Re: [asterisk-users] queue not skipping ringing phone
Am Mittwoch 21 Dezember 2011, 07:04:03 schrieb Matt Hamilton: I have a queue that distributes calls among 3 phones. When a phone is in use (including on hold), queue skips that device and sends the call to the next available one as expected. On the other hand, if a call comes in while one of the phones is ringing, the queue doesn't seem to recognize that phone as in use and sends the second call to the ringing phone. If the first call is answered, the second call is sent to the next available phone right away. I'm new to asterisk and wondering if this is normal; I thought the ringing phone would be skipped as in use as well. Is there a setting on the asterisk side that I can use to force the queue to skip the ringing phone, or should this somehow be done on the phone itself? Thanks, Matt I think it is up to your phones to allow only one concurrent session, you could check call-waiting is deactivated on your phones?! If your phones allow more than one active dialog you probably wont have that much fun with queues... And make sure you have read the Queue Empty Options section of the queues.conf example as some parameters changed to be more flexible (joinempty = ringing etc... ). That could be interesting too... hth, Sebastian Denz -- Sebastian Denz d...@gonicus.de (Senior Technical Consultant) * GONICUS GmbH * Moehnestrasse 11-17 * D-59755 Arnsberg * Tel.: +49 (0) 29 32 / 9 16 - 0 * Fax: +49 (0) 29 32 / 9 16 - 270 * http://www.GONICUS.de * http://twitter.com/gonicus *Sitz der Gesellschaft: Moehnestrasse 11-17 * D-59755 Arnsberg *Geschaeftsfuehrer: Rainer Luelsdorf, Alfred Schroeder *Vorsitzender des Beirats: Juergen Michels *Amtsgericht Arnsberg * HRB 1968 -- _ -- 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] DTMF Testing software to test IVR system
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can used it?
like A script make the call by using localdail, .call file or any method.
And after landing the call we send dtmf to IVR system automatically as per
my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same = n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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Re: [asterisk-users] DTMF Testing software to test IVR system
Create a callfile with local channel and once first call leg is answered,
use wait() and senddtmf() application on second call leg.
CALLFILE sample:
Channel: LOCAL/1234\@test_ivr
Context: senddtmf
Extension: s
Priority: 1
Extensions.conf sample:
;-- FIRST LEG CALL --;
[test_ivr]
exten = 1234,1,Answer()
same = n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
;--SECOND LEG CALL --;
[senddtmf]
exten = s,1,Noop(# TEST:IVR ##)
; We should wait atleast 'n' of seconds. Where n is length of IVR file in
seconds.
same = n,Wait(10)
same = n,SendDTMF(1)
--SATISH BAROT
On Wed, Dec 28, 2011 at 1:55 PM, virendra bhati virbh...@gmail.com wrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can used
it?
like A script make the call by using localdail, .call file or any method.
And after landing the call we send dtmf to IVR system automatically as per
my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same =
n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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Re: [asterisk-users] DTMF Testing software to test IVR system
Hi Satish,
Thank you Satish. I did the same before your e-mail i saw. But i got
another issue in such case.
DTMF is passed to that channels but in case I will make the complete IVR
system for calling server end. and which become so complected to do it.
Is there any alternate way by which I get the response and send DTMF only.
So that complete IVR flow willn't be required to implement at originator
server.
On Wed, Dec 28, 2011 at 2:50 PM, Satish Barot satish4aster...@gmail.comwrote:
Create a callfile with local channel and once first call leg is answered,
use wait() and senddtmf() application on second call leg.
CALLFILE sample:
Channel: LOCAL/1234\@test_ivr
Context: senddtmf
Extension: s
Priority: 1
Extensions.conf sample:
;-- FIRST LEG CALL --;
[test_ivr]
exten = 1234,1,Answer()
same = n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
;--SECOND LEG CALL --;
[senddtmf]
exten = s,1,Noop(# TEST:IVR ##)
; We should wait atleast 'n' of seconds. Where n is length of IVR file in
seconds.
same = n,Wait(10)
same = n,SendDTMF(1)
--SATISH BAROT
On Wed, Dec 28, 2011 at 1:55 PM, virendra bhati virbh...@gmail.comwrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can used
it?
like A script make the call by using localdail, .call file or any method.
And after landing the call we send dtmf to IVR system automatically as per
my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same =
n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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Re: [asterisk-users] DTMF Testing software to test IVR system
Hi,
You can use combination of SendDTMF() and wait() in such a way that you
traverse through the IVR tree just as Satish mentioned.
SendDTMF(1)
Wait(3)
SendDTMF(2)
Wait(2)
SendDTMF(5678123490)
See also:
*WaitForNoise()* , WaitForSilence(), AMD()
Regards,
Sammy.
On Wed, Dec 28, 2011 at 2:32 PM, virendra bhati virbh...@gmail.com wrote:
Hi Satish,
Thank you Satish. I did the same before your e-mail i saw. But i got
another issue in such case.
DTMF is passed to that channels but in case I will make the complete IVR
system for calling server end. and which become so complected to do it.
Is there any alternate way by which I get the response and send DTMF only.
So that complete IVR flow willn't be required to implement at originator
server.
On Wed, Dec 28, 2011 at 2:50 PM, Satish Barot
satish4aster...@gmail.comwrote:
Create a callfile with local channel and once first call leg is answered,
use wait() and senddtmf() application on second call leg.
