[asterisk-users] EAGI script with missing audio on /dev/fd/3
Dear all i'm trying to access to the input audio raw stream with a very basic EAGI script: #!/bin/sh echo "EXEC Queue 2001" cat /dev/fd/3 > /tmp/pippo This is my dialplan: exten => 001,NoOp(test) exten => 001,n,Answer exten => 001,n,EAGI(/tmp/my-eagi.agi) When i call, the script is executed and the call goes in queue, i can hear the MOH, the file /tmp/pippo is created but it is empty. Any idea or suggestion? PS: if i use the application monitor or MixMonitor the call is recorded correctly. I'm using Asterisk 1.6.2.9-2+squeeze12 Thanks -- /*/ nik600 http://www.kumbe.it -- _ -- 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] how to join 2 channels using AGI/AMI
finally i've found that the SIP gateway i'm using is based on a DAHDI channel and it seems that on outgoing calls, if the called leg sends some digit they are not forwarded toAsterisk. i'm investigating on it 2016-07-01 4:25 GMT+02:00 Steve Edwards <asterisk@sedwards.com>: > On Fri, 1 Jul 2016, nik600 wrote: > > i've tried rfc2833,inband and info having the same behaviour in all >> situation. >> >> 2016-06-30 23:53 GMT+02:00 nik600 <nik...@gmail.com>: >> sorry for top-posting, the two topics started with 2 different >> reason subject, but then we finished on the same problem. >> btw,the 2 show channel are reported above: >> >> the channel with DTMF working >> >> kcenter*CLI> core show channel SIP/pbx2-04b9 >> -- General -- >>Name: SIP/pbx2-04b9 >>Type: SIP >>UniqueID: 1467323106.1275 >> Caller ID: >> Caller ID Name: >> DNID Digits: >>Language: en >> State: Up (6) >> Rings: 0 >> NativeFormats: 0x4 (ulaw) >> WriteFormat: 0x4 (ulaw) >> ReadFormat: 0x4 (ulaw) >> WriteTranscode: No >> ReadTranscode: No >> 1st File Descriptor: 29 >> Frames in: 325 >> Frames out: 44 >> Time to Hangup: 0 >>Elapsed Time: 0h0m6s >> Direct Bridge: >> Indirect Bridge: >> -- PBX -- >> Context: c_Queues >> Extension: 01 >>Priority: 1 >> Call Group: 0 >>Pickup Group: 0 >> Application: Read >>Data: RESPONSE,beep,1,s,3,5 >> Blocking in: ast_waitfor_nandfds >> >> >> the channel with DTMF not working: >> >> kcenter*CLI> core show channel Local/user1@c_Queues-5d47;1 >> -- General -- >>Name: Local/user1@c_Queues-5d47;1 >>Type: Local >>UniqueID: 1467323176.1277 >> Caller ID: zzz >> Caller ID Name: zzz >> DNID Digits: (N/A) >>Language: en >> State: Ringing (5) >> Rings: 0 >> NativeFormats: 0x4 (ulaw) >> WriteFormat: 0x4 (ulaw) >> ReadFormat: 0x4 (ulaw) >> WriteTranscode: No >> ReadTranscode: No >> 1st File Descriptor: -1 >> Frames in: 1 >> Frames out: 0 >> Time to Hangup: 0 >>Elapsed Time: 0h0m13s >> Direct Bridge: >> Indirect Bridge: >> -- PBX -- >> Context: c_Queues >> Extension: 01 >>Priority: 1 >> Call Group: 0 >>Pickup Group: 0 >> Application: AppQueue >>Data: (Outgoing Line) >> Blocking in: ast_waitfor_nandfds >> >> the only difference i see is the "1st File Descriptor" pointing to -1 >> > > 1) The 'frames' counts look odd to me. > > 2) Does a comparison of 'sip show channel' yield any clues? > > 3) Can you use 'sipdtmfmode()' to set a mode that works? > > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > https://www.linkedin.com/in/steve-edwards-4244281 > > -- > _ > -- 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 > -- /*/ nik600 http://www.kumbe.it -- _ -- 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] how to join 2 channels using AGI/AMI
to simplify the scenario, i've changed some settings to create a more
simple test-case:
i'm using this callfile:
Channel: DAHDI/g0/{mycellnumber}
Context:mytestdtmf
Extension:01
Priority:1
and this is my dialplan:
[mytestdtmf]
exten =>01,1,Answer
exten =>01,n,Read(digito,,1)
exten =>01,n,SayDigits(${digito})
Any idea?
2016-07-01 0:13 GMT+02:00 nik600 <nik...@gmail.com>:
> i've tried rfc2833,inband and info having the same behaviour in all
> situation.
>
> 2016-06-30 23:53 GMT+02:00 nik600 <nik...@gmail.com>:
>
>> sorry for top-posting, the two topics started with 2 different reason
>> subject, but then we finished on the same problem.
>>
>> btw,the 2 show channel are reported above:
>>
>> the channel with DTMF working
>>
>> kcenter*CLI> core show channel SIP/pbx2-04b9
>> -- General --
>>Name: SIP/pbx2-04b9
>>Type: SIP
>>UniqueID: 1467323106.1275
>> Caller ID:
>> Caller ID Name:
>> DNID Digits:
>>Language: en
>> State: Up (6)
>> Rings: 0
>> NativeFormats: 0x4 (ulaw)
>> WriteFormat: 0x4 (ulaw)
>> ReadFormat: 0x4 (ulaw)
>> WriteTranscode: No
>> ReadTranscode: No
>> 1st File Descriptor: 29
>> Frames in: 325
>> Frames out: 44
>> Time to Hangup: 0
>>Elapsed Time: 0h0m6s
>> Direct Bridge:
>> Indirect Bridge:
>> -- PBX --
>> Context: c_Queues
>> Extension: 01
>>Priority: 1
>> Call Group: 0
>>Pickup Group: 0
>> Application: Read
>>Data: RESPONSE,beep,1,s,3,5
>> Blocking in: ast_waitfor_nandfds
>>
>>
>> the channel with DTMF not working:
>>
>> kcenter*CLI> core show channel Local/user1@c_Queues-5d47;1
>> -- General --
>>Name: Local/user1@c_Queues-5d47;1
>>Type: Local
>>UniqueID: 1467323176.1277
>> Caller ID: zzz
>> Caller ID Name: zzz
>> DNID Digits: (N/A)
>>Language: en
>> State: Ringing (5)
>> Rings: 0
>> NativeFormats: 0x4 (ulaw)
>> WriteFormat: 0x4 (ulaw)
>> ReadFormat: 0x4 (ulaw)
>> WriteTranscode: No
>> ReadTranscode: No
>> 1st File Descriptor: -1
>> Frames in: 1
>> Frames out: 0
>> Time to Hangup: 0
>>Elapsed Time: 0h0m13s
>> Direct Bridge:
>> Indirect Bridge:
>> -- PBX --
>> Context: c_Queues
>> Extension: 01
>>Priority: 1
>> Call Group: 0
>>Pickup Group: 0
>> Application: AppQueue
>>Data: (Outgoing Line)
>> Blocking in: ast_waitfor_nandfds
>>
>> the only difference i see is the "1st File Descriptor" pointing to -1
>>
>> 2016-06-30 23:29 GMT+02:00 Steve Edwards <asterisk@sedwards.com>:
>>
>>> Please don't top post.
>>>
>>> On Thu, 30 Jun 2016, nik600 wrote:
>>>
>>> this is the point, and the strange thing:DTMF is set to rfc2833, but is
>>>> working both on incoming and outgoing calls, it is not working only on
>>>> calls generated with the Originate AMI command, or with the queue member
>>>> that point to Local dialplan, as you suggested
>>>>
>>>
>>> Does 'show channel' on a leg originated by a handset differ from a leg
>>> originated by AMI?
>>>
>>> --
>>> Thanks in advance,
>>> -----
>>> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
>>> PST
>>> https://www.linkedin.com/in/steve-edwards-4244281
>>> --
>>> _
>>> -- 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
>>>
>>
>>
>>
>> --
>> /*/
>> nik600
>> http://www.kumbe.it
>>
>
>
>
> --
> /*/
> nik600
> http://www.kumbe.it
>
--
/*/
nik600
http://www.kumbe.it
--
_
-- 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
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Re: [asterisk-users] how to join 2 channels using AGI/AMI
i've tried rfc2833,inband and info having the same behaviour in all situation. 2016-06-30 23:53 GMT+02:00 nik600 <nik...@gmail.com>: > sorry for top-posting, the two topics started with 2 different reason > subject, but then we finished on the same problem. > > btw,the 2 show channel are reported above: > > the channel with DTMF working > > kcenter*CLI> core show channel SIP/pbx2-04b9 > -- General -- >Name: SIP/pbx2-04b9 >Type: SIP >UniqueID: 1467323106.1275 > Caller ID: > Caller ID Name: > DNID Digits: >Language: en > State: Up (6) > Rings: 0 > NativeFormats: 0x4 (ulaw) > WriteFormat: 0x4 (ulaw) > ReadFormat: 0x4 (ulaw) > WriteTranscode: No > ReadTranscode: No > 1st File Descriptor: 29 > Frames in: 325 > Frames out: 44 > Time to Hangup: 0 >Elapsed Time: 0h0m6s > Direct Bridge: > Indirect Bridge: > -- PBX -- > Context: c_Queues > Extension: 01 >Priority: 1 > Call Group: 0 >Pickup Group: 0 > Application: Read >Data: RESPONSE,beep,1,s,3,5 > Blocking in: ast_waitfor_nandfds > > > the channel with DTMF not working: > > kcenter*CLI> core show channel Local/user1@c_Queues-5d47;1 > -- General -- >Name: Local/user1@c_Queues-5d47;1 >Type: Local >UniqueID: 1467323176.1277 > Caller ID: zzz > Caller ID Name: zzz > DNID Digits: (N/A) >Language: en > State: Ringing (5) > Rings: 0 > NativeFormats: 0x4 (ulaw) > WriteFormat: 0x4 (ulaw) > ReadFormat: 0x4 (ulaw) > WriteTranscode: No > ReadTranscode: No > 1st File Descriptor: -1 > Frames in: 1 > Frames out: 0 > Time to Hangup: 0 >Elapsed Time: 0h0m13s > Direct Bridge: > Indirect Bridge: > -- PBX -- > Context: c_Queues > Extension: 01 >Priority: 1 > Call Group: 0 >Pickup Group: 0 > Application: AppQueue >Data: (Outgoing Line) > Blocking in: ast_waitfor_nandfds > > the only difference i see is the "1st File Descriptor" pointing to -1 > > 2016-06-30 23:29 GMT+02:00 Steve Edwards <asterisk@sedwards.com>: > >> Please don't top post. >> >> On Thu, 30 Jun 2016, nik600 wrote: >> >> this is the point, and the strange thing:DTMF is set to rfc2833, but is >>> working both on incoming and outgoing calls, it is not working only on >>> calls generated with the Originate AMI command, or with the queue member >>> that point to Local dialplan, as you suggested >>> >> >> Does 'show channel' on a leg originated by a handset differ from a leg >> originated by AMI? >> >> -- >> Thanks in advance, >> - >> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST >> https://www.linkedin.com/in/steve-edwards-4244281 >> -- >> _____ >> -- 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 >> > > > > -- > /*/ > nik600 > http://www.kumbe.it > -- /*/ nik600 http://www.kumbe.it -- _ -- 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] how to join 2 channels using AGI/AMI
sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-04b9 -- General -- Name: SIP/pbx2-04b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: Caller ID Name: DNID Digits: Language: en State: Up (6) Rings: 0 NativeFormats: 0x4 (ulaw) WriteFormat: 0x4 (ulaw) ReadFormat: 0x4 (ulaw) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 29 Frames in: 325 Frames out: 44 Time to Hangup: 0 Elapsed Time: 0h0m6s Direct Bridge: Indirect Bridge: -- PBX -- Context: c_Queues Extension: 01 Priority: 1 Call Group: 0 Pickup Group: 0 Application: Read Data: RESPONSE,beep,1,s,3,5 Blocking in: ast_waitfor_nandfds the channel with DTMF not working: kcenter*CLI> core show channel Local/user1@c_Queues-5d47;1 -- General -- Name: Local/user1@c_Queues-5d47;1 Type: Local UniqueID: 1467323176.1277 Caller ID: zzz Caller ID Name: zzz DNID Digits: (N/A) Language: en State: Ringing (5) Rings: 0 NativeFormats: 0x4 (ulaw) WriteFormat: 0x4 (ulaw) ReadFormat: 0x4 (ulaw) WriteTranscode: No ReadTranscode: No 1st File Descriptor: -1 Frames in: 1 Frames out: 0 Time to Hangup: 0 Elapsed Time: 0h0m13s Direct Bridge: Indirect Bridge: -- PBX -- Context: c_Queues Extension: 01 Priority: 1 Call Group: 0 Pickup Group: 0 Application: AppQueue Data: (Outgoing Line) Blocking in: ast_waitfor_nandfds the only difference i see is the "1st File Descriptor" pointing to -1 2016-06-30 23:29 GMT+02:00 Steve Edwards <asterisk@sedwards.com>: > Please don't top post. > > On Thu, 30 Jun 2016, nik600 wrote: > > this is the point, and the strange thing:DTMF is set to rfc2833, but is >> working both on incoming and outgoing calls, it is not working only on >> calls generated with the Originate AMI command, or with the queue member >> that point to Local dialplan, as you suggested >> > > Does 'show channel' on a leg originated by a handset differ from a leg > originated by AMI? > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > https://www.linkedin.com/in/steve-edwards-4244281 > -- > _ > -- 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 > -- /*/ nik600 http://www.kumbe.it -- _ -- 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] how to join 2 channels using AGI/AMI
this is the point, and the strange thing:
DTMF is set to rfc2833, but is working both on incoming and outgoing calls,
it is not working only on calls generated with the Originate AMI command,
or with the queue member that point to Local dialplan, as you suggested
2016-06-30 22:53 GMT+02:00 John Kiniston <johnkinis...@gmail.com>:
> Looking at your logs it looks like you may need to modify your sip.conf,
> Check with your provider as to what kind of DTMF they support and configure
> sip.conf to use that type of signalling.
