[asterisk-users] Hangup not detected
Hi, I just realize in these few days there are many calls that already hangup but not detected by Asterisk. Those calls occupy PSTN lines and need to be manually terminated through Flash Operation Panel or phycally disconnect the PSTN lines. This never happen before but as long as I can remember, there are no change in configuration. Any ideas how to solve this? Thanks :-) Anam. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup not detected
On Tuesday 18 September 2012, Satria Anamarta wrote: Hi, I just realize in these few days there are many calls that already hangup but not detected by Asterisk. Those calls occupy PSTN lines and need to be manually terminated through Flash Operation Panel or phycally disconnect the PSTN lines. This never happen before but as long as I can remember, there are no change in configuration. Any ideas how to solve this? If you are using analogue phone lines in some country that uses a British- style telephone system (line wires called A and B, not tip and ring; polarity reversal before ringing; double ring on incoming call), then by design only the calling party can terminate a call once established. If someone rings you and you hang up but they stay on the line, you will still be connected to them if you later pick up the phone -- the call is only disconnected once the calling party hangs up. Asterisk is aware of this, and takes steps to mitigate it. The fix is simply to make sure you specify the correct country in your DAHDI configuration. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup not detected
Hi AJS, Thank you for your reply , I am using this in IRAN so please guide me what to do and and explain me more. Look forward to hearing from your side. Regards, Mehdi On Tue, Sep 18, 2012 at 11:28 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Tuesday 18 September 2012, Satria Anamarta wrote: Hi, I just realize in these few days there are many calls that already hangup but not detected by Asterisk. Those calls occupy PSTN lines and need to be manually terminated through Flash Operation Panel or phycally disconnect the PSTN lines. This never happen before but as long as I can remember, there are no change in configuration. Any ideas how to solve this? If you are using analogue phone lines in some country that uses a British- style telephone system (line wires called A and B, not tip and ring; polarity reversal before ringing; double ring on incoming call), then by design only the calling party can terminate a call once established. If someone rings you and you hang up but they stay on the line, you will still be connected to them if you later pick up the phone -- the call is only disconnected once the calling party hangs up. Asterisk is aware of this, and takes steps to mitigate it. The fix is simply to make sure you specify the correct country in your DAHDI configuration. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup not detected
Hello In indications.com are the tones for several countries On Sep 18, 2012 4:34 AM, Mehdi Rahimi mrm.ci...@gmail.com wrote: Hi AJS, Thank you for your reply , I am using this in IRAN so please guide me what to do and and explain me more. Look forward to hearing from your side. Regards, Mehdi On Tue, Sep 18, 2012 at 11:28 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Tuesday 18 September 2012, Satria Anamarta wrote: Hi, I just realize in these few days there are many calls that already hangup but not detected by Asterisk. Those calls occupy PSTN lines and need to be manually terminated through Flash Operation Panel or phycally disconnect the PSTN lines. This never happen before but as long as I can remember, there are no change in configuration. Any ideas how to solve this? If you are using analogue phone lines in some country that uses a British- style telephone system (line wires called A and B, not tip and ring; polarity reversal before ringing; double ring on incoming call), then by design only the calling party can terminate a call once established. If someone rings you and you hang up but they stay on the line, you will still be connected to them if you later pick up the phone -- the call is only disconnected once the calling party hangs up. Asterisk is aware of this, and takes steps to mitigate it. The fix is simply to make sure you specify the correct country in your DAHDI configuration. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup not detected
On Tuesday 18 September 2012, Mehdi Rahimi wrote: Hi AJS, Thank you for your reply , I am using this in IRAN so please guide me what to do and and explain me more. Look forward to hearing from your side. Regards, Mehdi Unfortunately I am not familiar with the Iranian telephone system. You might have to search for relevant technical standards documentation. For a start, try setting your location to UK -- and if it behaves a bit better, that will be your problem. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup not detected?