CALLFILE sample:
Channel: LOCAL/1234\@test_ivr
Context: senddtmf
Extension: s
Priority: 1
Extensions.conf sample:
;-- FIRST LEG CALL --;
[test_ivr]
exten = 1234,1,Answer()
same = n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
;--SECOND LEG CALL --;
[senddtmf]
exten = s,1,Noop(# TEST:IVR ##)
; We should wait atleast 'n' of seconds. Where n is length of IVR file in
seconds.
same = n,Wait(10)
same = n,SendDTMF(1)
--SATISH BAROT
On Wed, Dec 28, 2011 at 1:55 PM, virendra bhati virbh...@gmail.comwrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can used
it?
like A script make the call by using localdail, .call file or any
method. And after landing the call we send dtmf to IVR system automatically
as per my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same =
n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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Thanks and regards
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Software Engineer
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Re: [asterisk-users] Is Asterisk 1.4 compatible with 1.8.7 ?
On 28 December 2011 03:02, Joseph syscon...@gmail.com wrote: No, it makes no difference, on the other end is asterisk 1.4.39 and 1.8.8 is still giving me: Executing [4@internal:1] Dial(SIP/11-0003, IAX2/home_server:@192.168.141.1/4,30,rw) in new stack -- Called IAX2/home_server:@192.168.141.1/4 [Dec 27 20:00:16] WARNING[16398]: chan_iax2.c:10672 socket_process: Call rejected by 192.168.141.1: Unable to negotiate codec -- Hungup 'IAX2/192.168.141.1:4569-5678' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [4@internal:2] Hangup(SIP/11-0003, ) in new stack == Spawn extension (internal, 4, 2) exited non-zero on 'SIP/11-0003' -- Joseph [snip] Have you tried enabling IAX2 debug at both ends to see if the packet decode provides any more clues? Regards, Steve -- _ -- 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] Direct media path on Avaya IPOFFICE and Asterisk with H323 Trunk
Hi List, I would like create a H323 trunk from Avaya IPOFFICE to Asterisk, but i would like activate a direct media path for the RTP transit directly between the phone and the Asterisk. Now, - H323 Trunk is OK - RTP from the phone transit directly to Asterisk (activate strictrtp=no in rtp.conf, and Allow Direct Media Path option in Avaya Ipoffice) H323: Phone -- Avaya Ipoffice -- Asterisk RTP: Phone -- Asterisk But, asterisk send RTP stream to the IPOFFICE and not to phone directly. H323: Asterisk -- Avaya ipoffice -- phone RTP: Asterisk -- Avaya ipoffice -- phone Can you help me on this issue ? It's very important that RTP stream transit directly. Thank in advance *h323.conf* * * [ipo-hexa] type=friend host=192.168.254.200 fastStart=no context=incoming_ipo h245Tunneling=yes port=1720; UPDATE with appropriate port e164=101 -- Cordialement Hubert Mickaël Hexanet -- _ -- 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] DTMF Testing software to test IVR system
What I understand from your reply is, you also like to have multiple Read()
in 'support' and 'help' extensions as well.
In that case you can have something like this in [senddtmf]
exten = s,1,Noop(# TEST:IVR ##)
; We should wait atleast 'n' of seconds. Where n is length of IVR file in
seconds.
same = n,Wait(10)
same = n,SendDTMF(1)
;- Wait for message in second Read --;
same = n,Wait(5)
same = n,SendDTMF(2)
;- Wait for message in third Read --;
same = n,Wait(15)
same = n,SendDTMF(1)
...
...
same = n,Wait(10)
same = n,SendDTMF(3)
Hope this helps you,
--SATISH BAROT
On Wed, Dec 28, 2011 at 3:02 PM, virendra bhati virbh...@gmail.com wrote:
Hi Satish,
Thank you Satish. I did the same before your e-mail i saw. But i got
another issue in such case.
DTMF is passed to that channels but in case I will make the complete IVR
system for calling server end. and which become so complected to do it.
Is there any alternate way by which I get the response and send DTMF only.
So that complete IVR flow willn't be required to implement at originator
server.
On Wed, Dec 28, 2011 at 2:50 PM, Satish Barot
satish4aster...@gmail.comwrote:
Create a callfile with local channel and once first call leg is answered,
use wait() and senddtmf() application on second call leg.
CALLFILE sample:
Channel: LOCAL/1234\@test_ivr
Context: senddtmf
Extension: s
Priority: 1
Extensions.conf sample:
;-- FIRST LEG CALL --;
[test_ivr]
exten = 1234,1,Answer()
same = n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
;--SECOND LEG CALL --;
[senddtmf]
exten = s,1,Noop(# TEST:IVR ##)
; We should wait atleast 'n' of seconds. Where n is length of IVR file in
seconds.
same = n,Wait(10)
same = n,SendDTMF(1)
--SATISH BAROT
On Wed, Dec 28, 2011 at 1:55 PM, virendra bhati virbh...@gmail.comwrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can used
it?
like A script make the call by using localdail, .call file or any
method. And after landing the call we send dtmf to IVR system automatically
as per my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same =
n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
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Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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[asterisk-users] Chan_ss7 clustering config with single point
Hi team, Can any one share with me clustering configuration file SS7.conf for single pointcode with four slc. two different machine each host having 2 slc respectively. Thanks Vinod Dharashive Sent from BlackBerry® on Airtel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Function TESTTIME example
Hi,
I do not know, whether this is the best way to use TESTTIME function, but for
me it is working in that way:
exten = 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
OR
You can use this:
Set(__TESTTIME=${STRPTIME(2011-12-25 18:00:00,Europe/Vilnius,%Y-%m-%d
%H:%M:%S)})
Best regards,
Mindaugas
Hi,
Has someone a dialplan example using TESTTIME function (see core show
function TESTTIME) ?