>
>
>
> On Thu, Jun 30, 2016 at 1:18 PM, nik600 <nik...@gmail.com> wrote:
>
>> thanks John
>>
>> yeah, your approach is much siple, i've tried it but i'm not able do
>> detect DTMF tones.
>>
>> it seems that on calls that i receive DTMF tones are handled correctly,
>> but on calls generated from Asterisk to the world when the called side
>> sends some DTMF digits they are not detected:
>>
>> -- Executing [s@macro-myconnector:1] NoOp("SIP/pbx2-04b2", "")
>> in new stack
>> -- Executing [s@macro-myconnector:2] Read("SIP/pbx2-04b2",
>> "RESPONSE,beep,1,s,3,5") in new stack
>> -- Accepting a maximum of 1 digits.
>> -- Playing 'beep.gsm' (language 'en')
>> ...
>> -- User entered nothing, 2 chances left
>> -- Playing 'beep.gsm' (language 'en')
>> ...
>> -- User entered nothing, 1 chance left
>> -- Playing 'beep.gsm' (language 'en')
>> ...
>> -- User entered nothing.
>> -- Executing [s@macro-myconnector:3] GotoIf("SIP/pbx2-04b2",
>> "1?REJECT,1") in new stack
>>
>> Any idea?
>>
>>
>>
>>
>>
>>
>> 2016-06-30 21:50 GMT+02:00 John Kiniston <johnkinis...@gmail.com>:
>>
>>> I think a simpler way to do this would be to define an member in your
>>> queues.conf that points to a local channel that calls the remote users cell
>>> phone.
>>>
>>> You can use the M option in your dial to run a macro to prompt the user
>>> to accept the call.
>>>
>>> Here's my connector macro, I call it with:
>>>
>>> Dial(LOCAL/${CELLPHONE}@intern,60,M(connector))
>>>
>>> [macro-connector]
>>> exten => s,1,NoOP()
>>> same => n(TOP),Read(RESPONSE,beep,1,s,3,5); 3 tries
>>> with 5 seconds to respond each time
>>> same => n,GotoIf($[${LEN(${RESPONSE})} = 0]?REJECT,1);If we
>>> didn't get a response try and fail gracefully
>>> same => n,GotoIf($["${RESPONSE}" = "1"]?ACCEPT,1);Take
>>> the call
>>> same => n,GotoIf($["${RESPONSE}" = "2"]?REJECT,1);Reject
>>> the Call
>>> same => n,Goto(s,TOP)
>>>
>>> exten => ACCEPT,1,NoOP();Just
>>> connect the caller and callee
>>> same => n,Playback(pls-wait-connect-call)
>>> same => n,MacroExit();Return
>>>
>>> exten => REJECT,1,NoOP()
>>> same => n,Playback(beep)
>>> same => n,Set(MACRO_RESULT=BUSY);Reject the
>>> call
>>> same => n,Hangup()
>>> same => n,MacroExit();Return
>>>
>>>
>>> On Thu, Jun 30, 2016 at 6:08 AM, nik600 <nik...@gmail.com> wrote:
>>>
>>>> Dear all
>>>>
>>>> i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if
>>>> is possible to configure a scenario like this:
>>>>
>>>> 1) receive a call and put it on-hold in a queue (OK)
>>>> 2) monitor the queue and trigger an outbound call to a remote number
>>>> using AMI, setting the channel of the on-hold on a specific var named
>>>> channel2Link (OK)
>>>> 3) when the remote number answer, trigger an AGI/diaplan script that
>>>> ask to accept the call pressing a specific key (OK)
>>>> 4) if right key is pressed redirect the current call to
>>>> the channel2Link, connecting the call in queue with the remote number (?)
>>>>
>>>> Step 1,2,3 works properly but i'm not able to link the two channels,
>>>> even using redirect,goto or pickupChan.
>>>>
>>>> Any idea or help will be appreciated!
>>>>
>>>> Thanks
>>>>
>>>> --
>>>> /*/
>>
Re: [asterisk-users] problem with DTMF detection on calls created with Originate AMI command
i'm using Asterisk 1.6.2.9-2+squeeze12
2016-06-30 22:14 GMT+02:00 Richard Mudgett <rmudg...@digium.com>:
>
>
> On Thu, Jun 30, 2016 at 3:00 PM, nik600 <nik...@gmail.com> wrote:
>
>> Dear all
>>
>> i'm creating an outgoing call to number xxx with this command:
>>
>> http://host:port/mxml?action=Originate=Local/xxx@to-external
>> =testDTMF=cRETEUNICA=1
>>
>> wich points correctly to this portion of dialplan:
>>
>> [cRETEUNICA]
>>
>> exten => testDTMF,1,Answer
>> exten => testDTMF,n,Read(digito,,1)
>> exten => testDTMF,n,SayDigits(${digito})
>>
>> The point is that the recognition goes in timeout and i get an error on
>> ast_waitfordigit_full
>>
>> -- Executing [testDTMF@cRETEUNICA:1] Answer("SIP/pbx2-04ad", "")
>> in new stack
>> -- Executing [testDTMF@cRETEUNICA:2] Read("SIP/pbx2-04ad",
>> "digito,,1") in new stack
>> [Jun 30 21:56:56] WARNING[5617]: channel.c:2558 ast_waitfordigit_full:
>> Unexpected control subclass '-1'
>> -- User entered nothing.
>>
>
> You didn't specify the Asterisk version. You can ignore this message.
> Current versions simply suppress this message for -1 in that routine.
>
> Richard
>
>
> --
> _
> -- 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
>
--
/*/
nik600
http://www.kumbe.it
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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Re: [asterisk-users] how to join 2 channels using AGI/AMI
thanks John
yeah, your approach is much siple, i've tried it but i'm not able do detect
DTMF tones.
it seems that on calls that i receive DTMF tones are handled correctly, but
on calls generated from Asterisk to the world when the called side sends
some DTMF digits they are not detected:
-- Executing [s@macro-myconnector:1] NoOp("SIP/pbx2-04b2", "") in
new stack
-- Executing [s@macro-myconnector:2] Read("SIP/pbx2-04b2",
"RESPONSE,beep,1,s,3,5") in new stack
-- Accepting a maximum of 1 digits.
-- Playing 'beep.gsm' (language 'en')
...
-- User entered nothing, 2 chances left
-- Playing 'beep.gsm' (language 'en')
...
-- User entered nothing, 1 chance left
-- Playing 'beep.gsm' (language 'en')
...
-- User entered nothing.
-- Executing [s@macro-myconnector:3] GotoIf("SIP/pbx2-04b2",
"1?REJECT,1") in new stack
Any idea?
2016-06-30 21:50 GMT+02:00 John Kiniston <johnkinis...@gmail.com>:
> I think a simpler way to do this would be to define an member in your
> queues.conf that points to a local channel that calls the remote users cell
> phone.
>
> You can use the M option in your dial to run a macro to prompt the user to
> accept the call.
>
> Here's my connector macro, I call it with:
>
> Dial(LOCAL/${CELLPHONE}@intern,60,M(connector))
>
> [macro-connector]
> exten => s,1,NoOP()
> same => n(TOP),Read(RESPONSE,beep,1,s,3,5); 3 tries with
> 5 seconds to respond each time
> same => n,GotoIf($[${LEN(${RESPONSE})} = 0]?REJECT,1);If we
> didn't get a response try and fail gracefully
> same => n,GotoIf($["${RESPONSE}" = "1"]?ACCEPT,1);Take the
> call
> same => n,GotoIf($["${RESPONSE}" = "2"]?REJECT,1);Reject
> the Call
> same => n,Goto(s,TOP)
>
> exten => ACCEPT,1,NoOP();Just
> connect the caller and callee
> same => n,Playback(pls-wait-connect-call)
> same => n,MacroExit();Return
>
> exten => REJECT,1,NoOP()
> same => n,Playback(beep)
> same => n,Set(MACRO_RESULT=BUSY);Reject the call
> same => n,Hangup()
> same => n,MacroExit();Return
>
>
> On Thu, Jun 30, 2016 at 6:08 AM, nik600 <nik...@gmail.com> wrote:
>
>> Dear all
>>
>> i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is
>> possible to configure a scenario like this:
>>
>> 1) receive a call and put it on-hold in a queue (OK)
>> 2) monitor the queue and trigger an outbound call to a remote number
>> using AMI, setting the channel of the on-hold on a specific var named
>> channel2Link (OK)
>> 3) when the remote number answer, trigger an AGI/diaplan script that ask
>> to accept the call pressing a specific key (OK)
>> 4) if right key is pressed redirect the current call to the channel2Link,
>> connecting the call in queue with the remote number (?)
>>
>> Step 1,2,3 works properly but i'm not able to link the two channels, even
>> using redirect,goto or pickupChan.
>>
>> Any idea or help will be appreciated!
>>
>> Thanks
>>
>> --
>> /*/
>> nik600
>> http://www.kumbe.it
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> A human being should be able to change a diaper, plan an invasion, butcher
> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate, act alone, solve equations, analyze a new problem, pitch manure,
> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
> ---Heinlein
>
> --
> _
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> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
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nik600
http://www.kumbe.it
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[asterisk-users] problem with DTMF detection on calls created with Originate AMI command
Dear all
i'm creating an outgoing call to number xxx with this command:
http://host:port/mxml?action=Originate=Local/xxx@to-external
=testDTMF=cRETEUNICA=1
wich points correctly to this portion of dialplan:
[cRETEUNICA]
exten => testDTMF,1,Answer
exten => testDTMF,n,Read(digito,,1)
exten => testDTMF,n,SayDigits(${digito})
The point is that the recognition goes in timeout and i get an error on
ast_waitfordigit_full
-- Executing [testDTMF@cRETEUNICA:1] Answer("SIP/pbx2-04ad", "") in
new stack
-- Executing [testDTMF@cRETEUNICA:2] Read("SIP/pbx2-04ad",
"digito,,1") in new stack
[Jun 30 21:56:56] WARNING[5617]: channel.c:2558 ast_waitfordigit_full:
Unexpected control subclass '-1'
-- User entered nothing.
Any idea?
if i call from number xxx to an extension that goes to testDTMF@cRETEUNICA
it works properly.