Swift() is an asterisk wrapper around the text-to-speech engine cepstral. Looks like this is a dev issue - I'll start a new thread on the dev mailing list. Justin Killen Senior Programmer / Analyst All American Asphalt 951-736-7600 x 2060 jkil...@allamericanasphalt.commailto:jkil...@allamericanasphalt.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada Sent: Thursday, May 24, 2012 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hangup not detected? Looks like Swift() (whatever that is) is not returning ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup not detected?
Here is the output from the cli: dozer*CLI core show channels Channel Location State Application(Data) DAHDI/5-1s@DB_LOOKUP:24 Up Swift(Schedule for employee 1 active channel 1 active call 1528 calls processed dozer*CLI core show channel dahdi/5-1 -- General -- Name: DAHDI/5-1 Type: DAHDI UniqueID: 1337821128.1363 LinkedID: 1337821128.1363 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) Rings: 1 NativeFormats: 0x4 (ulaw) WriteFormat: 0x4 (ulaw) ReadFormat: 0x4 (ulaw) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 15 Frames in: 3967 Frames out: 15882 Time to Hangup: 0 Elapsed Time: 20h56m23s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: DB_LOOKUP Extension: s Priority: 24 Call Group: 0 Pickup Group: 0 Application: Swift Data: Schedule for employee number : Thursday, May 24th, 2012, you are scheduled at XX Blocking in: (Not Blocking) Variables: READSTATUS=TIMEOUT return_id= MAX_REPEAT=4 ODBCSTATUS=SUCCESS ODBCROWS=1 COUNTER=2 AAA_OUTPUT=Schedule for employee number : Thursday, May 24th, 2012, you are scheduled at XX.. data=Thursday, May 24th, 2012, you are scheduled at XX id= ODBC_FETCH_STATUS=SUCCESS ~ODBCFIELDS~=id,data ODBC_ID=903 ID_VALIDATED=AAA_VALIDATE_EMP_NUM(27,) account_id= read_length=7 get_param2=E get_param1=27 validate_func=AAA_VALIDATE_EMP_NUM truck_text=employee number readprompt=AAA/enter_employee_number comp_num=27 BACKGROUNDSTATUS=SUCCESS CDR Variables: level 1: dnid= level 1: dst=4 level 1: dcontext=default level 1: channel=DAHDI/5-1 level 1: lastapp=Swift level 1: lastdata=Schedule for employee number : Thursday, May 24th, 2012, you are schedu level 1: start=2012-05-23 17:58:48 level 1: answer=2012-05-23 17:58:54 level 1: duration=75383 level 1: billsec=75377 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: accountcode=27_EMP level 1: uniqueid=1337821128.1363 level 1: linkedid=1337821128.1363 level 1: userfield=2885 level 1: sequence=1363 Since the 'lastapp' variable is 'Swift', this would indicate that the cepstral wrapper is having a problem, correct? Justin Killen From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Tuesday, May 22, 2012 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hangup not detected? Okay, the next time it gets in this state I'll gather that information. Justin Killen From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Monday, May 21, 2012 1:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hangup not detected? On Fri, May 18, 2012 at 12:00 PM, Justin Killen jkil...@allamericanasphalt.commailto:jkil...@allamericanasphalt.com wrote: I have and automated call-in dispatch system where hundreds of people call in daily for 2-3 minutes each. The extension is set up to get their information, then text-to-speech the dispatch information (via odbc). It then loops 5 times then ends the call. These calls are being handled by an 8 port analog digium card. Sometimes though, I see calls via 'core show channel dahdi/1-1' that have a time of 16 hours. I'm not sure if this is a result of dahdi missing the hangup, ODBC timing out, or TTS failing for some reason. When a channel gets in this state, the call doesn't seem to progress through the dialplan, they always display the TTS line. Doing a 'dahdi destroy channel 1-1' doesn't seem to be effective - the only way I've been able to clear the calls is to do a 'dahdi restart' and/or restart the asterisk service. For TTS I'm using cepstral with the Swift wrapper. Here is a snippet of my dialplan: Can you post the CLI output of a call that gets hung? I'd like to see where it's hanging on. Also, as a work-around to attempt to solve the symptom and not the underlying issue, you could maybe setup a cron job that runs once every ten minutes that checks for stale calls using AMI, and then hangs up any calls up that are over 10 minutes long? Using the AMI Hangup command? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.comhttp://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
Re: [asterisk-users] hangup not detected?