I'm only getting replies such as Function TESTTIME cannot be read.
Regards
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Re: [asterisk-users] Function TESTTIME example
Hi,
Thanks for replying.
I'm afraid this :
[foobar]
exten = 123,1,Verbose(0,Into context ${CONTEXT})
exten = 123,n,Verbose(0,Time is ${STRFTIME()})
exten = 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
exten = 123,n,Verbose(0,Time is ${STRFTIME()})
exten = 123,n,HangUp()
... gives this:
-- Executing [123@foobar:1] Verbose(SIP/7005-006b, 0,Into
context foobar) in new stack
Into context foobar
-- Executing [123@foobar:2] Verbose(SIP/7005-006b, 0,Time
is Wed Dec 28 16:25:59 2011) in new stack
Time is Wed Dec 28 16:25:59 2011
-- Executing [123@foobar:3] Set(SIP/7005-006b,
TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius) in new stack
-- Executing [123@foobar:4] Verbose(SIP/7005-006b, 0,Time
is Wed Dec 28 16:25:59 2011) in new stack
Time is Wed Dec 28 16:25:59 2011
-- Executing [123@foobar:5] Hangup(SIP/7005-006b, ) in new stack
Do you see the same behaviour ?
2011/12/28, Mindaugas Jasiulis mindaugas.jasiu...@mediafon.lt:
Hi,
I do not know, whether this is the best way to use TESTTIME function, but
for me it is working in that way:
exten = 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
OR
You can use this:
Set(__TESTTIME=${STRPTIME(2011-12-25 18:00:00,Europe/Vilnius,%Y-%m-%d
%H:%M:%S)})
Best regards,
Mindaugas
Hi,
Has someone a dialplan example using TESTTIME function (see core show
function TESTTIME) ?
I'm only getting replies such as Function TESTTIME cannot be read.
Regards
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Re: [asterisk-users] DTMF Testing software to test IVR system
On 11-12-28 03:25 AM, virendra bhati wrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can used it?
like A script make the call by using localdail, .call file or any method.
And after landing the call we send dtmf to IVR system automatically as per
my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same = n,Read(value,pleasePress1forSupportPress2forHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(suppot,1))
same = n,ExecIf($[${value}=2]?Goto(help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
We have DTMF based tests for the testsuite[1] that you could use.
[1] http://svn.asterisk.org/svn/testsuite/asterisk/trunk/
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Re: [asterisk-users] Is Asterisk 1.4 compatible with 1.8.7 ?
On 12/28/11 10:07, Steve Davies wrote: On 28 December 2011 03:02, Joseph syscon...@gmail.com wrote: No, it makes no difference, on the other end is asterisk 1.4.39 and 1.8.8 is still giving me: Executing [4@internal:1] Dial(SIP/11-0003, IAX2/home_server:@192.168.141.1/4,30,rw) in new stack -- Called IAX2/home_server:@192.168.141.1/4 [Dec 27 20:00:16] WARNING[16398]: chan_iax2.c:10672 socket_process: Call rejected by 192.168.141.1: Unable to negotiate codec -- Hungup 'IAX2/192.168.141.1:4569-5678' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [4@internal:2] Hangup(SIP/11-0003, ) in new stack == Spawn extension (internal, 4, 2) exited non-zero on 'SIP/11-0003' -- Joseph [snip] Have you tried enabling IAX2 debug at both ends to see if the packet decode provides any more clues? Regards, Steve I've enabled iax2 debug on both ends and on asterisk-1.4.39 I get: NOTICE[2412]: chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. On asterisk-1.8.8 I get: -- Executing [4@internal:1] Dial(SIP/11-0002, IAX2/home_server:546987@192.168.141.1/4,30,rw) in new stack -- Called IAX2/home_server:546987@192.168.141.1/4 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00019ms SCall: 01953 DCall: 0 [192.168.141.1:4569] VERSION : 2 CALLED NUMBER : 4 CODEC_PREFS : () CALLING NUMBER : 11 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Joseph LANGUAGE: en USERNAME: home_server FORMAT : 2 FORMAT2 : gsm CAPABILITY : 1795 CAPABILITY2 : unknown ADSICPE : 2 DATE TIME : 2011-12-28 10:21:08 Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN Timestamp: 00019ms SCall: 1 DCall: 01953 [192.168.141.1:4569] CALLTOKEN : 51 bytes Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00039ms SCall: 01953 DCall: 0 [192.168.141.1:4569] VERSION : 2 CALLED NUMBER : 4 CODEC_PREFS : () CALLING NUMBER : 11 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Joseph LANGUAGE: en USERNAME: home_server FORMAT : 2 FORMAT2 : gsm CAPABILITY : 1795 CAPABILITY2 : unknown ADSICPE : 2 DATE TIME : 2011-12-28 10:21:08 CALLTOKEN : 51 bytes Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 5ms SCall: 07261 DCall: 01953 [192.168.141.1:4569] AUTHMETHODS : 3 CHALLENGE : \x35\x34\x38\x39\x33\x31\x37\x34\x38 USERNAME: home_server Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00059ms SCall: 01953 DCall: 07261 [192.168.141.1:4569] MD5 RESULT : f841f666557e416cb45f4f69fb0e74b2 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT Timestamp: 00024ms SCall: 07261 DCall: 01953 [192.168.141.1:4569] CAUSE : Unable to negotiate codec CAUSE CODE : 58 [Dec 28 10:21:09] WARNING[14693]: chan_iax2.c:10672 socket_process: Call rejected by 192.168.141.1: Unable to negotiate codec Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00024ms SCall: 01953 DCall: 07261 [192.168.141.1:4569] -- Hungup 'IAX2/192.168.141.1:4569-1953' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [4@internal:2] Hangup(SIP/11-0002, ) in new stack == Spawn extension (internal, 4, 2) exited non-zero on 'SIP/11-0002' -- Joseph -- _ -- 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] [SOLVED] Is Asterisk 1.4 compatible with 1.8.7 ?