Thanks
--
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nik600
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Re: [asterisk-users] how to join 2 channels using AGI/AMI
oh, yes! Many thanks 2016-06-30 15:28 GMT+02:00 Guido Falsi <m...@madpilot.net>: > On 06/30/16 15:08, nik600 wrote: > > Dear all > > > > i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is > > possible to configure a scenario like this: > > > > 1) receive a call and put it on-hold in a queue (OK) > > 2) monitor the queue and trigger an outbound call to a remote number > > using AMI, setting the channel of the on-hold on a specific var named > > channel2Link (OK) > > 3) when the remote number answer, trigger an AGI/diaplan script that ask > > to accept the call pressing a specific key (OK) > > 4) if right key is pressed redirect the current call to > > the channel2Link, connecting the call in queue with the remote number (?) > > > > Step 1,2,3 works properly but i'm not able to link the two channels, > > even using redirect,goto or pickupChan. > > > > Any idea or help will be appreciated! > > > > I think the way to achieve that is by using the Bridge application: > > https://wiki.asterisk.org/wiki/display/AST/Bridge+Application > https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Bridge > > -- > Guido Falsi <m...@madpilot.net> > > -- > _ > -- 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 > -- /*/ nik600 http://www.kumbe.it -- _ -- 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] how to join 2 channels using AGI/AMI
Dear all i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is possible to configure a scenario like this: 1) receive a call and put it on-hold in a queue (OK) 2) monitor the queue and trigger an outbound call to a remote number using AMI, setting the channel of the on-hold on a specific var named channel2Link (OK) 3) when the remote number answer, trigger an AGI/diaplan script that ask to accept the call pressing a specific key (OK) 4) if right key is pressed redirect the current call to the channel2Link, connecting the call in queue with the remote number (?) Step 1,2,3 works properly but i'm not able to link the two channels, even using redirect,goto or pickupChan. Any idea or help will be appreciated! Thanks -- /*****/ nik600 http://www.kumbe.it -- _ -- 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] how to know status of asterisk from php
Hi, you can use the PHPAgi project http://phpagi.sourceforge.net/ Otherwise, if you want a more high-level approach you can use the MXML interface, you will communicate with HTTP GET request and obtaing XML response directly from Asterisk. Enabling the http manager interface you will get enabled some manager commands on the port 8088 Ie, you can Login with: http://your-asterisk-ip:8088/mxml?action=loginusername=$this-usersecret=$this-pass Some example commands: http://your-asterisk-ip:8088/mxml?action=queuestatus http://your-asterisk-ip:8088/mxml?action=SipPeers http://your-asterisk-ip:8088/mxml?action=status http://your-asterisk-ip:8088/mxml?action=DBputfamily=$familykey=$keyVal=$val http://your-asterisk-ip:8088/mxml?action=QueueAddqueue=$queueinterface=$interface http://your-asterisk-ip:8088/mxml?action=QueueRemovequeue=$queueinterface=$interface http://your-asterisk-ip:8088/mxml?action=QueuePausequeue=$queueinterface=$interfacePaused=1 http://your-asterisk-ip:8088/mxml?action=QueuePausequeue=$queueinterface=$interfacePaused=0 And so on On Wed, Apr 27, 2011 at 1:22 PM, virendra bhati virbh...@gmail.com wrote: Hi How to know status of Asterisk,Mysql. PRI lines and other services from PHP scripts ? Thanks and regards Virendra Bhati +91-9172341457 -- _ -- 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 -- /*/ nik600 http://www.kumbe.it -- _ -- 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] understand which asterisk thread is consuming CPU
Dear all using top -H i can see that some asterisk thread are consuming many CPU (sometimes more than 50%) Is there a way to understand what is doing the process with pid 9429 ? i've tried the core show thread command, but it doesn't seem to print any PID information. Thanks to all in advance PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 9429 root 20 0 662m 93m 5596 S 23 3.1 29:28.91 asterisk 13261 root 20 0 662m 93m 5596 S 10 3.1 0:04.54 asterisk 15646 root 20 0 662m 93m 5596 S4 3.1 0:00.82 asterisk 15648 root 20 0 662m 93m 5596 S3 3.1 0:00.88 asterisk 9413 root 20 0 662m 93m 5596 S3 3.1 1:25.85 asterisk 13987 root 20 0 662m 93m 5596 S3 3.1 0:03.22 asterisk 15743 root 20 0 662m 93m 5596 S2 3.1 0:00.82 asterisk 9432 root 20 0 662m 93m 5596 S1 3.1 13:06.55 asterisk 13778 root 20 0 662m 93m 5596 S1 3.1 0:04.82 asterisk 9412 root 20 0 662m 93m 5596 S1 3.1 0:34.84 asterisk 9465 root 20 0 662m 93m 5596 S1 3.1 0:39.63 asterisk 13351 root 20 0 662m 93m 5596 S1 3.1 0:03.02 asterisk 13654 root 20 0 662m 93m 5596 S1 3.1 0:02.64 asterisk 14758 root 20 0 662m 93m 5596 S1 3.1 0:02.22 asterisk 14911 root 20 0 662m 93m 5596 S1 3.1 0:03.28 asterisk 15004 root 20 0 662m 93m 5596 S1 3.1 0:02.04 asterisk 15006 root 20 0 662m 93m 5596 S1 3.1 0:02.68 asterisk 15126 root 20 0 662m 93m 5596 S1 3.1 0:02.50 asterisk 15127 root 20 0 662m 93m 5596 S1 3.1 0:02.82 asterisk 15711 root 20 0 662m 93m 5596 S1 3.1 0:00.76 asterisk 15892 root 20 0 662m 93m 5596 S1 3.1 0:00.68 asterisk 15956 root 20 0 662m 93m 5596 S1 3.1 0:00.68 asterisk -- /*/ nik600 http://www.kumbe.it -- _ -- 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] 1.6 how to use groupcount and counteronpeer in queues to avoid ringinuse
Dear all i'm planning an upgrade of some asterisk installation from 1.4.32 to 1.6.0.28 (as i think it should be the most stable now). Reading the UPGRADE-1.6.txt file i've noticed that: * SIP: The call-limit option is marked as deprecated. It still works in this version of Asterisk, but will be removed in the following version. Please use the groupcount functions in the dialplan to enforce call limits. The limitonpeer configuration option is now renamed to counteronpeer. As i've experienced some problem with 1.4 release about call-limit, i'd like to test this new counteronpeer functionality, but how to handle the ringinuse parmeter in queues.conf ? Basically i need that a sip user can make and receive more than one call (like a call-limit 3 setting) but i don't want that this interface rings if it is in a queue. Is it possible to do that? How? Thanks to all -- /*/ nik600 http://www.kumbe.it -- _ -- 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] [compat] section in asterisk.conf : compatibility with pipe delimiter
Dear all after an upgrade to 1.6 from 1.4 (as explained in the UPGRADE-1.6.txt file) the | delimiter is not working by default. I've added a compat section in asterisk.conf a [options] dontwarn = yes [compat] pbx_realtime=1.4 res_agi=1.4 app_set=1.4 And restarted Asterisk, but i still have problem to have the | delimiter working, [Jun 9 23:20:54] DEBUG[11744]: pbx.c:3122 pbx_extension_helper: Launching 'Queue' -- Executing [...@queues:4] Queue(SIP/PL1999-0003, queue_130) in new stack [Jun 9 23:20:54] DEBUG[11744]: app_queue.c:4804 queue_exec: NO QUEUE_PRIO variable found. Using default. [Jun 9 23:20:54] DEBUG[11744]: app_queue.c:4841 queue_exec: queue: queue_130, options: (null), url: (null), announce: (null), expires: 0, priority: 0 [Jun 9 23:20:54] WARNING[11744]: app_queue.c:4853 queue_exec: Unable to join queue 'queue_130' It seems that Asterisk ignores the | delimiter, if i try with the comma it works. Reading the the upgrade file it seems that the pbx_realtime should affect also the extension.conf settings... where am i wrong? Thanks to all in advance -- /*/ nik600 http://www.kumbe.it -- _ -- 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] problem with ringinuse=no, queue members receive randomly two calls
i've also tied this tests: - changed hardware - upgrade to 1.4.31 - kernel recompiled with 1000 Hz option - changed SO (Slackware 13) - run the system on hardware (no ESXi) But i've not resolved the problem. Do you have any idea? On Thu, May 6, 2010 at 11:54 AM, nik600 nik...@gmail.com wrote: i get may debug messages like this: DEBUG[30684] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=-1) Is because dahdi is not installed? Can this be a possible cause of this behaviour? -- /*/ nik600 http://www.kumbe.it -- _ -- 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] problem with ringinuse=no, queue members receive randomly two calls
i get may debug messages like this: DEBUG[30684] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=-1) Is because dahdi is not installed? Can this be a possible cause of this behaviour? On Tue, May 4, 2010 at 9:54 PM, nik600 nik...@gmail.com wrote: Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the user goes in a status Ring+Inuse. Any idea? Why the call is still dispatched to the user if it is not in the Not in use status? Thanks to all in advance * * LOG (core debug and verbose set to 5) * * #grep PL1038 full [May 4 16:21:08] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '6' (Ringing) [May 4 16:21:08] DEBUG[3035] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:08] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:08] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:08] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 6 (Ringing) [May 4 16:21:08] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '6' (Ringing) [May 4 16:21:08] VERBOSE[30453] logger.c: -- SIP/PL1038-5f7d is ringing [May 4 16:21:08] DEBUG[3035] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:08] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:08] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:08] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 6 (Ringing) [May 4 16:21:08] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '6' (Ringing) [May 4 16:21:08] VERBOSE[30268] logger.c: -- SIP/PL1038-5f7e is ringing [May 4 16:21:10] DEBUG[3035] chan_sip.c: T38 state changed to 0 on channel SIP/PL1038-5f7e [May 4 16:21:10] DEBUG[3035] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:10] DEBUG[3035] chan_sip.c: build_route: Contact hop: sip:pl1...@10.192.37.119 [May 4 16:21:10] DEBUG[30268] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:10] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:10] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:10] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 7 (Ring+Inuse) [May 4 16:21:10] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '7' (Ring+Inuse) [May 4 16:21:10] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:10] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:10] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 7 (Ring+Inuse) [May 4 16:21:10] VERBOSE[30268] logger.c: -- SIP/PL1038-5f7e answered SIP/192.168.55.32-5f59 [May 4 16:21:10] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '7' (Ring+Inuse) [May 4 16:21:14] VERBOSE[30268] logger.c: -- Native bridging SIP/192.168.55.32-5f59 and SIP/PL1038-5f7e [May 4 16:21:14] DEBUG[3035] chan_sip.c: T38 state changed to 0 on channel SIP/PL1038-5f7e [May 4 16:21:14] DEBUG[3035] devicestate.c: Notification of state change to be queued on device/channel SIP/PL1038 [May 4 16:21:14] DEBUG[3035] chan_sip.c: T38 state changed to 0 on channel SIP/PL1038-5f7e [May 4 16:21:14] DEBUG[3022] devicestate.c: No provider found, checking channel drivers for SIP - PL1038 [May 4 16:21:14] DEBUG[3022] chan_sip.c: Checking device state for peer PL1038 [May 4 16:21:14] DEBUG[3022] devicestate.c: Changing state for SIP/PL1038 - state 7 (Ring+Inuse) [May 4 16:21:14] DEBUG[3034] app_queue.c: Device 'SIP/PL1038' changed to state '7' (Ring+Inuse) [May 4 16:21:15] DEBUG[29938] app_queue.c: Trying 'SIP/PL1038' with metric 0 [May 4 16:21:15] DEBUG[29938] app_queue.c: SIP/PL1038 in use, can't receive call [May 4 16:21:16] DEBUG[30097] app_queue.c: Trying 'SIP/PL1038' with metric 0 [May 4 16:21:16] DEBUG[30097] app_queue.c: SIP/PL1038 in use, can't receive call [ * * config * * sip users: [PL1039] context=mycontext callerid=PhoneLine1039 1039 secret=pwd1039 type=peer host=dynamic call-limit=3
[asterisk-users] problem with ringinuse=no, queue members receive randomly two calls
[queue_3] weight=10 wrapuptime=0 strategy=leastrecent joinempty=no retry=0 autopause=yes setinterfacevar=yes eventwhencalled=yes eventmemberstatus=yes ringinuse=no member = SIP/PL1039 -- /*/ nik600 http://www.kumbe.it -- _ -- 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] problems originating an outgoing IAX2 call
I've resolved...it was a limitation of the provider for calls without a CallerID
On Sun, Apr 18, 2010 at 7:43 PM, nik600 nik...@gmail.com wrote:
Dear all
i'm trying to originate an outgoing call with the command originate,
from Asterisk's CLI i'm typing:
CLI originate IAX2/my-iax-provider/number2call application wait 10
[Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr:
prepending 40 to prefs
-- Call accepted by 62.149.202.150 (format ilbc)
-- Format for call is ilbc
-- Hungup 'IAX2/my-iax-provider-5647'
... and nothing happend the hangup is given after 3-4 seconds of the
command
But, if i try to call a dialplan extenstion from a local IAX user the
call works properly
[outgoing_voipvoice]
exten = _X.,1,Dial(IAX2/my-iax-provider/${EXTEN})
-- Accepting AUTHENTICATED call from 82.56.46.69:
requested format = gsm,
requested prefs = (),
actual format = gsm,
host prefs = (),
priority = mine
-- Executing [number2c...@outgoing_voipvoice:1]
Dial(IAX2/localuser-3519, IAX2/my-iax-provider/number2call) in new
stack
[Apr 18 19:34:22] DEBUG[32577]: chan_iax2.c:4000 create_addr:
prepending 2 to prefs
-- Called my-iax-provider/number2call
-- Call accepted by 62.149.202.150 (format ilbc)
-- Format for call is ilbc
-- IAX2/my-iax-provider-25 is making progress passing it to
IAX2/localuser-3519
-- IAX2/my-iax-provider-25 is ringing
-- IAX2/my-iax-provider-25 is making progress passing it to
IAX2/localuser-3519
-- IAX2my-iax-provider-25 stopped sounds
-- IAX2/my-iax-provider-25 answered IAX2/localuser-3519
-- Operating with different codecs 2[0x2 (gsm)] 1024[0x400 (ilbc)]
, can't native bridge...
-- Hungup 'IAX2/my-iax-provider-25'
== Spawn extension (outgoing_voipvoice, number2call, 1) exited
non-zero on 'IAX2/localuser-3519'
-- Hungup 'IAX2/localuser-3519'
I'm having the same problem using the dial from console:
CLI console dial number2c...@outgoing_voipvoice
-- Executing [number2c...@outgoing_voipvoice:1]
Dial(Console/dsp, IAX2/ my-iax-provider/number2call) in new stack
[Apr 18 19:39:42] DEBUG[32645]: chan_iax2.c:4000 create_addr:
prepending 40 to prefs
-- Called my-iax-provider/number2call
-- Call accepted by 62.149.202.150 (format ilbc)
-- Format for call is ilbc
-- IAX2/my-iax-provider-361 is circuit-busy
-- Hungup 'IAX2/my-iax-provider-361'
== Everyone is busy/congested at this time (1:0/1/0)
Have you got any idea?
