks pj.

type=friend
call-limit=2
busy-level=1

[general]
limitonpeer=yes

This should be added in the wiki.

The bug can be closed.
By: Yuri (ys) 2007-11-12 03:11:32.000-0600

are busy-level not exist in 1.4 branch?
By: Joshua C. Colp (jcolp) 2007-11-12 07:33:27.000-0600

Correct, it does *not* exist in the 1.4 branch.
By: pj (pj) 2007-11-12 08:16:26.000-0600

'busy-level' exist in trunk, but probably not working as most people expect, i.e. to trigger busy tone, when eg. one call (busy-level=1) is on user phone (no matter if incomming or outgoing). probably because 'busy-level' isn't applied for both 'peer' or 'user' part of astrisk phone configuration.
bug ASTERISK-1112180
By: Jason Parker (jparker) 2007-12-12 15:48:14.000-0600

Closing, per reporter.