On 12/28/11 10:45, Joseph wrote: [snip] Have you tried enabling IAX2 debug at both ends to see if the packet decode provides any more clues? Regards, Steve I've enabled iax2 debug on both ends and on asterisk-1.4.39 I get: NOTICE[2412]: chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. On asterisk-1.8.8 I get: -- Executing [4@internal:1] Dial(SIP/11-0002, IAX2/home_server:546987@192.168.141.1/4,30,rw) in new stack -- Called IAX2/home_server:546987@192.168.141.1/4 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00019ms SCall: 01953 DCall: 0 [192.168.141.1:4569] VERSION : 2 CALLED NUMBER : 4 CODEC_PREFS : () CALLING NUMBER : 11 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Joseph LANGUAGE: en USERNAME: home_server FORMAT : 2 FORMAT2 : gsm CAPABILITY : 1795 CAPABILITY2 : unknown ADSICPE : 2 DATE TIME : 2011-12-28 10:21:08 Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN Timestamp: 00019ms SCall: 1 DCall: 01953 [192.168.141.1:4569] CALLTOKEN : 51 bytes Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00039ms SCall: 01953 DCall: 0 [192.168.141.1:4569] VERSION : 2 CALLED NUMBER : 4 CODEC_PREFS : () CALLING NUMBER : 11 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Joseph LANGUAGE: en USERNAME: home_server FORMAT : 2 FORMAT2 : gsm CAPABILITY : 1795 CAPABILITY2 : unknown ADSICPE : 2 DATE TIME : 2011-12-28 10:21:08 CALLTOKEN : 51 bytes Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 5ms SCall: 07261 DCall: 01953 [192.168.141.1:4569] AUTHMETHODS : 3 CHALLENGE : \x35\x34\x38\x39\x33\x31\x37\x34\x38 USERNAME: home_server Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00059ms SCall: 01953 DCall: 07261 [192.168.141.1:4569] MD5 RESULT : f841f666557e416cb45f4f69fb0e74b2 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT Timestamp: 00024ms SCall: 07261 DCall: 01953 [192.168.141.1:4569] CAUSE : Unable to negotiate codec CAUSE CODE : 58 [Dec 28 10:21:09] WARNING[14693]: chan_iax2.c:10672 socket_process: Call rejected by 192.168.141.1: Unable to negotiate codec Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00024ms SCall: 01953 DCall: 07261 [192.168.141.1:4569] -- Hungup 'IAX2/192.168.141.1:4569-1953' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [4@internal:2] Hangup(SIP/11-0002, ) in new stack == Spawn extension (internal, 4, 2) exited non-zero on 'SIP/11-0002' -- Joseph -- _ -- 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 I noticed above: FORMAT : 2 FORMAT2 : gsm Why isn't iax2 recognizes ulaw / alaw ? I have specified in iax.conf: [clinic_server] type=friend host=dynamic context=internal disallow=all allow=ulaw allow=alaw I have added coded gsm allow=gsm and now it is working :-/ -- Joseph -- _ -- 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] queue not skipping ringing phone
Thanks Sebastian. It was a phone related issue. Factory resetting the phones and reconfiguring them fixed it. It probably was a CW issue as you suggested. I think it is up to your phones to allow only one concurrent session, you could check call-waiting is deactivated on your phones?! If your phones allow more than one active dialog you probably wont have that much fun with queues... And make sure you have read the Queue Empty Options section of the queues.conf example as some parameters changed to be more flexible (joinempty = ringing etc... ). That could be interesting too... hth, Sebastian Denz -- _ -- 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] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed yet
Hi everyone, I see that there was a bug in version 1.8.5.x and people were advised to move to 1.8.7.1 but now I have 1.8.7.1 and experiencing the same problem. Here is the output: *chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x100 (g729)/0x100 (g729)* * * Now, I see an update to 1.8.8.1 is available. I am wondering if this issue is fixed in this version or not? Furthermore, has anyone tested 1.8.8.1 yet? Are there any other problems to that? It's frustrating as I see we should once again move back to 1.6x and forget about 1.8x all together. Any input is appreciated. -- _ -- 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] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed yet
The issue is not fixed in 1.8.8.0 either. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Wednesday, December 28, 2011 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed yet Hi everyone, I see that there was a bug in version 1.8.5.x and people were advised to move to 1.8.7.1 but now I have 1.8.7.1 and experiencing the same problem. Here is the output: chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x100 (g729)/0x100 (g729) Now, I see an update to 1.8.8.1 is available. I am wondering if this issue is fixed in this version or not? Furthermore, has anyone tested 1.8.8.1 yet? Are there any other problems to that? It's frustrating as I see we should once again move back to 1.6x and forget about 1.8x all together. Any input is appreciated. -- _ -- 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] Monitor Command Records separate channales
Hi All, I am trying to record Call, but when the call is done I have one file but the conversation inside it is separate into calls conversation and receiver its single file but separate recording, How can I make it mixed together so the conversation will be normal? Thanx -- _ -- 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] Monitor Command Records separate channales
Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, December 28, 2011 2:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Monitor Command Records separate channales Hi All, I am trying to record Call, but when the call is done I have one file but the conversation inside it is separate into calls conversation and receiver its single file but separate recording, How can I make it mixed together so the conversation will be normal? Thanx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed yet