Thanks to all in advance
--
/*/
nik600
http://www.kumbe.it
--
/*/
nik600
http://www.kumbe.it
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[asterisk-users] problems originating an outgoing IAX2 call
Dear all
i'm trying to originate an outgoing call with the command originate,
from Asterisk's CLI i'm typing:
CLI originate IAX2/my-iax-provider/number2call application wait 10
[Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr:
prepending 40 to prefs
-- Call accepted by 62.149.202.150 (format ilbc)
-- Format for call is ilbc
-- Hungup 'IAX2/my-iax-provider-5647'
... and nothing happend the hangup is given after 3-4 seconds of the command
But, if i try to call a dialplan extenstion from a local IAX user the
call works properly
[outgoing_voipvoice]
exten = _X.,1,Dial(IAX2/my-iax-provider/${EXTEN})
-- Accepting AUTHENTICATED call from 82.56.46.69:
requested format = gsm,
requested prefs = (),
actual format = gsm,
host prefs = (),
priority = mine
-- Executing [number2c...@outgoing_voipvoice:1]
Dial(IAX2/localuser-3519, IAX2/my-iax-provider/number2call) in new
stack
[Apr 18 19:34:22] DEBUG[32577]: chan_iax2.c:4000 create_addr:
prepending 2 to prefs
-- Called my-iax-provider/number2call
-- Call accepted by 62.149.202.150 (format ilbc)
-- Format for call is ilbc
-- IAX2/my-iax-provider-25 is making progress passing it to
IAX2/localuser-3519
-- IAX2/my-iax-provider-25 is ringing
-- IAX2/my-iax-provider-25 is making progress passing it to
IAX2/localuser-3519
-- IAX2my-iax-provider-25 stopped sounds
-- IAX2/my-iax-provider-25 answered IAX2/localuser-3519
-- Operating with different codecs 2[0x2 (gsm)] 1024[0x400 (ilbc)]
, can't native bridge...
-- Hungup 'IAX2/my-iax-provider-25'
== Spawn extension (outgoing_voipvoice, number2call, 1) exited
non-zero on 'IAX2/localuser-3519'
-- Hungup 'IAX2/localuser-3519'
I'm having the same problem using the dial from console:
CLI console dial number2c...@outgoing_voipvoice
-- Executing [number2c...@outgoing_voipvoice:1]
Dial(Console/dsp, IAX2/ my-iax-provider/number2call) in new stack
[Apr 18 19:39:42] DEBUG[32645]: chan_iax2.c:4000 create_addr:
prepending 40 to prefs
-- Called my-iax-provider/number2call
-- Call accepted by 62.149.202.150 (format ilbc)
-- Format for call is ilbc
-- IAX2/my-iax-provider-361 is circuit-busy
-- Hungup 'IAX2/my-iax-provider-361'
== Everyone is busy/congested at this time (1:0/1/0)
Have you got any idea?
Thanks to all in advance
--
/*/
nik600
http://www.kumbe.it
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Live Audio Streaming- From Aux interface-Online resource
Many thanks Jonathan! On Wed, Mar 31, 2010 at 10:29 AM, cov...@ccs.covici.com wrote: What is the significance of /dev/fd/3 where does it come from? I'ts the file descriptor 3 for the EAGI process, wich contains the audio. -- /*/ nik600 http://www.kumbe.it -- _ -- 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] Live Audio Streaming- From Aux interface-Online resource
On Fri, Mar 19, 2010 at 2:30 PM, Jonathan Addleman j...@redowl.ca wrote: If that doesn't work for some reason (In my case, I needed to stream through a flash applet on a web page, so it needed to be an mp3 stream), you can use an eagi that pipes through an encoder and then to your streaming software. In my case, I piped the audio through ffmpeg and then to ezstream which sent it to icecast. -- Jon-o Addleman - http://www.redowl.ca i'm looking for that, can you kindly give me a more detailed example? I was trying to record a call usng Mixmonitor and then convert it using ffmpeg but the recording file is continuosly growing and ffmpeg ends the conversion before of the call completion. If you can give me a practical example i'll appreciate it a lot. Bye -- /*/ nik600 http://www.kumbe.it -- _ -- 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] distribuited ACD on many asterisk nodes
Dear All i'm planning to develop for a customer a particular implementation of Asterisk. The aim of the project is to share different users between different Asterisk inbound call center . I'm planning to have a sync for some of the QueueMemberStatus informations between all the nodes, then a particular (external) ACD algorithm will decide to transfer a parked call to the final user. I want to trigger an action when an event of type QueueMemberStatus is detected on the manager socket, and then propagate this information to the other Asterisk nodes using some XMPP features or something else. This architecture allows to share users between different call center without having a complete replication of all the nodes (each node can decide how much resources give to the cluster of call center). So each node can have its own configuration and requires only a manager access to share users information and thansfer call. Do you know if there is something similar somewhere ? Maybe Asterisk has already some magic sauce to do that ? ;-) Thanks to all -- /*/ nik600 http://www.kumbe.it -- _ -- 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] usage of manager events to create custom reports
Dear all due to some custom requirements we are planning to use the manager events for creating some custom reports. I've enabled cdr_manager, then in manager.conf i've enabled timestampevents = yes and in queue.conf eventmemberstatus = yes. I know that these settings can generate a lot of manager events but i'm planning to have a very simple application on the Asterisk server that keep all that events from the manager socket and put them into a separate file for each call. To decide when to write the call file i'm planning to wait for the Hangup event). So the call-flow in the events listener will be: 1) new event detected 2) check if the event has an Uniqueid information 3) push the event into a stack reserved for Uniqueid 4) if the event if Hangup write the information of the stack reserved fro Uniqueid and then free memory I'm planning to write this in php, i think that this code is very light to be run even after a lot of events because i free memory after the conclusion of each call. Then (on a separate server) there will be a re-processing of the file extracting all the information required from a call. I'm writing to you just to know: - what do you think about this kind of approach - if someone else has done something similar and wants to share his experience - how much is affordable the events generation excpecially in system with a high load Thanks to all for any contribute. Hi -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to announce the agent answering in a queue to the caller
I've tested and confirm that the AGI script can do that. i had to enable setinterfacevar=yes in the queue conf and then can read the MEMBERINTERFACE channel variable. Just because it can be useful for someone else. On Fri, Oct 23, 2009 at 9:44 PM, nik600 nik...@gmail.com wrote: Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the call? After the AGI script the call is linked with the operator even if there is an Answer into the AGI? Thanks to all -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to announce the agent answering in a queue to the caller
Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the call? After the AGI script the call is linked with the operator even if there is an Answer into the AGI? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrond DTMF detection on Zap channel
After a lot of debugging i have reproduced the error and the behaviour look me very strage: i've tried to change dtmfthereresold, vpmdtmfsupport and other kernel module settings without noting any significative change. But what i've notice (recording all the IVR calls and then listening the registration of the call) is that DTMF tones are not recognized by the system when the DTMF tone is clearly listenable in the audio recording!! Riassuming: good quality in voice and very low quality in the audio DTMF detected: the DTMF tone is recognized, is logged in che console (i've enabled dtmf log in full and console) and correctly detected by the AGI script good quality in voice and good quality in the audio DTMF detected: the DTMF tone is NOT recognized anything is logged in the console and the AGI script goes in timeout I've also upgraded asterisk to asterisk-1.4.26.2 dahdi-linux-complete-2.2.0.2 libpri-1.4.10.1 Any idea? On Tue, Oct 13, 2009 at 11:51 PM, nik600 nik...@gmail.com wrote: for disabling the hardware DTMF you intend to recompile zaptel with vpmdtmfsupport=0? Thanks On Fri, Oct 9, 2009 at 6:54 PM, C F shma...@gmail.com wrote: are you using chan_local? -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrond DTMF detection on Zap channel
for disabling the hardware DTMF you intend to recompile zaptel with vpmdtmfsupport=0? Thanks On Fri, Oct 9, 2009 at 6:54 PM, C F shma...@gmail.com wrote: are you using chan_local? try disabling the hardware DTMF. Sent using my wired Blueberry. On 10/9/09, nik600 nik...@gmail.com wrote: Dear all i have a TE205P connected to an Asterisk 1.2.18. Yes i know, the version is old but since now the system was stable and i don't have the necessity of an upgrade. The system provide an IVR service that: 1) receive the call 2) verify the queue length 3) hangup if queue length is 1 4) put the call in the queue othervise Then, there is an AGI php script that 1) verify the queue 2) wait 5 seconds if the queue is empty 3) pick-up a call from the queue and transfer it to an extension othervise Finally, the extension lanuch another AGI php script that requires some DTMF tone to the user to perform some actions. This system is working properly since 2006. Well, the problem during last days is that it seems that sometimes the DTMF recognition doesn't work, in the debug i get: AGI Tx 200 result=0 But users complains to me because they assure to have digited something different than 0. The problem seems to be reproducible when the system is loaded (i don't have information on the SO but we receive abut 2500 calls per hour each call is very short because usually it is hangup after a very short time, as the queue length is very often 1) It's not an AGI application problem as i get the wrong dtmf tone directly from Asterisk. It's not a phone problem as the same phone may retry and then it works. Is it possible to relate it with the load of the server? Can you suggest me something? Thansk -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] live audio streaming using monitor, mixmonitor or chanspy
Hi to all, is it possible to setup a live audio streaming in Asterisk using for source monitor, mixmonitor or chanspy? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrond DTMF detection on Zap channel
I'm using Zap, not chan_local i've tried to record the call and have seen that the audio DTMF toned received is very poor, i've tried to put relaxdtmf=yes in zapata.conf and increare rxgain and txgain from 0 to 5 but it doesn't seems to be much better. Is there something else to do? Thanks On Fri, Oct 9, 2009 at 6:54 PM, C F shma...@gmail.com wrote: are you using chan_local? try disabling the hardware DTMF. Sent using my wired Blueberry. On 10/9/09, nik600 nik...@gmail.com wrote: Dear all i have a TE205P connected to an Asterisk 1.2.18. Yes i know, the version is old but since now the system was stable and i don't have the necessity of an upgrade. The system provide an IVR service that: 1) receive the call 2) verify the queue length 3) hangup if queue length is 1 4) put the call in the queue othervise Then, there is an AGI php script that 1) verify the queue 2) wait 5 seconds if the queue is empty 3) pick-up a call from the queue and transfer it to an extension othervise Finally, the extension lanuch another AGI php script that requires some DTMF tone to the user to perform some actions. This system is working properly since 2006. Well, the problem during last days is that it seems that sometimes the DTMF recognition doesn't work, in the debug i get: AGI Tx 200 result=0 But users complains to me because they assure to have digited something different than 0. The problem seems to be reproducible when the system is loaded (i don't have information on the SO but we receive abut 2500 calls per hour each call is very short because usually it is hangup after a very short time, as the queue length is very often 1) It's not an AGI application problem as i get the wrong dtmf tone directly from Asterisk. It's not a phone problem as the same phone may retry and then it works. Is it possible to relate it with the load of the server? Can you suggest me something? Thansk -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] wrond DTMF detection on Zap channel
Dear all i have a TE205P connected to an Asterisk 1.2.18. Yes i know, the version is old but since now the system was stable and i don't have the necessity of an upgrade. The system provide an IVR service that: 1) receive the call 2) verify the queue length 3) hangup if queue length is 1 4) put the call in the queue othervise Then, there is an AGI php script that 1) verify the queue 2) wait 5 seconds if the queue is empty 3) pick-up a call from the queue and transfer it to an extension othervise Finally, the extension lanuch another AGI php script that requires some DTMF tone to the user to perform some actions. This system is working properly since 2006. Well, the problem during last days is that it seems that sometimes the DTMF recognition doesn't work, in the debug i get: AGI Tx 200 result=0 But users complains to me because they assure to have digited something different than 0. The problem seems to be reproducible when the system is loaded (i don't have information on the SO but we receive abut 2500 calls per hour each call is very short because usually it is hangup after a very short time, as the queue length is very often 1) It's not an AGI application problem as i get the wrong dtmf tone directly from Asterisk. It's not a phone problem as the same phone may retry and then it works. Is it possible to relate it with the load of the server? Can you suggest me something? Thansk -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] put some IVR into a queue after the call queuing
any interest in it? I'm evauating to add this feature but before to do that i'd like to know if there is some other approach that can avoid some developement. Regards On Wed, Sep 30, 2009 at 12:48 PM, nik600 nik...@gmail.com wrote: Dear all is it possible to handle a queue using a programmed IVR? As i understood, is possible to: - do some IVR in the dialplan BEFORE to queue the call - put a timeout to exit from the call and then do some IVR in the dialplan - intercept a single dialtone to exit the queue and performe some IVR in the dialplan (context setting in the queue) I've tested these things but in each case if i re-queue the call thi queue_log file reports the wrong total queued time. I'm wondering if is possible to bluild a script like that: 1) queue the call 2) after x seconds prompt message A 3) after y seconds prompt message B 4) after z seconds prompt message C 5) after t seconds prompt message Z with DTMF options 1,2,3 if option is 1 = remain in queue if option is 2 = ask the user to be recalled if option is 3 = transfer to In each moment (1,2,3,4,5) if a member queue gets available the call is routed to him. I belive that the only thing to do that is to do something like: 1) Queue A ... timeount 2) Queue B ... timeout 3) Queue C ...Timeout 4) Queue D ...periodic-announce - context set to xxx [xxx] 1,1,Queue D 2,1,Goto (.IVR to be recalled) 3,1,Goto ( transfer) And then manually match information between unique ID and queue_log to consider info on queue A,B,C,D, as a single queue. Or is there some magic sauce to specify an IVR script that is executed when a call is in a queue? Thanks -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] put some IVR into a queue after the call queuing