Looks like Swift() (whatever that is) is not returning ? On 24 May 2012 23:07, Justin Killen jkil...@allamericanasphalt.com wrote: ** ** ** Here is the output from the cli: ** ** dozer*CLI core show channels Channel Location State Application(Data) DAHDI/5-1s@DB_LOOKUP:24 Up Swift(Schedule for employee 1 active channel 1 active call 1528 calls processed dozer*CLI core show channel dahdi/5-1 -- General -- Name: DAHDI/5-1 Type: DAHDI UniqueID: 1337821128.1363 LinkedID: 1337821128.1363 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) Rings: 1 NativeFormats: 0x4 (ulaw) WriteFormat: 0x4 (ulaw) ReadFormat: 0x4 (ulaw) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 15 Frames in: 3967 Frames out: 15882 Time to Hangup: 0 Elapsed Time: 20h56m23s Direct Bridge: none Indirect** **Bridge: none -- PBX -- Context: DB_LOOKUP Extension: s Priority: 24 Call Group: 0 Pickup Group: 0 Application: Swift Data: Schedule for employee number : Thursday, May 24th, 2012, you are scheduled at XX Blocking in: (Not Blocking) Variables: READSTATUS=TIMEOUT return_id= MAX_REPEAT=4 ODBCSTATUS=SUCCESS ODBCROWS=1 COUNTER=2 AAA_OUTPUT=Schedule for employee number : Thursday, May 24th, 2012, you are scheduled at XX.. data=Thursday, May 24th, 2012, you are scheduled at XX id= ODBC_FETCH_STATUS=SUCCESS ~ODBCFIELDS~=id,data ODBC_ID=903 ID_VALIDATED=AAA_VALIDATE_EMP_NUM(27,) account_id= read_length=7 get_param2=E get_param1=27 validate_func=AAA_VALIDATE_EMP_NUM truck_text=employee number readprompt=AAA/enter_employee_number comp_num=27 BACKGROUNDSTATUS=SUCCESS ** ** CDR Variables: level 1: dnid= level 1: dst=4 level 1: dcontext=default level 1: channel=DAHDI/5-1 level 1: lastapp=Swift level 1: lastdata=Schedule for employee number : Thursday, May 24th, 2012, you are schedu level 1: start=2012-05-23 17:58:48 level 1: answer=2012-05-23 17:58:54 level 1: duration=75383 level 1: billsec=75377 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: accountcode=27_EMP level 1: uniqueid=1337821128.1363 level 1: linkedid=1337821128.1363 level 1: userfield=2885 level 1: sequence=1363 ** ** ** ** ** ** ** ** ** ** Since the ‘lastapp’ variable is ‘Swift’, this would indicate that the cepstral wrapper is having a problem, correct? ** ** Justin Killen -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Justin Killen *Sent:* Tuesday, May 22, 2012 8:53 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] hangup not detected? ** ** Okay, the next time it gets in this state I’ll gather that information.*** * ** ** Justin Killen -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren Selby *Sent:* Monday, May 21, 2012 1:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] hangup not detected? ** ** On Fri, May 18, 2012 at 12:00 PM, Justin Killen jkil...@allamericanasphalt.com wrote: I have and automated call-in dispatch system where hundreds of people call in daily for 2-3 minutes each. The extension is set up to get their information, then text-to-speech the dispatch information (via odbc). It then loops 5 times then ends the call. These calls are being handled by an 8 port analog digium card. Sometimes though, I see calls via ‘core show channel dahdi/1-1’ that have a time of 16 hours. I’m not sure if this is a result of dahdi missing the hangup, ODBC timing out, or TTS failing for some reason. When a channel gets in this state, the call doesn’t seem to progress through the dialplan, they always display the TTS line. Doing a ‘dahdi destroy channel 1-1’ doesn’t seem to be effective – the only way I’ve been able to clear the calls is to do a ‘dahdi restart’ and/or restart the asterisk service.* *** For TTS I’m using cepstral with the Swift wrapper. Here is a snippet of my
[asterisk-users] hangup not detected?