This might or might not help, but here is the offending code in 1.8.8
case AST_FRAME_VOICE:
if (!(frame-subclass.codec ast-nativeformats)) {
char s1[512], s2[512], s3[512];
ast_log(LOG_WARNING, Asked to transmit frame type
%s, while native formats is %s read/write = %s/%s\n,
ast_getformatname(frame-subclass.codec),
ast_getformatname_multiple(s1, sizeof(s1),
ast-nativeformats AST_FORMAT_AUDIO_MASK),
ast_getformatname_multiple(s2, sizeof(s2),
ast-readformat),
ast_getformatname_multiple(s3, sizeof(s3),
ast-writeformat));
and the comparable code in 10.0.0
case AST_FRAME_VOICE:
if (!(ast_format_cap_iscompatible(ast-nativeformats,
frame-subclass.format))) {
char s1[512];
ast_log(LOG_WARNING, Asked to transmit frame type
%s, while native formats is %s read/write = %s/%s\n,
ast_getformatname(frame-subclass.format),
ast_getformatname_multiple(s1, sizeof(s1),
ast-nativeformats),
ast_getformatname(ast-readformat),
ast_getformatname(ast-writeformat));
I personally avoided the 1.6 and 1.8 branches like the plague and don't know
if this bug is corrected by the other fixes in 10.0.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, December 28, 2011 2:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed
yet
The issue is not fixed in 1.8.8.0 either.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Wednesday, December 28, 2011 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed yet
Hi everyone,
I see that there was a bug in version 1.8.5.x and people were advised to
move to 1.8.7.1 but now I have 1.8.7.1 and experiencing the same problem.
Here is the output:
chan_sip.c: Asked to transmit frame type ulaw, while native formats is
0x100 (g729) read/write = 0x100 (g729)/0x100 (g729)
Now, I see an update to 1.8.8.1 is available. I am wondering if this issue
is fixed in this version or not? Furthermore, has anyone tested 1.8.8.1 yet?
Are there any other problems to that? It's frustrating as I see we should
once again move back to 1.6x and forget about 1.8x all together.
Any input is appreciated.
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Re: [asterisk-users] Monitor Command Records separate channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command
this is how I use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Suggestion 1 - mixmonitor instead of monitor
Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitor Command Records separate channales
Hi All,
I am trying to record Call, but when the call is done I have one file but
the conversation inside it is separate into calls conversation and receiver
its single file but separate recording, How can I make it mixed
together so the conversation will be normal?
Thanx
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[asterisk-users] 1.6 and 1.8
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, December 28, 2011 3:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed yet I personally avoided the 1.6 and 1.8 branches like the plague and don't know if this bug is corrected by the other fixes in 10.0. -- We have had good luck with recent 1.6.2.x releases. -- _ -- 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] Monitor Command Records separate channales
According to the monitor documentation, the format you specified should be
calling SOX and mixing on call completion. What versions of SOX and
Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command this is how I use
Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H
:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitor Command Records separate channales
Hi All,
I am trying to record Call, but when the call is done I have one file but
the conversation inside it is separate into calls conversation and receiver
its single file but separate recording, How can I make it mixed
together so the conversation will be normal?
Thanx
--
_
-- 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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Re: [asterisk-users] Monitor Command Records separate channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version, my
problem is not mixing It's the same file but inside that file two seperate
records first callers then reciever
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
According to the monitor documentation, the format you specified should be
calling SOX and mixing on call completion. What versions of SOX and
Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command this is how I use
Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H
:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitor Command Records separate channales
Hi All,
I am trying to record Call, but when the call is done I have one file but
the conversation inside it is separate into calls conversation and receiver
its single file but separate recording, How can I make it mixed
together so the conversation will be normal?
Thanx
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Re: [asterisk-users] Monitor Command Records separate channales
Can you post a CLI output of the Monitor output? I'm supposing that something
in your $(STRFTIME) string might be eating the M option.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version, my
problem is not mixing It's the same file but inside that file two seperate
records first callers then reciever
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
According to the monitor documentation, the format you specified
should be calling SOX and mixing on call completion. What versions of
SOX and Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command this is how I
use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m
-%d_%H
:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitor Command Records separate channales
Hi All,
I am trying to record Call, but when the call is done I have one file
but the conversation inside it is separate into calls conversation and
receiver its single file but separate recording, How can I make
it mixed together so the conversation will be normal?