Dear all is it possible to handle a queue using a programmed IVR? As i understood, is possible to: - do some IVR in the dialplan BEFORE to queue the call - put a timeout to exit from the call and then do some IVR in the dialplan - intercept a single dialtone to exit the queue and performe some IVR in the dialplan (context setting in the queue) I've tested these things but in each case if i re-queue the call thi queue_log file reports the wrong total queued time. I'm wondering if is possible to bluild a script like that: 1) queue the call 2) after x seconds prompt message A 3) after y seconds prompt message B 4) after z seconds prompt message C 5) after t seconds prompt message Z with DTMF options 1,2,3 if option is 1 = remain in queue if option is 2 = ask the user to be recalled if option is 3 = transfer to In each moment (1,2,3,4,5) if a member queue gets available the call is routed to him. I belive that the only thing to do that is to do something like: 1) Queue A ... timeount 2) Queue B ... timeout 3) Queue C ...Timeout 4) Queue D ...periodic-announce - context set to xxx [xxx] 1,1,Queue D 2,1,Goto (.IVR to be recalled) 3,1,Goto ( transfer) And then manually match information between unique ID and queue_log to consider info on queue A,B,C,D, as a single queue. Or is there some magic sauce to specify an IVR script that is executed when a call is in a queue? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] the correct way to setup a transfer with REFER in SIP
Hi to all excuse me but i don't understand what is the correct configuration needed to setup a transfer with REFER in SIP. I've tried the transfer() application, but i've experienced some problem, i can't reproduce the error in a clear debug environment but randomly the call crash before to be transferred to the final peer. on the wiki (http://www.voip-info.org/wiki/view/Asterisk+cmd+Transfer) it is reported as a partial implementation of the REFER functionality. I've tried both atxfer and blindxfer in features.conf but it seems that asterisk make a simple Dial between the two peers. I've also taked a look at ChannelRedirect(channel|[[context|]extension|]priority) but it doesn't seem to be what i need. This is my scenario: A is a SIP Phone registered on the SIP PBX test B is a SIP Phone registered on the SIP PBX test Asterisk is registered on the SIP PBX test with the user C D is a SIP Phone registered on Asterisk. 1) A dial C 2) C (that is Asterisk) execute the dialpan and dial D 3) A and D talks directly as the native bridging is enabled by canreinvite=yes and the codecs are compatible 4) D transfer the call to B What is the configuration needed for the 4th action? My aim is to make a REFER to b...@test and free completely Asterisk. Thanks to all in advance, bye. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with transfer application (REFER)
Hi Giorgio i've tried to upgrade to 1.4.25.1 but still have the same problem. It seems that the problem is that the call is hangup by asterisk before the connection with the transferred peer, but i haven't already been able to reproduce it in a clear debug environment. Is someone can helps... Bye On Fri, Jun 12, 2009 at 2:45 PM, Giorgio Incantalupogincantal...@fgasoftware.com wrote: Hi nik600, I had some trouble transferring calls with that version of Asterisk even if I used the normal transfer via features.conf. Upgrading to 1.4.24 helped a bit (even if not completely). My advice is to upgrade to 1.4.24 or the latest. Giorgio -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with transfer application (REFER)
I'm experiencing some problem using the transfer() application,expecially when a call in received from a queue. I'm using Asterisk 1.4.22.1 This is my scenario: ; this is the piece of code in extensions.conf that place the call in the queue when is called exten = ,1,Answer exten = ,n,Queue(2000|t) ;this is the piece of code that calls the user test when is called exten = ,1,Dial(SIP/test) ; this is the piece of code that transfer the call using REFER exten = ,1,Transfer(SIP/endpo...@x.y.z.t) Calling the call is placed on the queue, and then answered from a member (SIP/test), when the member try to transfer the call to the call ends with an error every time. Calling the call is placed directly to the user SIP/test, when the user try to transfer the call to SOMETIMES the call ends with an error. Sometimes asterisk says: Auto fallthrough, channel 'SIP/xx' status is 'ANSWER' and sometimes it says Auto fallthrough, channel 'SIP/xx' status is 'UNKNOWN' Can you help me to guess the problem? I've read that the REFER implementation in the transfer application is not complete, is it true? Is there any procedure / configuration to use a complete and stable implementation of the REFER functionality? Thanks to all in advance -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: add a new queue strategy: SBR
Thanks, this is interesting. I'm still looking with a customer on a possible implementation of sbr, this is my proposal: Example of skill.conf [default] ; ; STATIC OR DYNAMIC DEFINITION ; ;skillpath=/etc/asterisk/skills.xml skillpath=http://x.x.x.x/skillgenerator.php ; STATIC DEFINITION [SIP/200] sbr_theme=,1 sbr_theme=,1 [SIP/201] sbr_theme=,1 sbr_theme=,1 * Example of XML file located in /etc/asterisk/skills.xml / or generated by http://x.x.x.x/skillgenerator.php skills member interface=default skill theme=z1/skill skill theme=y2/skill /member member interface=SIP/200 skill theme=z2/skill skill theme=y1/skill /member member interface=SIP/300 skill theme=y1/skill skill theme=x2/skill /member /skills * you can set some variables in the channel before to queue it: QUEUE_SBR_THEME_z QUEUE_SBR_THEME_y QUEUE_SBR_THEME_x you can also set in queues.conf the theme for each queue [queueA] sbr_theme=z sbr_theme=y sbr_theme=x On Sun, Apr 5, 2009 at 3:57 PM, Florian Hackenberger f.hackenber...@chello.at wrote: On Sunday 08 March 2009 17:11:33 nik600 wrote: Hi to all isn't there any plan to add the Skills Based Routing strategy in queues.conf? I think that it will be enough to add an int skill to the struct member and then order the member by skill desc. Is it enough to add this type of strategy in calc_metric in app_queue.c ? Hi! I have written a patch implementing skill based routing for asterisk 1.4.17 (can be ported to later versions quite easily). It works like this: You define a database table which stores the skills: columns: membername, skillname, skill_level You set the strategy to skill based and set a variable for each incoming call which specifies which skills to take into account, the weight of the skill and the minimum level (optional). When selecting agents to ring, asterisk picks the agents according to the highest value of weighted skills (skill level multiplied by skill weight for all skills taken into account for that particular call). If an agent does not satisfy the minimum, this agent does not ring at all. You can for example use the minimum to make sure only agents speaking a particular language get a call which requires that language. The implementation is finished and we are currently testing it. Unfortunately I'm quite busy at the moment and it may take about 2 months before I can take the time to release the code. Unless someone hires me as a consultant to work on it. Cheers, Florian -- DI Florian Hackenberger flor...@hackenberger.at www.hackenberger.at -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] what can we do with lost voice packet on a congestioned VPN?
Hi to all in a scenario where: - the bandwith is shared with other traffic (HTTP,VPN,ecc) - the PBX is on a remote VPN peer - due to many reasons Qos is not usable There is a IAX trunk between 2 Asterisk 1.4 i've tried different codecs (ulaw,alaw,gsm) but the main problem still remain the same: too many voice packet get lost. The main problem is surely on the network, but the strange thing is that on the same network there is an H323 trunk from an Alcatel and a Cisco CCM (using g711 codec) and in that case the voice isn't so bad! i've tried to enable jitterbuffer but i can't notice some difference. Is there something else that i can do? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call
I've seen that the CDR manager and i think that it can be enough for my needs, with the timestamp=yes action. I think that it wouldn't be too much difficult to set in the manager_event function (main/manager.c) a condition that if is set events_on_db=yes in the manager.conf it store the information in a db. But, the question is: is manager.conf the correct place to set this kind of configuration? At the moment, i will just set up a manager connection (socket), save results on a file and then (each 5/10 minutes) parse the file and store information on db. I don't want to introduce too much delay to work on db in real time so using a file will be more faster. But i also think that in some situation the possibility to store events directly into db will be more useful, at this point, this feature is independent from the CDR. What do you think about that? -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] log to cdr each dialpan action, not only one record for each call
Hi to all. What can i do if a customer needs to log in the CDR all the dialpan actions related to a call? I mean, not only the lastapp e the lastdata but all the dialpan actions! I know that the actual CDR system store one record for each call (and for billing purposes this can be correct) but in some cases the approach needed is something similar to the queue_log. I know that exists ResetCDR and ForkCDR but they don't do what i need, expecially because they fill-in lastdata and lastapp with ResetCDR So, what can i do? Is it better to do some customization to generate a CDR event on each dialplan step or is better to parse the logfile and extract the information needed? I'm using Asterisk 1.4.23.1 TIA -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call
On Thu, Mar 12, 2009 at 8:13 PM, Matt Riddell li...@venturevoip.com wrote: On 13/03/2009 8:02 a.m., nik600 wrote: Hi to all. What can i do if a customer needs to log in the CDR all the dialpan actions related to a call? I mean, not only the lastapp e the lastdata but all the dialpan actions! I know that the actual CDR system store one record for each call (and for billing purposes this can be correct) but in some cases the approach needed is something similar to the queue_log. I know that exists ResetCDR and ForkCDR but they don't do what i need, expecially because they fill-in lastdata and lastapp with ResetCDR So, what can i do? Use the Asterisk Manager with UserEvent? -- Kind Regards, UserEvent can be useful, but i have to place it into the dialplan in many points. With a large dialplan it's a problem. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call
On Thu, Mar 12, 2009 at 8:44 PM, Steve Murphy m...@parsetree.com wrote: My current thinking is to specify exactly which app invocations you want to track; those involved with dialing would be automatically tracked. Or time groups of invocations via forcing a leg-split via a simple dialplan application call... well, this will be surely the best. I'll read the documentation and let you know, thanks. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call
On Thu, Mar 12, 2009 at 9:22 PM, BJ Weschke bwesc...@gmail.com wrote: We generated a patch for a client probably about a year ago against the 1.4 branch that logged apps for each call, params, and exit status codes into a separate file. Like others have said, it generates a tremendous amount of data and probably does impact performance on very high load servers, but it was very useful to determine EXACTLY what happened with a given call. You know, sometimes the information is more important that the space required to store it. It depends due to the client needs. If it's possible, can you tell me where you have to place the code to log when an app is called? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: add a new queue strategy: SBR
On Tue, Mar 10, 2009 at 4:01 AM, James Sneeringer jsnee...@gmail.com wrote: On Mon, Mar 9, 2009 at 5:44 PM, nik600 nik...@gmail.com wrote: Thanks, i've tested and it works (1.4.23.1). Just 2 questions: 1) this approach seems to be an hack and not the implementation of a feature is it really used in corporate solutions? 2) using queue show 001 i can't see the ringing status, is that correct (In Use, Not in Use,Paused works now properly)? I've never really noticed the lack of a ringing status. Our queue setup has just worked, so I usually only have to use queue show when there's a problem. I do know that the AMI reports the ringing status. The Local/n solution has the added problem of not handling attended transfers correctly. When using a Local channel with the /n flag, if an agent performs an attended or SIP transfer, or does a 3-way call on their own phone and then hangs up, Queue() will still consider the agent In Use until the original transferred call is hung up. Maybe polling the device state using the SIP channel would be better, but as you told me this feature is available only on 1.6.x. It was backported to 1.4.19, but the patch no longer applies cleanly to newer versions. There were some locking changes just after that version. If you want to give it a try, I found it at: http://ftp.iq-labs.net/state_interface-1.4/ Then there's this: http://reviewboard.digium.com/r/116/ The corresponding func_devstate has also been backported, but it's pretty old: http://svncommunity.digium.com/view/russell/asterisk-1.4/func_devstate-1.4/ I got the 1.4.19 backport to compile against a 1.4.20.1 codebase, but Asterisk would core as soon as app_queue.so loaded, so clearly I didn't quite get it right. I eventually punted and changed my dynamic queues to just use the actual SIP/x channel names. It's been working fine for over a year now. thanks for these explanation, at this point i think that the better thing is to use the SIP/ channel and do something else on a third party system to store an additional information about the agent using that phone, it's more stable and clear on asterisk side. Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: add a new queue strategy: SBR
On Mon, Mar 9, 2009 at 3:16 PM, James Sneeringer jsnee...@gmail.com wrote:
If you are using dynamic queues with Local channels (as described in
doc/queues-with-callback-members.txt in the Asterisk source), you can
also optionally implement this functionality directly in the dialplan.
This has the added benefit of allowing you to choose on a per-agent
basis who is eligible for autopause.
-James
thanks for your reply, infact i've implemented the agents in the
dialplan as explained in queues-with-callback-members.txt but this
approach doesn't manage the status of the agent!
I can add / remove / pause / unpause the member interface but what
about the in use status?
The extension in the context will be every time Not in use or shall
i implement hints?
Here there is a piece of my extensions.conf:
[default]
; login procedure for queue 001
exten = _001,1,Answer
exten = _001,n,AddQueueMember(001,Local/${EXTEN:3...@agents)
exten = _001,n,Set(DB(agents/${EXTEN:3})=SIP/${CALLERID(num)})
[agents]
exten = _,hint,${DB(agents/${EXTEN})}
exten = _,1,Dial(${DB(agents/${EXTEN})})
and there isn't an agent but only an extension on a queue.
What do you think about that?
maybe i should open a new post but i think that this kind of approach
isn't much better than the callback functionality, what do you think
about that?
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Re: [asterisk-users] Fwd: add a new queue strategy: SBR
On Mon, Mar 9, 2009 at 8:39 PM, Mark Michelson mmichel...@digium.com wrote:
The reason that the member always appears to be not in use is that local
channels are optimized away once they are bridged to their real destination.
The
result of this is that since the channel does not exist anymore, the device
state engine interprets the interface to be not in use anymore. One way to
handle this issue is to change your AddQueueMember call to use
Local/${EXTEN:3...@agents/n (notice the /n at the end). The /n tells the local
channel driver to not attempt to optimize the local channel away.
If you are using Asterisk version 1.6.0 or above, an even better method would
be
to specify a second interface to poll for device state when adding the queue
member. Assuming that the member at Local/${EXTEN:3...@agents will always call
SIP/${EXTEN:3}, then what you are really interested in when receiving device
state notifications is the SIP channel, not the local channel. You can specify
this second state interface in AddQueueMember like so:
AddQueueMember(001,Local/${EXTEN:3...@agentsSIP/${EXTEN:3})
Doing this will tell app_queue to use the SIP channel's device state to
determine if the member is available, but when it comes time to call the
agent,
it will actually place the call to the local channel provided.
