[Serusers] asterisk can not hangup .user Wildcard X100P

dev2003 dev2003 at mail.ustc.edu.cn
Fri Nov 5 01:53:22 CET 2004


Flynn

	
 modules being loaded properly.
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    -- Remote UNIX connection
    -- Starting simple switch on 'Zap/1-1'
Nov  5 09:06:43 NOTICE[1118463168]: chan_zap.c:5055 ss_thread: Got event 2 (Ring/Answered)...
Nov  5 09:06:45 NOTICE[1118463168]: chan_zap.c:5055 ss_thread: Got event 2 (Ring/Answered)...
Nov  5 09:06:46 NOTICE[1118463168]: chan_zap.c:5055 ss_thread: Got event 2 (Ring/Answered)...
    -- Executing Goto("Zap/1-1", "default|12345|1") in new stack
    -- Goto (default,12345,1)
    -- Executing Ringing("Zap/1-1", "") in new stack
    -- Executing Answer("Zap/1-1", "") in new stack
    -- Executing AGI("Zap/1-1", "xml.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/xml.agi
    -- Playing 'agent-pass' (language 'en')
    --  xml.agi: The Input data is #,queuename is group2
    --  xml.agi: User Input : #
    --  xml.agi: get in GetAgent group2
    --  xml.agi: no agent is avail
    --  xml.agi: no agent
    -- Started music on hold, class 'default', on Zap/1-1
    --  xml.agi: get in GetAgent group2
    --  xml.agi: no agent is avail
    --  xml.agi: get in GetAgent group2
    --  xml.agi: no agent is avail
    --  xml.agi: get in GetAgent group2
    --  xml.agi: no agent is avail
    --  xml.agi: get in GetAgent group2
    --  xml.agi: no agent is avail
........................................


    --soft hangup zap/1-1
Requested Hangup on channel 'Zap/1-1'
    -- Stopped music on hold on Zap/1-1
  == Spawn extension (default, 12345, 3) exited non-zero on 'Zap/1-1'
    -- Executing Hangup("Zap/1-1", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'


// I have found from google
it is common problem


http://www.marko.net/asterisk/archives/0208/0329.html

This has been covered in many other messages, but in order for the X100P 
to detect hangup, you must have disconnect supervision on the phone line. 
If the line you are providing is from another PBX, then it is highly 
unlikely that it supplies the disconnect supervision. Most telephone 
switches do support disconnect supervision, but it's not always on by 
default. You can tell by using a lighted keypad which receives its power 
only from the phone line, and then calling it and hanging up on it. If 
the lighted keypad blinks off for a moment then your line has the 
disconnect supervision, otherwise it doesn't.


http://www.marko.net/asterisk/archives/0206/0198.html

"ks" is the right one. Have you confirmed that your line supports 
disconnect supervision? Here is an easy test: 


Get a phone that has a lighted keypad, where the lighted keypad is powered 
ONLY by the phone line's power. Call the phone and then hang up on it. 
If the lighted keypad goes out for a brief time, then you do have 
disconnect supervision and we have to figure out how to tune the driver to 
see it.



>why don't you try using the stable version of asterisk? right now i'm
>using version 1.0RC2 (which is the version just before the actual
>1.0.0) and it's been fantastic.
>
>What you're describing could be a whole bunch of things -- IRQ sharing
>problems, faulty configuration files, etc. Why don't you attach a bit
>more information, perhaps the zapata.conf file.
>
>are the appropriate modules being loaded properly? Try watching the
>output of "dmesg" to see if there are any problems with the digium
>card.
>
>Flynn
>
>
>On Thu, 4 Nov 2004 10:08:34 +0800, dev2003 <dev2003 at mail.ustc.edu.cn> wrote:
>> Flynn,
>> 
>>               this  sort of behaviour happenning   all the time.
>>            such as  when call 11,then I hangup.
>>            but when I recall 11,then it is busy.
>>            os redhat 9.
>> /usr/sbin/asterisk -r
>> Asterisk CVS-HEAD-09/10/04-21:34:12, Copyright (C) 1999-2004 Digium.
>> Written by Mark Spencer <markster at digium.com>
>> 
>> zaptel-0.9.0
>> 
>> 
>> 
>> 
>> >can you give more details about your asterisk settings? when is this
>> >sort of behaviour happenning -- all the time? or only when a specific
>> >condition occurs?
>> >
>> >flynn
>> >
>> >p/s including your config files wouldn't be a bad idea as well
>> >
>> >
>> >On Wed, 3 Nov 2004 18:33:28 +0800, dev2003 <dev2003 at mail.ustc.edu.cn> wrote:
>> >> serusers,您好!
>> >>
>> >>                asterisk can not hangup .user Wildcard X100P.
>> >>            when using phone call,asterisk can not hangup.
>> >>           I should tpye:
>> >>                   soft hangup zap/1-1
>> >>                then can hangup.
>> >>
>> >> dev2003
>> >> dev2003 at mail.ustc.edu.cn
>> >> 2004-11-03
>> >>
>> >> _______________________________________________
>> >> Serusers mailing list
>> >> Serusers at iptel.org
>> >> http://mail.iptel.org/mailman/listinfo/serusers
>> >>
>> >>
>> >>
>> 
>> = = = = = = = = = = = = = = = = = = = =
>> 
>>>> 礼!
>> 
>> 
>> dev2003
>> dev2003 at mail.ustc.edu.cn
>> 2004-11-04
>> 
>> 
>>

= = = = = = = = = = = = = = = = = = = =
			

        致
礼!
 
				 
        dev2003
        dev2003 at mail.ustc.edu.cn
          2004-11-05

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