Thanx
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[asterisk-users] Question on hung channel
I ran into a rare situation today. A really short message is being played over the ALSA or console channel from one asterisk box to another. Both running 1.4.30. the incoming context on the ALSA or Console port box first runs an AGI before connecting the audio path. The AGI got hung up for an non-asterisk issue. then finally continued. The audio from the server was DONE, the server already hung up the call. So the console port is just setting there with nothing to do. the next call in says - opps the port is busy so just give busy signal. The original call never frees up and goes away. How do I get the orphaned call to just close and go away? Thanks, Jerry -- _ -- 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] [SOLVED] Re: CSipSimple audio issue with DAHDI/IAX2 calls
On 12/02/2011 11:37 AM, Anthony Messina wrote: I've just connected my new Android (Motorola RAZR) phone to Asterisk using CSipSimple and have discovered that on any call between CSipSimple and an Asterisk DAHDI or IAX2 channel, the 'other' end of the call will hear a rhythmic tapping as if my voice stream is being chopped up in equal parts about every 500ms or so. I can always hear the remote party without issue, regardless of the channel type. The issue occurs only on connections to DAHDI channels (even those that don't pass through the PSTN), and IAX2 connections to remote Asterisk servers. This issue occurs whether I am using WiFi, 3G or 4G connections on the Android. This does NOT occur on any SIP channels, local to my Asterisk box, or to others. I've investigated changing just about every setting on the Android with no resolution. It seems like some sort of timing issue and is strange to me that this issue is confined to DAHDI and IAX2 channels, but I'm no expert. I have tested using only res_timing_dadhi.so since I have the card, but that did not help either. Would anyone be willing to point me in the right direction for resolving this issue? Please let me know if any more information is required. Thanks in advance. -A Enabling the jitterbuffer=yes on the iax channel and setting Set(JITTERBUFFER(fixed)=default) prior to any calls to DAHDI channels seems to resolve the issue for now. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] Monitor Command Records separate channales
Can u plz tell me how , I forgot how to run asterisk cli
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
Can you post a CLI output of the Monitor output? I'm supposing that
something in your $(STRFTIME) string might be eating the M option.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version, my
problem is not mixing It's the same file but inside that file two
seperate records first callers then reciever
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
According to the monitor documentation, the format you specified
should be calling SOX and mixing on call completion. What versions of
SOX and Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command this is how I
use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m
-%d_%H
:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitor Command Records separate channales
Hi All,
I am trying to record Call, but when the call is done I have one file
but the conversation inside it is separate into calls conversation and
receiver its single file but separate recording, How can I make
it mixed together so the conversation will be normal?
Thanx
--
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Re: [asterisk-users] Monitor Command Records separate channales
Asterisk -vvvrc
Is how you would get it live
After the fact you might find it in /var/log/asterisk/full
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Can u plz tell me how , I forgot how to run asterisk cli
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
Can you post a CLI output of the Monitor output? I'm supposing that
something in your $(STRFTIME) string might be eating the M option.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version,
my problem is not mixing It's the same file but inside that file
two seperate records first callers then reciever
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
According to the monitor documentation, the format you specified
should be calling SOX and mixing on call completion. What versions
of SOX and Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command this is how
I use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%
m
-%d_%H
:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitor Command Records separate channales
Hi All,
I am trying to record Call, but when the call is done I have one file
but the conversation inside it is separate into calls conversation
and receiver its single file but separate recording, How can I
make it mixed together so the conversation will be normal?
Thanx
--
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Re: [asterisk-users] 1.6 and 1.8
Can somebody point me to an explanation from Kevin or Tzafir or someone else up the food chain explaining the differences/benefits of 1.6/1.8 vs 1.4/10.0? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, December 28, 2011 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 1.6 and 1.8 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, December 28, 2011 3:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed yet I personally avoided the 1.6 and 1.8 branches like the plague and don't know if this bug is corrected by the other fixes in 10.0. -- We have had good luck with recent 1.6.2.x releases. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor Command Records separate channales
My call happens with a queue , there is no full file but there is queue and
queue is useless, can u give me unix command to search all log files and print
moniter line?
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:
Asterisk -vvvrc
Is how you would get it live
After the fact you might find it in /var/log/asterisk/full
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Can u plz tell me how , I forgot how to run asterisk cli
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
Can you post a CLI output of the Monitor output? I'm supposing that
something in your $(STRFTIME) string might be eating the M option.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version,
my problem is not mixing It's the same file but inside that file
two seperate records first callers then reciever
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
According to the monitor documentation, the format you specified
should be calling SOX and mixing on call completion. What versions
of SOX and Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command this is how
I use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%
m
-%d_%H
:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitor Command Records separate channales
Hi All,
I am trying to record Call, but when the call is done I have one file
but the conversation inside it is separate into calls conversation
and receiver its single file but separate recording, How can I
make it mixed together so the conversation will be normal?
Thanx
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Monitor Command Records separate channales
I already searched using grep for the monitor word ... It doesn't exists
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:١٥ م, Faraj Khasib fkha...@iconnecths.com wrote:
My call happens with a queue , there is no full file but there is queue and
queue is useless, can u give me unix command to search all log files and
print moniter line?
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:
Asterisk -vvvrc
Is how you would get it live
After the fact you might find it in /var/log/asterisk/full
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Can u plz tell me how , I forgot how to run asterisk cli
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
Can you post a CLI output of the Monitor output? I'm supposing that
something in your $(STRFTIME) string might be eating the M option.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version,
my problem is not mixing It's the same file but inside that file
two seperate records first callers then reciever
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
According to the monitor documentation, the format you specified
should be calling SOX and mixing on call completion. What versions
of SOX and Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command this is how
I use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%
m
-%d_%H
:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitor Command Records separate channales
Hi All,
I am trying to record Call, but when the call is done I have one file
but the conversation inside it is separate into calls conversation
and receiver its single file but separate recording, How can I
make it mixed together so the conversation will be normal?