Mark Michelson
Thanks, i've tested and it works (1.4.23.1).
Just 2 questions:
1) this approach seems to be an hack and not the implementation of a
feature is it really used in corporate solutions?
2) using queue show 001 i can't see the ringing status, is that
correct (In Use, Not in Use,Paused works now properly)?
Maybe polling the device state using the SIP channel would be better,
but as you told me this feature is available only on 1.6.x.
Thanks for your time.
Bye
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[asterisk-users] Fwd: add a new queue strategy: SBR
Hi., do you think that sbr policy in queue strategy will be useful? Bye -- Forwarded message -- From: nik600 nik...@gmail.com Date: Sat, 7 Mar 2009 15:21:14 +0100 Subject: add a new queue strategy: SBR To: Asterisk Developers Mailing List asterisk-...@lists.digium.com Hi to all isn't there any plan to add the Skills Based Routing strategy in queues.conf? I think that it will be enough to add an int skill to the struct member and then order the member by skill desc. Is it enough to add this type of strategy in calc_metric in app_queue.c ? thanks -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: add a new queue strategy: SBR
but priority are se to the call, not to the agent! or am i wrong? On Sun, Mar 8, 2009 at 5:32 PM, David fire ddf...@gmail.com wrote: the queue already have prioritys. David 2009/3/8 nik600 nik...@gmail.com Hi., do you think that sbr policy in queue strategy will be useful? Bye -- Forwarded message -- From: nik600 nik...@gmail.com Date: Sat, 7 Mar 2009 15:21:14 +0100 Subject: add a new queue strategy: SBR To: Asterisk Developers Mailing List asterisk-...@lists.digium.com Hi to all isn't there any plan to add the Skills Based Routing strategy in queues.conf? I think that it will be enough to add an int skill to the struct member and then order the member by skill desc. Is it enough to add this type of strategy in calc_metric in app_queue.c ? thanks -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] put the hostname of asterisk in the callerid uri to avoid NAT problems
On Wed, Feb 11, 2009 at 2:49 AM, Steven J. Douglas stev...@moij.biz wrote: Hi, Have you tried using externip in your sip.conf? By setting the correct localnet, any SIP packets that goes elsewhere will use the value in externip. This might solve your problem. Regards, Steve yes i've done it. The rtp traffic is redirect correctly but the SIP INVITE contains the ip of the lan and not of the nat. I'll try with SipAddHeader and then let you know... thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] put the hostname of asterisk in the callerid uri to avoid NAT problems
On Sat, Feb 7, 2009 at 8:31 AM, nik600 nik...@gmail.com wrote: hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk is behind NAT - Asterisk ip is 10.10.10.2 - SIP user view Asterisk as 10.10.15.2 (Asterisk is behind NAT) When the CCM calls the SIP user the call works perfectly. The problem is that the SIP user receives the call with this uri: sip:x...@10.10.10.2 The call works properly and the audio goes in both directios, BUT if the SIP user does a redial (after the hangup) the call is forwarded to x...@10.10.10.2 that is the wrong address. I've tried to force SIP_HEADER(CONTACT) in the dialplan with a Set but it seems that i can't due to security reason. Is it possible to avoid this problem? Thanks -- /*/ nik600 http://www.kumbe.it Do you think that is a bug or a miss configuration, or simply is not possible to avoid that because it is hard-coded? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] put the hostname of asterisk in the callerid uri to avoid NAT problems
hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk is behind NAT - Asterisk ip is 10.10.10.2 - SIP user view Asterisk as 10.10.15.2 (Asterisk is behind NAT) When the CCM calls the SIP user the call works perfectly. The problem is that the SIP user receives the call with this uri: sip:x...@10.10.10.2 The call works properly and the audio goes in both directios, BUT if the SIP user does a redial (after the hangup) the call is forwarded to x...@10.10.10.2 that is the wrong address. I've tried to force SIP_HEADER(CONTACT) in the dialplan with a Set but it seems that i can't due to security reason. Is it possible to avoid this problem? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] server sizing for ~ 200 simultaneous call
Hi to all i'm planning the migration of a company on Asterisk, i have planned this scenario: 2 server with * 4 GB RAM * 2 CPU 64 bit dual core * RAID 1 * 2 network interfaces 1000 Mbit/s Each server will have a virtual interface that will be switched from one to the other in case of hardware problem. The question is: can one server with those settings manage up to 200 simultaneous call? The server will receive SIP calls and forward them through a CISCO router. Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with PlayDTMF: no error but no tone
I think to have guess the problem, or maybe the work-around (maybe can be useful for someone). in sip.conf dtmfmode was set to default. I've tried to set to rfc2833,info,inband and auto. info and inband works, auto and rfc2833 not. The strange thing is: auto : Use rfc2833 if offered, inband otherwise. It means that rfc2833 was offered, but doesn't work! Well, info and inband works. Bye On Thu, Jan 22, 2009 at 11:18 AM, nik600 nik...@gmail.com wrote: Is there the possibility to increase the debug of an AJAM command? If DTMF works on channel, and my command is queued successfully, what can be the problem? Thanks On Thu, Jan 15, 2009 at 4:34 PM, nik600 nik...@gmail.com wrote: Hi to all i'm using PlayDTMF with AJAM, after the authentication, i make a request like this: host:8088/asterisk/mxml?action=PlayDTMFChannel=SIP/200-sdadsadkioahDigit=1 the result is: ajax-response response type='object' id='unknown'generic response='Success' message='DTMF successfully queued' //response /ajax-response But i can't heard nothing on the channel, i've tried to send the tone both on channel and link, but with no results. If i use normal dtmf from keyboards they works properly. What can i check? Thanks -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with PlayDTMF: no error but no tone
Is there the possibility to increase the debug of an AJAM command? If DTMF works on channel, and my command is queued successfully, what can be the problem? Thanks On Thu, Jan 15, 2009 at 4:34 PM, nik600 nik...@gmail.com wrote: Hi to all i'm using PlayDTMF with AJAM, after the authentication, i make a request like this: host:8088/asterisk/mxml?action=PlayDTMFChannel=SIP/200-sdadsadkioahDigit=1 the result is: ajax-response response type='object' id='unknown'generic response='Success' message='DTMF successfully queued' //response /ajax-response But i can't heard nothing on the channel, i've tried to send the tone both on channel and link, but with no results. If i use normal dtmf from keyboards they works properly. What can i check? Thanks -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with PlayDTMF: no error but no tone
Hi to all i'm using PlayDTMF with AJAM, after the authentication, i make a request like this: host:8088/asterisk/mxml?action=PlayDTMFChannel=SIP/200-sdadsadkioahDigit=1 the result is: ajax-response response type='object' id='unknown'generic response='Success' message='DTMF successfully queued' //response /ajax-response But i can't heard nothing on the channel, i've tried to send the tone both on channel and link, but with no results. If i use normal dtmf from keyboards they works properly. What can i check? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with dahdi and meetme
Hi to all. I'm trying to use meetme on asterisk 1.4.22.1. On a debian i've compiled (as i need h323 support) openh323_v1_18_0 pwlib_v1_10_0 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-1.4.22.1 All works fine, dahdi status is: asterik:/data/programmi# /etc/init.d/dahdi status ### Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) asterik:/data/programmi# lsmod | grep dah dahdi_dummy 5224 0 dahdi 186280 1 dahdi_dummy crc_ccitt 2240 1 dahdi rtc12372 1 dahdi_dummy if i start asterisk i get: asterik:/data/programmi# asterisk -cvvv Asterisk 1.4.22.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log [Jan 12 13:38:23] ERROR[12617]: asterisk.c:3036 main: Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection. You have options: 1. You only have to compile DAHDI support into Asterisk if you need it. One option is to recompile without DAHDI support. 2. You only have to load DAHDI drivers if you want to take advantage of DAHDI services. One option is to unload DAHDI modules if you don't need them. 3. If you need DAHDI services, you must correctly configure DAHDI. Where am i wrong? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with dahdi and meetme
PS: asterisk is compiled with dahdi support On Mon, Jan 12, 2009 at 1:39 PM, nik600 nik...@gmail.com wrote: Hi to all. I'm trying to use meetme on asterisk 1.4.22.1. On a debian i've compiled (as i need h323 support) openh323_v1_18_0 pwlib_v1_10_0 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-1.4.22.1 All works fine, dahdi status is: asterik:/data/programmi# /etc/init.d/dahdi status ### Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) asterik:/data/programmi# lsmod | grep dah dahdi_dummy 5224 0 dahdi 186280 1 dahdi_dummy crc_ccitt 2240 1 dahdi rtc12372 1 dahdi_dummy if i start asterisk i get: asterik:/data/programmi# asterisk -cvvv Asterisk 1.4.22.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log [Jan 12 13:38:23] ERROR[12617]: asterisk.c:3036 main: Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection. You have options: 1. You only have to compile DAHDI support into Asterisk if you need it. One option is to recompile without DAHDI support. 2. You only have to load DAHDI drivers if you want to take advantage of DAHDI services. One option is to unload DAHDI modules if you don't need them. 3. If you need DAHDI services, you must correctly configure DAHDI. Where am i wrong? Thanks -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4 with h323 for debian
hi to all. Do you know if there is an asterisk 1.4 package with h323 support for debian? I've found this http://packages.debian.org/etch/asterisk-h323 but has asterisk 1.2.13. Thanks to all. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] busy-level / busy-limit Asterisk 1.4.22
sorry if i ask it again, but where can i find the patch for enable busy-level/limit in 1.4 ? thanks On Tue, Nov 18, 2008 at 12:09 PM, nik600 nik...@gmail.com wrote: Thanks, is it possibile to retrieve a patch from Asterisk trunk? how? On Tue, Nov 18, 2008 at 11:54 AM, Steve Howes st...@geekinter.net wrote: On 18 Nov 2008, at 10:30, nik600 wrote: the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? http://www.voip-info.org/wiki/view/Asterisk+sip+busy-level ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set the busy signal usign softphones
Ok, i've resolved, the problem was related to the sip type settings.
It must be peer instead of fried.
Bye
On Fri, Jan 2, 2009 at 5:41 PM, nik600 nik...@gmail.com wrote:
Thanks for your reply.
Now, i use devstate too, but it doesn't work (or, maybe i suppose that
it should work differently) when the called user has an outgoing call.
this is my extension.conf:
exten = _XXX,1,ExecIf($[${DEVSTATE(SIP/${EXTEN})} = INUSE],Busy)
exten = _XXX,2,Dial(SIP/${EXTEN})
Now, suppose to have 3 users: 200,201,202.
201 calls 200 (devstate of SIP/200 is NOT_INUSE) - OK
202 calls 200 (devstate of SIP/200 is INUSE) - OK.
BUT, with these scenario:
200 calls 201 (devstate of SIP/201 is NOT_INUSE) - OK
202 calls 200 (devstate of SIP/200 is NOT_INUSE) - ?? why ???.
Thanks to all
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Re: [asterisk-users] how to set the busy signal usign softphones
Thanks for your reply.
Now, i use devstate too, but it doesn't work (or, maybe i suppose that
it should work differently) when the called user has an outgoing call.
this is my extension.conf:
exten = _XXX,1,ExecIf($[${DEVSTATE(SIP/${EXTEN})} = INUSE],Busy)
exten = _XXX,2,Dial(SIP/${EXTEN})
Now, suppose to have 3 users: 200,201,202.
201 calls 200 (devstate of SIP/200 is NOT_INUSE) - OK
202 calls 200 (devstate of SIP/200 is INUSE) - OK.
BUT, with these scenario:
200 calls 201 (devstate of SIP/201 is NOT_INUSE) - OK
202 calls 200 (devstate of SIP/200 is NOT_INUSE) - ?? why ???.
Thanks to all
On Sat, Dec 20, 2008 at 5:50 PM, Yehavi Bourvine
yehavi.bourv...@gmail.com wrote:
I use the dev_state() function to find the status of the called phone. If it
is BUSY then I call the busy() application to signal a busy tone.
Firthermore, I also consult a MySQL table to see whether the user wants
waiting calls or not and decide accordingly.
__Yehavi:
2008/12/20 nik600 nik...@gmail.com
On Sat, Dec 20, 2008 at 2:25 PM, Benoit maver...@maverick.eu.org wrote:
Have you tried to set the call-limit to 10 or 2 for example, i know it's
what's needed for the queue system to detect
busy on sip softphone
Yes, but if i set the call-limit to 2 the user receive more than 1
call (correctly...up to 2 calls), even when he is busy.