I have and automated call-in dispatch system where hundreds of people call in
daily for 2-3 minutes each. The extension is set up to get their information,
then text-to-speech the dispatch information (via odbc). It then loops 5 times
then ends the call. These calls are being handled by an 8 port analog digium
card.
Sometimes though, I see calls via 'core show channel dahdi/1-1' that have a
time of 16 hours. I'm not sure if this is a result of dahdi missing the
hangup, ODBC timing out, or TTS failing for some reason. When a channel gets
in this state, the call doesn't seem to progress through the dialplan, they
always display the TTS line. Doing a 'dahdi destroy channel 1-1' doesn't seem
to be effective - the only way I've been able to clear the calls is to do a
'dahdi restart' and/or restart the asterisk service.
For TTS I'm using cepstral with the Swift wrapper.
Here is a snippet of my dialplan:
[AAA_27_EMP]
exten = s,1,Answer
same = n,Set(CDR(accountcode)=27_EMP)
same = n,Set(comp_num=27)
same = n,Set(readprompt=AAA/enter_employee_number)
same = n,Set(truck_text=employee number)
same = n,Set(validate_func=AAA_VALIDATE_EMP_NUM)
same = n,Set(get_param1=27)
same = n,Set(get_param2=E)
same = n,Set(read_length=7)
same = n,Goto(DB_LOOKUP,s,1)
[DB_LOOKUP]
exten = s,1,NoOp()
same = n(getid),Read(account_id,${readprompt},${read_length},,3,5)
same = n,Gotoif($[ ${LEN(${account_id})} = 0]?timeout_hangup)
same = n(validateid),Verbose(validating id ${account_id})
same = n,Set(CDR(userfield)=${account_id})
same = n,GotoIf($[${account_id}==*]?AAACompMenu,s,1)
same =
n,Set(ID_VALIDATED=${validate_func}(${get_param1},${account_id}))
same = n,GotoIf($[${ID_VALIDATED}==0]?badid)
same = n(goodid),Verbose(getting schedule for id ${account_id}
AAA_GET_SCHEDULE(${get_param1},${get_param2},${account_id})))
same =
n,Set(ODBC_ID=${AAA_GET_SCHEDULE(${get_param1},${get_param2},${account_id})})
same = n,GotoIf($[${ODBCROWS} 1]?no_schedule)
same = n,Verbose(odbcrows count: ${ODBCROWS})
same = n,Set(COUNTER=1)
same = n,Set(AAA_OUTPUT=Schedule for ${truck_text} ${account_id}: )
same = n,While($[${COUNTER} = ${ODBCROWS}])
same = n,Set(ARRAY(id,data)=${ODBC_FETCH(${ODBC_ID})})
same = n,Set(AAA_OUTPUT=${AAA_OUTPUT}${data}. )
;same = n,Swift(${data})
same = n,Set(COUNTER=$[${COUNTER} + 1])
same = n,EndWhile()
same = n,ODBCFinish()
same = n,NoOp(${get_param2})
same = n,Set(AAA_CHECKED_IN()=${comp_num}, ${get_param2},
${account_id}, ${AAA_OUTPUT}, S, ${CALLERID(num)}, ${CALLERID(all)},
${UNIQUEID})
same = n,Set(MAX_REPEAT=5)
same = n(readschedule),Swift(${AAA_OUTPUT})
same = n,Set(MAX_REPEAT=$[${MAX_REPEAT}-1])
same = n,Gotoif($[${MAX_REPEAT} = 0]?timeout_hangup)
same = n,Read(return_id,AAA/end_of_schedule,${read_length},,,2)
same = n,Gotoif($[ ${LEN(${return_id})} = 0]?readschedule)
same = n,Set(account_id=${return_id})
same = n,Goto(validateid)
same = n(timeout_hangup),Swift(No ${truck_text} entered. Goodbye)
same = n,Hangup()
same = n(badid),Set(AAA_OUTPUT=Invalid ${truck_text} ${account_id})
same = n,Set(AAA_CHECKED_IN()=${comp_num}, ${get_param2},
${account_id}, ${AAA_OUTPUT}, I, ${CALLERID(num)}, ${CALLERID(all)},
${UNIQUEID})
same = n,Swift(${AAA_OUTPUT})
same = n,Goto(getid)
same = n(no_schedule),Set(AAA_OUTPUT=No schedule found for