Thanx
--
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Re: [asterisk-users] Monitor Command Records separate channales
Even using Queue there should still be a /var/log/asterisk/full that records
the Monitor then the following Queue/Dial commands. What is in your
/var/log/asterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
My call happens with a queue , there is no full file but there is queue and
queue is useless, can u give me unix command to search all log files and print
moniter line?
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:
Asterisk -vvvrc
Is how you would get it live
After the fact you might find it in /var/log/asterisk/full
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 3:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Can u plz tell me how , I forgot how to run asterisk cli
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
Can you post a CLI output of the Monitor output? I'm supposing that
something in your $(STRFTIME) string might be eating the M option.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version,
my problem is not mixing It's the same file but inside that file
two seperate records first callers then reciever
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
According to the monitor documentation, the format you specified
should be calling SOX and mixing on call completion. What versions
of SOX and Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command this is how
I use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-
%
m
-%d_%H
:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas [da...@debsinc.com]
Sent: Wednesday, December 28, 2011 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitor Command Records separate channales
Hi All,
I am trying to record Call, but when the call is done I have one
file but the conversation inside it is separate into calls
conversation and receiver its single file but separate
recording, How can I make it mixed together so the conversation will be
normal?
Thanx
--
_
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-- 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] Monitor Command Records separate channales
see attached ...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 3:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Even using Queue there should still be a /var/log/asterisk/full that records
the Monitor then the following Queue/Dial commands. What is in your
/var/log/asterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
My call happens with a queue , there is no full file but there is queue and
queue is useless, can u give me unix command to search all log files and print
moniter line?
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:
Asterisk -vvvrc
Is how you would get it live
After the fact you might find it in /var/log/asterisk/full
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 3:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Can u plz tell me how , I forgot how to run asterisk cli
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
Can you post a CLI output of the Monitor output? I'm supposing that
something in your $(STRFTIME) string might be eating the M option.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version,
my problem is not mixing It's the same file but inside that file
two seperate records first callers then reciever
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
According to the monitor documentation, the format you specified
should be calling SOX and mixing on call completion. What versions
of SOX and Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command this is how
I use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-
%
m
-%d_%H
:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas [da...@debsinc.com]
Sent: Wednesday, December 28, 2011 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitor Command Records separate channales
Hi All,
I am trying to record Call, but when the call is done I have one
file but the conversation inside it is separate into calls
conversation and receiver its single file but separate
recording, How can I make it mixed together so the conversation will be
normal?
Thanx
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
-- New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth
Re: [asterisk-users] Monitor Command Records separate channales
I would wager that your setup dumps what would normally be in /v/l/a/full into
/v/l/a/messages
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
see attached ...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 3:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Even using Queue there should still be a /var/log/asterisk/full that records
the Monitor then the following Queue/Dial commands. What is in your
/var/log/asterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
My call happens with a queue , there is no full file but there is queue and
queue is useless, can u give me unix command to search all log files and print
moniter line?
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:
Asterisk -vvvrc
Is how you would get it live
After the fact you might find it in /var/log/asterisk/full
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 3:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Can u plz tell me how , I forgot how to run asterisk cli
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
Can you post a CLI output of the Monitor output? I'm supposing that
something in your $(STRFTIME) string might be eating the M option.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version,
my problem is not mixing It's the same file but inside that file
two seperate records first callers then reciever
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
According to the monitor documentation, the format you specified
should be calling SOX and mixing on call completion. What versions
of SOX and Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command this is how
I use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-
%
m
-%d_%H
:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas [da...@debsinc.com]
Sent: Wednesday, December 28, 2011 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Monitor Command Records separate channales
Hi All,
I am trying to record Call, but when the call is done I have one
file but the conversation inside it is separate into calls
conversation and receiver its single file but separate
recording, How can I make it mixed together so the conversation will be
normal?
Thanx
--
Re: [asterisk-users] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed yet
So, what is really the effect of this and why is it hard to fix? Does this
bug disrupt processing the call? I see the log filled up with this error. I
do have a BUSY showing on forwarding to a number outside and that is what
concerns me. Not sure if caused by this bug. From reading CHANGES log, I
see that this has to do something with oversize packets in g729. Maybe it's
a setting issue?
Regards,
On Wed, Dec 28, 2011 at 3:19 PM, Danny Nicholas da...@debsinc.com wrote:
This might or might not help, but here is the offending code in 1.8.8
case AST_FRAME_VOICE:
if (!(frame-subclass.codec ast-nativeformats)) {
char s1[512], s2[512], s3[512];
ast_log(LOG_WARNING, Asked to transmit frame type
%s, while native formats is %s read/write = %s/%s\n,
ast_getformatname(frame-subclass.codec),
ast_getformatname_multiple(s1, sizeof(s1),
ast-nativeformats AST_FORMAT_AUDIO_MASK),
ast_getformatname_multiple(s2, sizeof(s2),
ast-readformat),
ast_getformatname_multiple(s3, sizeof(s3),
ast-writeformat));
and the comparable code in 10.0.0
case AST_FRAME_VOICE:
if (!(ast_format_cap_iscompatible(ast-nativeformats,
frame-subclass.format))) {
char s1[512];
ast_log(LOG_WARNING, Asked to transmit frame type
%s, while native formats is %s read/write = %s/%s\n,
ast_getformatname(frame-subclass.format),
ast_getformatname_multiple(s1, sizeof(s1),
ast-nativeformats),
ast_getformatname(ast-readformat),
ast_getformatname(ast-writeformat));
I personally avoided the 1.6 and 1.8 branches like the plague and don't
know
if this bug is corrected by the other fixes in 10.0.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, December 28, 2011 2:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed
yet
The issue is not fixed in 1.8.8.0 either.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Wednesday, December 28, 2011 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.8.7.1 forcing uLaw bug NOT fixed yet
Hi everyone,
I see that there was a bug in version 1.8.5.x and people were advised to
move to 1.8.7.1 but now I have 1.8.7.1 and experiencing the same problem.