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[asterisk-users] how to set the busy signal usign softphones
Hi to all. I'm using Asterisk 1.4 with Sjphone as softphone. My problem is that when a SIP user is busy, he still receive calls from asterisk. I've tried to setup the call-limit preference to 1, but with this kind of configuration the user can't transfer calls, as the system block the 2nd call generated to do the transfer. I've also tried to set the user as friend, limitonpeers = yes and call-limit =1. In that case the work-around works but only when the user is the receiver of the call that makes him busy. If the user is the caller, he still receive a second call. So, isn't there any method to limit the call available for a user to 1 but granting him the possibility to transfer a call? I know that there is the busy-level settings, but i'ts available only in 1.6. Thanks to all in advance. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set the busy signal usign softphones
On Sat, Dec 20, 2008 at 2:25 PM, Benoit maver...@maverick.eu.org wrote: Have you tried to set the call-limit to 10 or 2 for example, i know it's what's needed for the queue system to detect busy on sip softphone Yes, but if i set the call-limit to 2 the user receive more than 1 call (correctly...up to 2 calls), even when he is busy. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] stream a file on a channel using AMI
Hi using AMI, is it possile to stream a file on a specific channel? Thanks to all in advance. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip trunking and call transfer
On Mon, Nov 24, 2008 at 12:14 AM, Raj Jain [EMAIL PROTECTED] wrote: Yes, by using the SIP REFER method. Caller 2 will send a SIP REFER to Caller 1 asking it to talk to Caller 3. This will cause Caller 1 to drop it's session with Caller 2 and send a new INVITE to Caller 3. So, this is how you do it from a SIP protocol perspective. I'm not sure to what extent Asterisk supports this capability. -- Raj Jain ok, thanks for your reply! I'll search about Asterisk SIP referer implementation. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip trunking and call transfer
Maybe my question is not clear or is too stupid? (sorry) Maybe this is already done in SIP trunking? Or (worste case) is impossible to do that? Thanks On Fri, Nov 21, 2008 at 8:53 AM, nik600 [EMAIL PROTECTED] wrote: Hi to all. i-ve got a question: what happen when a call between 2 trunks is transferred to another trunk? For example, suppose that i have 4 trunk A,B,C,D: Caller 1 - Trunk A/B - Caller2 Then Caller 2 transfer to Caller 3 behind Trunk B/C What happend? a) Caller 1 - Trunk A/B - Trunk B/C - Caller3 or b) Caller 1 - Trunk A/C - Caller3 So: is it possible to avoid the scenario a) ? Thanks to all -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip trunking and call transfer
Hi to all. i-ve got a question: what happen when a call between 2 trunks is transferred to another trunk? For example, suppose that i have 4 trunk A,B,C,D: Caller 1 - Trunk A/B - Caller2 Then Caller 2 transfer to Caller 3 behind Trunk B/C What happend? a) Caller 1 - Trunk A/B - Trunk B/C - Caller3 or b) Caller 1 - Trunk A/C - Caller3 So: is it possible to avoid the scenario a) ? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] busy-level / busy-limit Asterisk 1.4.22
Hi to all the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? This is a piece of my sip.conf: [202] type=friend secret=202 host=dynamic; This device registers with us username=202; Username to use when calling this device before registration limitonpeers = yes call-limit = 2 busy-level = 1 The directive busy-level is ignored I've also tried busy-limit but without any result... Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] busy-level / busy-limit Asterisk 1.4.22
Thanks, is it possibile to retrieve a patch from Asterisk trunk? how? On Tue, Nov 18, 2008 at 11:54 AM, Steve Howes [EMAIL PROTECTED] wrote: On 18 Nov 2008, at 10:30, nik600 wrote: the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? http://www.voip-info.org/wiki/view/Asterisk+sip+busy-level ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] view the current calls and their codec
Hi to all. Is possible with the Asterisk 1.4 cli view the current calls and their codec? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] view the current calls and their codec
And if i have an h323 configuration? Thanks On Tue, Nov 11, 2008 at 4:17 PM, Tim Nelson [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] ~]# asterisk -rx 'sip show channels' assuming you want SIP... substitute sip for iax2 if you prefer... Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - nik600 [EMAIL PROTECTED] wrote: Hi to all. Is possible with the Asterisk 1.4 cli view the current calls and their codec? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] view the current calls and their codec
thanks a lot! On Tue, Nov 11, 2008 at 6:06 PM, Tim Nelson [EMAIL PROTECTED] wrote: 'oh323 show channels' I would assume... I don't have a box handy with h323 loaded to verify. Check http://astrecipes.net/index.php?n=89 Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - nik600 [EMAIL PROTECTED] wrote: And if i have an h323 configuration? Thanks On Tue, Nov 11, 2008 at 4:17 PM, Tim Nelson [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] ~]# asterisk -rx 'sip show channels' assuming you want SIP... substitute sip for iax2 if you prefer... Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - nik600 [EMAIL PROTECTED] wrote: Hi to all. Is possible with the Asterisk 1.4 cli view the current calls and their codec? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] skype and Asterisk opensource integration
Hi to all except of some commercial hardware / software gateways, is there any opensource or free project to setup a Skype Account on Asterisk? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with asterisk 1.4.21.1 and h323
Hi to all, i'm experiencing a problem with an h323 trunk between a
Cisco Callmanager 4.2.
I'm using asterisk 1.4.21.1, openh323_v1_18_0, pwlib_v1_10_0
The problem is that sometimes (1 call every 20... but sometimes often)
the call arrives correctly on Call Manager side, and when is answered
after 1-2 seconds Asterisk gives a service unavailable error.
I've noticed enabling h323 trace that when the call is rejectedi i've
got an empty capabilityTable in trace.
When the call works i have:
capabilityTable = 10 entries {
[0]={I
capabilityTableEntryNumber = 1
capability = receiveAudioCapability g7231 {
maxAl_sduAudioFrames = 1
silenceSuppression = TRUE
}CLI
}1*CLI
[1]={I
capabilityTableEntryNumber = 2
capability = receiveAudioCapability g7231 {
maxAl_sduAudioFrames = 1
silenceSuppression = FALSE
}CLI
}1*CLI
[2]={I
capabilityTableEntryNumber = 3
capability = receiveAudioCapability gsmFullRate {
audioUnitSize = 33
comfortNoise = FALSE
scrambled = FALSE
}CLI
}1*CLI
[3]={I
capabilityTableEntryNumber = 4
capability = receiveAudioCapability g711Ulaw64k 20
}1*CLI
[4]={I
capabilityTableEntryNumber = 5
capability = receiveAudioCapability g711Alaw64k 20
}1*CLI
[5]={I
capabilityTableEntryNumber = 6
capability = receiveAudioCapability g729AnnexA 2
}1*CLI
[6]={I
capabilityTableEntryNumber = 7
capability = receiveAudioCapability g729 2
}1*CLI
[7]={I
capabilityTableEntryNumber = 8
capability = receiveUserInputCapability hookflash null
}1*CLI
[8]={I
capabilityTableEntryNumber = 9
capability = receiveRTPAudioTelephonyEventCapability {
dynamicRTPPayloadType = 101
audioTelephoneEvent = 0-16
}CLI
}1*CLI
[9]={I
capabilityTableEntryNumber = 10
capability = receiveUserInputCapability dtmf null
}1*CLI
}k01*CLI
When the call doesn't works i haven't any capabilityTable in trace.
How can i fix that?
My h323.conf is very simple:
[general]
port = 1720
bindaddr = 192.168.1.1
allow=all
tunneling=cisco
[ccm01]
type=peer
host=192.168.1.2
fastStart=no
Thanks to all in advance
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[asterisk-users] OpenH323 and ptlib version for asterisk 1.4.21.1
Hi what version of openh323 and pwlib are suggested for asterisk 1.4.21.1.? Thanks to all -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenH323 and ptlib version for asterisk 1.4.21.1
thanks for your reply. I've installed them but i'm experiencing this problem: i've configured in h323.conf 2 peers: one to an 3.3 CCM Cisco one to an 4.2 CCM Cisco each CCM has the preferred codec set up as G711 ulaw. I can forward calls from a SIP account on asterisk (using Xten-xlite as softphone) to both the peers and talk with their extensions without any problem. I can forward calls from both the peers to Asterisk (and for example place the call in queue or background some sound files) BUT when i try to call from the CCM 3.3 to Asterisk, and then dial from the dialplan a SIP account, when the SIP user accept the call (using Xten-xlite as softphone) asterisk dies with a segmentation fault error. This happend only with CCM 3.3, with 4.2 there is no problem. I've got a backtrace of the error, it seems a codec problem, as the parameter passed to ast_rtp_new_source is null. #0 ast_rtp_new_source (rtp=0x0) at rtp.c:2002 2002 rtp-set_marker_bit = 1; (gdb) bt #0 ast_rtp_new_source (rtp=0x0) at rtp.c:2002 #1 0xb6cfc346 in oh323_indicate (c=0x8205ea0, condition=20, data=0x0, datalen=0) at chan_h323.c:919 #2 0x08081ece in ast_indicate_data (chan=0x8205ea0, condition=20, data=0x0, datalen=0) at channel.c:2372 #3 0x0808698c in ast_channel_bridge (c0=0x8205ea0, c1=0x820acf8, config=0xb60e0de8, fo=0xb60dff38, rc=0xb60dff34) at channel.c:2358 #4 0xb6fad295 in ast_bridge_call (chan=0x8205ea0, peer=0x820acf8, config=0xb60e0de8) at res_features.c:1422 #5 0xb6ae0893 in dial_exec_full (chan=0x8205ea0, data=0xb6ae26fb, peerflags=0xb60e0ea4, continue_exec=0x0) at app_dial.c:1699 #6 0xb6ae1cd2 in dial_exec (chan=0x8205ea0, data=0xb60e2f18) at app_dial.c:1753 #7 0x080c6f36 in pbx_extension_helper (c=0x8205ea0, con=0x0, context=0x8206020 from-h323, exten=0x8206070 54, priority=1, label=0x0, callerid=0x8205830 419, action=E_SPAWN) at pbx.c:537 #8 0x080c8fb5 in __ast_pbx_run (c=0x8205ea0) at pbx.c:2317 #9 0x080c9e7e in pbx_thread (data=0x8205ea0) at pbx.c:2636 #10 0x080f8fab in dummy_start (data=0x8205ce8) at utils.c:895 #11 0xb7f56383 in start_thread () from /lib/libpthread.so.0 #12 0xb731905e in clone () from /lib/libc.so.6 Can someone help me please? Thanks in advance to all On 7/17/08, Patrick [EMAIL PROTECTED] wrote: On Thu, 2008-07-17 at 19:34 +0200, nik600 wrote: Hi what version of openh323 and pwlib are suggested for asterisk 1.4.21.1.? Thanks to all Iirc it is openh323 1.18.0 and pwlib 1.10.1. Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] disable DTMF on a particular channel
Hi to all is it possibile (via AMI or dialplan) to disable the DTMF tone on a particular channel? Thanks in advance -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play sound on a specific channel
i've seen that there is the PlayDTMF command. Bye On Tue, Jun 24, 2008 at 8:37 AM, nik600 [EMAIL PROTECTED] wrote: any idea? On Sat, Jun 14, 2008 at 9:50 AM, nik600 [EMAIL PROTECTED] wrote: Hi to all can i play a sound or a dtmf tone on a specific channel using AMI? Thanks to all -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play sound on a specific channel
any idea? On Sat, Jun 14, 2008 at 9:50 AM, nik600 [EMAIL PROTECTED] wrote: Hi to all can i play a sound or a dtmf tone on a specific channel using AMI? Thanks to all -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] retrieve the status of a sip user using AMI
Hi to all. How can i retrieve the status of a user using the subscription? For example, if i use: exten = 200,hint,SIP/200 exten = 200,1,Dial(SIP/200) After that, how can i retrieve the status of the SIP/200 user using AMI ? Thanks to all in advance -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] play sound on a specific channel
Hi to all can i play a sound or a dtmf tone on a specific channel using AMI? Thanks to all -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] use of AJAM wth high load
Hi to all i'm planning to use AJAM to obtain xml information about queue status, extensions, ecc ecc. Someone of you has some experience about this tool in an enviroment with high load? I'm planning to use it in an installation with 5000 extensions and about 500 simultaneous call. Thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Patch for app_asr.c: DTMF instead of goto
Hi to all
if someone of you is interested on it, i've changed the code of app_asr.c
With these patch you can use the ASR application to play DTMF tones,
so you can have your own AGI application that uses the ASR and manages
the DTMF tones without change the dialplan.
EXAMPLE
exten = 003,1,Ringing
exten = 003,2,Wait(3)
exten = 003,3,Answer
exten = 003,4,ASR(t5000c80l4,100,200:pippo,300:pluto,400:paperino)
exten = 003,5,Read(digito||3)
exten = 003,6,SayDigits(${digito})
exten = 003,7,Wait(30)
The old app_asr will send you to the 200,300 or 400 extension.
With the modified app_asr you will hear (and Asterisk can detects, via
AGI or dialplan) 200,300,400 DTMF tones.
You can find more information here.
http://www.kumbe.it/pagine/dettaglio/34/206.html
Bye
--
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
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Re: [asterisk-users] asterisk virtualization on VMWARESX infrastructure
On Thu, May 22, 2008 at 9:51 PM, Sam Tam [EMAIL PROTECTED] wrote: Why if you have 50 operator then I would even consider using dual server running backup So the idea of using vmware may really be very risky, let alone not talk about performance issue well vmware will not be installed on a single machine, i intend an enterprise SX infrastructure with multiple nodes and auto failover policy. If Asterisk doens't suffer a virtualization, a service virtualized on a solid infrastructure is more scalable and hardware independent -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] upgrade of asterisk .... to what?
Hi to all i'm managing a call center with 20 operators using Asterisk. I'm still using Asterisk 1.2.x as i love his stability. Now, i'm planning to migrate to 1.4.x, but i don't know what version to use! 1.4.20 has been released a few days ago, but now there is 1.4.21. Is there a rule to determine what is beta and what is stable? Thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] upgrade of asterisk .... to what?