${truck_text} ${account_id})
same = n,Set(AAA_CHECKED_IN()=${comp_num}, ${get_param2},
${account_id}, ${AAA_OUTPUT}, N, ${CALLERID(num)}, ${CALLERID(all)},
${UNIQUEID})
same = n,Swift(${AAA_OUTPUT})
same = n,Goto(getid)
Thanks in advance
-Justin
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup not detected on callback
Hi,
I'm trying to set up a call-back system using auto-dialout files. I
want the call to be terminated when a specific timeout (defined in the
.call file) is detected. Both parties should then be hangup.
The problem is that the timeout is never detected... How to solve this?
Thank you,
Pierre
.call file
--
Channel: IAX2/:@xxx.xxx.xxx.xxx/01
Callerid: 1
MaxRetries: 5
RetryTime: 60
WaitTime: 30
Context: test
Extension: 02
Priority: 1
SetVar: ato=30
SetVar: act=testaccount
extensions.conf
---
[test]
exten = _XX,1,SetAccount(${act})
exten = _XX,2,AbsoluteTimeout(${ato})
exten = _XX,3,Answer()
exten = _XX,4,Dial(IAX2/:@xxx.xxx.xxx.xxx/${EXTEN})
exten = _XX,5,Congestion()
exten = _XX,102,Busy()
exten = s,1,DigitTimeout,10
exten = s,2,ResponseTimeout,10
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
exten = T,1,Playback(vm-goodbye)
exten = T,2,Hangup
CLI output
--
-- Attempting call on IAX2/:@xxx.xxx.xxx.xxx/01 for
[EMAIL PROTECTED]:1 (Retry 1)
-- Call accepted by xxx.xxx.xxx.xxx (format ulaw)
-- Format for call is ulaw
Channel IAX2/xxx.xxx.xxx.xxx:4569/1 was answered.
-- Executing SetAccount(IAX2/xxx.xxx.xxx.xxx:4569/1,
testaccount) in new stack
-- Executing AbsoluteTimeout(IAX2/xxx.xxx.xxx.xxx:4569/1, 30)
in new stack
-- Set Absolute Timeout to 30
-- Executing Answer(IAX2/xxx.xxx.xxx.xxx:4569/1, ) in new stack
-- Executing Dial(IAX2/xxx.xxx.xxx.xxx:4569/1,
IAX2/:@xxx.xxx.xxx.xxx/02) in new stack
-- Called :@xxx.xxx.xxx.xxx/02
-- Call accepted by xxx.xxx.xxx.xxx (format ulaw)
-- Format for call is ulaw
-- IAX2/xxx.xxx.xxx.xxx:4569/2 is ringing
-- IAX2/xxx.xxx.xxx.xxx:4569/2 stopped sounds
-- IAX2/xxx.xxx.xxx.xxx:4569/2 answered IAX2/xxx.xxx.xxx.xxx:4569/1
-- Attempting native bridge of IAX2/xxx.xxx.xxx.xxx:4569/1 and
IAX2/xxx.xxx.xxx.xxx:4569/2
-- Channel 'IAX2/xxx.xxx.xxx.xxx:4569/2' ready to transfer
-- Channel 'IAX2/xxx.xxx.xxx.xxx:4569/1' ready to transfer
-- Releasing IAX2/xxx.xxx.xxx.xxx:4569/1 and
IAX2/xxx.xxx.xxx.xxx:4569/2
-- Hungup 'IAX2/xxx.xxx.xxx.xxx:4569/2'
== Spawn extension (test, 02, 4) exited non-zero on
'IAX2/xxx.xxx.xxx.xxx:4569/1'
Oct 3 19:14:04 NOTICE[1041]: chan_iax2.c:1378 iax2_destroy: Avoiding
IAX destroy deadlock
-- Hungup 'IAX2/xxx.xxx.xxx.xxx:4569/1'
Oct 3 19:14:04 NOTICE[1092]: pbx_spool.c:242 attempt_thread: Call
completed to IAX2/:@xxx.xxx.xxx.xxx/01
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hangup not detected on X100P
What version of *? I'm using 0.7.1 and it still has occasional problems detecting call hangup. John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, March 31, 2004 8:54 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Hangup not detected on X100P On Wed, 2004-03-31 at 10:14, Matt Bridges wrote: I've configured my [*] to dial the pstn which is working like a charm. I've also configured an extension to ring when the PSTN line is ringing which is also working brilliantly, but, sometimes it doesn't detect that the call has been hungup. I've had a look on voip-info and checked the conf files but I can't see anything that I've missed. Google for disconnect supervision. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hangup not detected on X100P