On Thu, May 22, 2008 at 12:31 PM, Gordon Henderson [EMAIL PROTECTED] wrote: On Thu, 22 May 2008, nik600 wrote: Hi to all i'm managing a call center with 20 operators using Asterisk. I'm still using Asterisk 1.2.x as i love his stability. Now, i'm planning to migrate to 1.4.x, but i don't know what version to use! 1.4.20 has been released a few days ago, but now there is 1.4.21. Are there any features in 1.4 that you desperately need? If not, then why upgrade? No, i'm just wondering because there is creating a greater difference between my installation and the actual Asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to retrieve sip tag from dialplan
Hi to all is it possible to retrieve the sip tag (server side) of a sip call from the dialplan? Thanks. -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need comments on CRM development / Asterisk Customization
Some times ago i've started these projects: https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker Now i am too busy to update them, but you can use the main logic of ccmanager and the flexibility of reportmaker (you can define your report via xml) to make your own statistic about queues. Bye On Sat, Apr 26, 2008 at 6:12 PM, Alan Lord [EMAIL PROTECTED] wrote: Kashif Naeem wrote: Hello All, A company has two requirements: 1) They are looking to develop its own CRM 2) Second thing is that they want to develop enhancements / new features in Asterisk like Thirdlane. What are your comments about technology to be used. Which one would be most beneficial in future ? PHP, JSP, ASP ? Can anyone suggest existing easy and generic CRM ? As well and Sugar and vtiger (PHP apps) also take a look at ConcursiveSuite (formerly known as CentricCRM) http://www.concursive.com. It has a crappy licence but has good asterisk integration. It's a JSP (Tomcat) application. HTH Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] disable call waiting by default
I've connected some analogic phone to some fxs modules on an analogic card. I want to disable by default the call waiting sound. I know that dialing *70 before to call the call waiting is disabled until the next call, but isn't there a setting or a dialplan command to set up this automatically? Thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed
On Dec 24, 2007 8:07 PM, Darrick Hartman [EMAIL PROTECTED] wrote: Tzafrir Cohen wrote: On Mon, Dec 24, 2007 at 05:11:44PM +0100, nik600 wrote: maybe i've guess the problem! on the same server, i've got a B800P. I've tried to manually remove all isdn module and zaptel modules. After that, i've done modprobe zaptel modprobe opvxa1200 and now the card has been correctly registered! That card was picked up by hisax? I doubt it. My guess is he never rmmod'd zaptel or rebooted the box before trying to modprobe opvxa1200. He was probably running the original zaptel module which didn't know about the new hardware. Darrick no, i've tried to reboot the box... these modules are loaded automatically, i have to remove them manually and reload them, and after that it works. I use slackware 12.0, in rc.modules and rc.local there is any entry but the modules are automatically loaded after each reboot... Thanks to all! -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed
On 12/23/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Dec 23, 2007 at 07:05:37PM +0100, nik600 wrote: Hi i've got an openvox a800p01 with 1 FXO and 4 FSX i've done the following: - downloaded zaptel-1.4.7.1 - downloaded the file opvxa1200.c - copied in zaptel-1.4.7.1/ - edited makefile adding opvxa1200 in the modules and the voice opvxa1200.o : zaptel.h wctdm.h - edited zaptel.sysconfig adding MODULES=$MODULES opvxa1200 # OPENVOXA1200P after that ive done: make clean, make, make install finally, if i do: modprobe opvxa1200 dmesg | tail Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.7.1 Zaptel Echo Canceller: MG2 if i launch ./zapconf cat /proc/zaptel/* [EMAIL PROTECTED]:~# cat /proc/zaptel/ cat: /proc/zaptel/: Is a directory [EMAIL PROTECTED]:~# ls -la /proc/zaptel/ total 0 dr-xr-xr-x 2 root root 0 2007-12-24 14:37 ./ dr-xr-xr-x 76 root root 0 2007-12-23 21:18 ../ It seems that the module doesn't recognize the card!... but the card is recognized by lspci... -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed
maybe i've guess the problem! on the same server, i've got a B800P. I've tried to manually remove all isdn module and zaptel modules. After that, i've done modprobe zaptel modprobe opvxa1200 and now the card has been correctly registered! On Dec 24, 2007 2:32 PM, nik600 [EMAIL PROTECTED] wrote: On 12/23/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Dec 23, 2007 at 07:05:37PM +0100, nik600 wrote: Hi i've got an openvox a800p01 with 1 FXO and 4 FSX i've done the following: - downloaded zaptel-1.4.7.1 - downloaded the file opvxa1200.c - copied in zaptel-1.4.7.1/ - edited makefile adding opvxa1200 in the modules and the voice opvxa1200.o : zaptel.h wctdm.h - edited zaptel.sysconfig adding MODULES=$MODULES opvxa1200 # OPENVOXA1200P after that ive done: make clean, make, make install finally, if i do: modprobe opvxa1200 dmesg | tail Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.7.1 Zaptel Echo Canceller: MG2 if i launch ./zapconf cat /proc/zaptel/* [EMAIL PROTECTED]:~# cat /proc/zaptel/ cat: /proc/zaptel/: Is a directory [EMAIL PROTECTED]:~# ls -la /proc/zaptel/ total 0 dr-xr-xr-x 2 root root 0 2007-12-24 14:37 ./ dr-xr-xr-x 76 root root 0 2007-12-23 21:18 ../ It seems that the module doesn't recognize the card!... but the card is recognized by lspci... -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed
Hi i've got an openvox a800p01 with 1 FXO and 4 FSX i've done the following: - downloaded zaptel-1.4.7.1 - downloaded the file opvxa1200.c - copied in zaptel-1.4.7.1/ - edited makefile adding opvxa1200 in the modules and the voice opvxa1200.o : zaptel.h wctdm.h - edited zaptel.sysconfig adding MODULES=$MODULES opvxa1200 # OPENVOXA1200P after that ive done: make clean, make, make install finally, if i do: modprobe opvxa1200 if i launch ./zapconf the file /etc/zaptel.conf still remains empty, if i force editing the file adding: fxsks=1 fxoks=2 fxoks=3 fxoks=4 fxoks=5 loadzone= it defaultzone = it and do a: ztcfg - i get: Zaptel Version: 1.4.7.1 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) Channel 05: FXO Kewlstart (Default) (Slaves: 05) 5 channels to configure. ZT_CHANCONFIG failed on channel 1: No such device or address (6) Can you help me to guess the problem? thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed
On Dec 23, 2007 7:24 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: xpp/utils/zapconf ? yes the file /etc/zaptel.conf still remains empty, Which suggests that the module hasn't really loaded or anyway did not register channels. Or it has, but they are for empty slots. can you suggest me some command to enable the debug of the module? the card has phisically installed the module, i've checked it. And it is correctly powered. Do you know some method to check if the card is working? here the lspci ouput: 00:0e.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 9100:0001 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at a800 [size=256] Memory at f800 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenVox B800P and asterisk 1.4/ mISDN-1_1_7
Hi i've installed this software: SOFTWARE mISDN-1_1_7 mISDNuser-1_1_7 Asterisk-1.4.15 SOFTWARE misdn is correctly loaded by misdn-inist start Here there is the misdn.conf (copied from an existing and working installation with Asterisk 1.2.x and one BN8S0) MISDN.CONF [general] misdn_init=/etc/misdn-init.conf debug=0 bridging=no stop_tone_after_first_digit=yes append_digits2exten=yes dynamic_crypt=no crypt_prefix=** crypt_keys=test,muh jitterbuffer=4000 jitterbuffer_upper_threshold=0 context=misdn language=en musicclass=maracaibo senddtmf=yes far_alerting=no allowed_bearers=all nationalprefix=0 internationalprefix=00 rxgain=0 txgain=0 te_choose_channel=no pmp_l1_check=yes need_more_infos=no method=standard dialplan=0 localdialplan=0 cpndialplan=0 early_bconnect=yes incoming_early_audio=no presentation=-1 screen=-1 echocancelwhenbridged=no jitterbuffer=4000 jitterbuffer_upper_threshold=0 hdlc=no [TEports] ports=1,2,3,4,5,6,7,8 context=from-pstn msns=* MISDN.CONF When i start asterisk i get tihis warning: ** ASTERISK CLI mISDN_close: fid(19) isize(131072) inbuf(0xb6fac008) irp(0xb6fac008) iend(0xb6fac008) == Parsing '/etc/asterisk/misdn.conf': Found [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:929 _build_general_config: misdn.conf: jitterbuffer=4000 (section: general) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:929 _build_general_config: misdn.conf: jitterbuffer_upper_threshold=0 (section: general) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:985 _build_port_config: misdn.conf: echocancelwhenbridged=no (section: default) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf: ports=3,4,5,6,7,8 (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf: ports=4,5,6,7,8 (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf: ports=5,6,7,8 (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf: ports=6,7,8 (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf: ports=7,8 (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf: ports=8 (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. [Dec 15 12:56:44] WARNING[4170]: misdn_config.c:977 _build_port_config: misdn.conf : ports=(null) (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. P[ 0] Got: 1 from get_ports P[ 1] this is a unknown port type 0x == Registered channel type 'mISDN' (Channel driver for mISDN Support (Bri/Pri)) == Registered application 'misdn_set_opt' == Registered application 'misdn_facility' == Registered application 'misdn_check_l2l1' P[ 0] -- mISDN Channel Driver Registered -- chan_misdn.so = (Channel driver for mISDN Support (BRI/PRI)) ** ASTERISK CLI and in the kernel prints that in dmesg: * DMESG mISDN_dsp: Audio DSP Rev. 1.29 (debug=0x0) EchoCancellor MG2 dtmfthreshold(100) mISDN_dsp: DSP clocks every 128 samples. This equals 4 jiffies. mISDN: INTERNAL ERROR in /data/programmi/install-misdn-mqueue/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range mISDN: INTERNAL ERROR in /data/programmi/install-misdn-mqueue/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range mISDN: INTERNAL ERROR in /data/programmi/install-misdn-mqueue/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range mISDN: INTERNAL ERROR in /data/programmi/install-misdn-mqueue/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range mISDN: INTERNAL ERROR in /data/programmi/install-misdn-mqueue/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range mISDN: INTERNAL ERROR in /data/programmi/install-misdn-mqueue/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range * DMESG Can you help me to guess the problem? Thanks -- /*/ nik600 https
[asterisk-users] new Asterisk installation with openvox 1.2 or 1.4?
Hi i need to install a server with this hardware: 1 OpenVox B800P 1 OpenVox A800P01 4 OpenVox FXS-100 FXS100 4 OctWare SoftEchoSOFTECHO Do you suggest 1.2 or 1.4 branch? Is now 1.4 stable ? I've tried 1.4 the last year but i've experienced many problems with misdn drivers. Thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] store 2 separate records in cdr when a call is transferd
Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to store two records in the CDR instead of one, when a call is transferd. Is it possibile now? Thanks to all -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] store 2 separate records in cdr when a call is transferd
for blind transfer! Many thanks! On Nov 20, 2007 2:24 PM, Atis Lezdins [EMAIL PROTECTED] wrote: nik600 wrote: Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to store two records in the CDR instead of one, when a call is transferd. Is it possibile now? Thanks to all You want to do that on blind transfer or attended transfer? I got it working on blindxfer - it's pretty simple. Do a ResetCDR(w) in the context defined within TRANSFER_CONTEXT var. Attended transfers are much more nightmare for CDRs.. There are several channels involved, so it would need some cleaning to get what you want (i just don't use them) Regards, Atis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call center manager for Asterisk (Release 0.5)
CCMANAGER 0.5 released!! NOTE: this is a previous alpha release, maybe there is some customization to do on the settings files, i can't write a clear and complete howto at the moment I don't have released upgrades in the last months but the project is still alive i'm too busy at the moment, i'm following other projects to have some resources (both money and time) and then i can continue this project. Otherwise, i think that new upgrades will follow in the next months, if you have requests post it to the mailinglist on sourceforge I'm still looking to people that want to join this project, the new steps are: - integration with AJAX - project and implementation of an XML layer to manage n server (load balancing, logging and so...) from one ccmanager NEWS: the most important news is that ccmanager reports now supports both the native format that the new reportmaker format ( http://sourceforge.net/projects/reportmaker ) FEATURES: - users management - call generation (making a GET or POST request on a certain URL) - queue management (LOGIN / LOGOUT / QUEUE STATUS) - pickup a call from a queue even if the user isn't logged in the queue - outbound call in customizable context - queue stats import from queue_log - queue reports creation (using an open xml format and reportmaker format) - report export in - html - rtf - xls - pdf FEATURES OF REPORTMAKER reportmaker allows you to define a generic report in xml containing sections,graphs,tables,images. The data can be retrieved directly with sql query. The report can be exported in various formats (html,xml,rtf,pdf) CHANGELOG: 20/08/2007 - added the possibility to specify a different database directly in the report - added the project reportmaker for the report generation - mantained the compatibility with old ccmanager report style - fixed the css for calendar 11/07/2007 - added the file update_stats.php - changed the update method 16/03/2007 - fixed an error for the stats / update script (event ABANDON) - changed the date fromat from Y-m-d h:i:s to Y-m-d H:i:s - addedd the possibility to have multiple graphs on a report - added 6 new reports 14/03/2007 - added the module reports - integrated the module reports with the module stats - now you can generate your reports using an xml format 11/03/2007 - added the module stats - updated the file db.sql with sql instructions for the creation of queue_stats table - added the files view.sql bye -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple iax users on the same host
Hi i'm setting up a hylafax server, using iaxmodem to talk with asterisk (asterisk and hylafax are both on the same lan). Can i setup on the same host (Hylafax) multiple iax accounts ? (each account is used by a iaxmodem instance). The account can be on the same port or should i change the port for each iax account? Thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