I've using CVS-03/30/04-14:38:02 Not sure where else to get the version number. -Original Message- From: John Vogel [mailto:[EMAIL PROTECTED] Sent: 01 April 2004 16:45 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Hangup not detected on X100P What version of *? I'm using 0.7.1 and it still has occasional problems detecting call hangup. John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, March 31, 2004 8:54 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Hangup not detected on X100P On Wed, 2004-03-31 at 10:14, Matt Bridges wrote: I've configured my [*] to dial the pstn which is working like a charm. I've also configured an extension to ring when the PSTN line is ringing which is also working brilliantly, but, sometimes it doesn't detect that the call has been hungup. I've had a look on voip-info and checked the conf files but I can't see anything that I've missed. Google for disconnect supervision. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup not detected on X100P
I've configured my [*] to dial the pstn which is working like a charm. I've also configured an extension to ring when the PSTN line is ringing which is also working brilliantly, but, sometimes it doesn't detect that the call has been hungup. I've had a look on voip-info and checked the conf files but I can't see anything that I've missed. Cheers Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup not detected on X100P
Matt Bridges wrote: I've configured my [*] to dial the pstn which is working like a charm. I've also configured an extension to ring when the PSTN line is ringing which is also working brilliantly, but, sometimes it doesn't detect that the call has been hungup. I've had a look on voip-info and checked the conf files but I can't see anything that I've missed. Funny you should say that.. I have just updated my system to the latest CVS of zaptel, libpri asterisk and asterisk-addons and now it is screwing up.. If a call comes in on the X100P the sip phones (3 of them) and one analog cordless phone (connected to a single port TDM400P) ring.. When I answer one of them the others continue to ring.. I have to go to each one and answer them to stop them ringing but I cannot answer the call.. The line is then tied up and I have to kill the server to get it cleared.. If I try and make a call I cannot hear any audio on the sip phones and when I hang up either side the line stays connected.. I thing the current CVS is broken.. What the latest stable version? 0.7.2 or 1.0 Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup not detected on X100P
On Wed, 2004-03-31 at 10:14, Matt Bridges wrote: I've configured my [*] to dial the pstn which is working like a charm. I've also configured an extension to ring when the PSTN line is ringing which is also working brilliantly, but, sometimes it doesn't detect that the call has been hungup. I've had a look on voip-info and checked the conf files but I can't see anything that I've missed. Google for disconnect supervision. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