Here is the output:
chan_sip.c: Asked to transmit frame type ulaw, while native formats is
0x100 (g729) read/write = 0x100 (g729)/0x100 (g729)
Now, I see an update to 1.8.8.1 is available. I am wondering if this issue
is fixed in this version or not? Furthermore, has anyone tested 1.8.8.1
yet?
Are there any other problems to that? It's frustrating as I see we should
once again move back to 1.6x and forget about 1.8x all together.
Any input is appreciated.
--
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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_
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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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_
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To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Monitor Command Records separate channales
but i tiried these commands and I didnt find anything about Monitor
[root@c-24-1-71-68 asterisk]# grep -R 'Monitor' *
[root@c-24-1-71-68 asterisk]# grep -R 'monitor' *
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 3:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales
I would wager that your setup dumps what would normally be in /v/l/a/full into
/v/l/a/messages
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
see attached ...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 3:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Even using Queue there should still be a /var/log/asterisk/full that records
the Monitor then the following Queue/Dial commands. What is in your
/var/log/asterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
My call happens with a queue , there is no full file but there is queue and
queue is useless, can u give me unix command to search all log files and print
moniter line?
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:
Asterisk -vvvrc
Is how you would get it live
After the fact you might find it in /var/log/asterisk/full
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 3:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Can u plz tell me how , I forgot how to run asterisk cli
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
Can you post a CLI output of the Monitor output? I'm supposing that
something in your $(STRFTIME) string might be eating the M option.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version,
my problem is not mixing It's the same file but inside that file
two seperate records first callers then reciever
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
According to the monitor documentation, the format you specified
should be calling SOX and mixing on call completion. What versions
of SOX and Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command this is how
I use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-
%
m
-%d_%H
:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas [da...@debsinc.com]
Sent: Wednesday, December 28, 2011 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Re: [asterisk-users] 1.6 and 1.8
On 12/28/2011 03:10 PM, Danny Nicholas wrote: Can somebody point me to an explanation from Kevin or Tzafir or someone else up the food chain explaining the differences/benefits of 1.6/1.8 vs 1.4/10.0? Every branch (1.0, 1.2, 1.4, 1.6.0, 1.6.1, 1.6.2, 1.8, 10) of Asterisk contains new features that previous branches did not have. Many of these changes are documented in http://svnview.digium.com/svn/asterisk/branches/10/CHANGES Each branch of Asterisk has a lifecycle, which is documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions. As you can see, 1.8 and 10 are the currently supported branches. 1.4 and 1.6.2 are in security maintenance mode, which means that the only issues that will be fixed are security issues. They will both be EOL in April 2012, and will no longer receive any updates. Short version: If you aren't already using Asterisk 1.8 or higher, you really should be - and soon. -- _ -- 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] MFCR2 Long distance calls not connected
Calls to long distance get disconnected before answer. Telco: Alestra Country: Mexico System: Elastix 2.2 Digital Card: Digium TE122 Log: [Dec 28 14:37:44] VERBOSE[4586] pbx.c: -- Executing [+525552622900@default:1] Set(SIP/OCS_TRUNK-01bf, EXT=015552622900) in new stack [Dec 28 14:37:44] VERBOSE[4586] pbx.c: -- Executing [+525552622900@default:2] Dial(SIP/OCS_TRUNK-01bf, DAHDI/g1/015552622900,60) in new stack [Dec 28 14:37:44] VERBOSE[4586] app_dial.c: -- Called DAHDI/g1/015552622900 [Dec 28 14:37:44] DEBUG[4586] chan_dahdi.c: bits changed in chan 1 [Dec 28 14:37:53] DEBUG[4586] chan_dahdi.c: disconnecting MFC/R2 call on chan 1 [Dec 28 14:37:53] DEBUG[4586] chan_dahdi.c: ast cause 0 resulted in openr2 cause 6/Normal Clearing [Dec 28 14:37:53] VERBOSE[4586] chan_dahdi.c: -- Hungup 'DAHDI/1-1' [Dec 28 14:37:53] VERBOSE[4586] pbx.c: == Spawn extension (default, +525552622900, 2) exited non-zero on 'SIP/OCS_TRUNK-01bf' [Dec 28 14:37:53] VERBOSE[9190] chan_dahdi.c: MFC/R2 call end on channel 1 Found this email list, but I think is too old. http://www.mail-archive.com/asterisk-users@lists.digium.com/msg205765.html No imprima este mail a menos que sea absolutamente necesario. Aviso de Confidencialidad: Este mensaje, incluyendo cualquier adjunto, es para uso exclusivo de el/los destinatario/s y puede contener información confidencial y/o privilegiada. Si usted no es uno de los destinatarios legítimos, por favor contacte al remitente y elimine el mensaje. Está prohibido utilizar la información contenida en el mismo sin autorización expresa. Confidentiality Notice: This message, including any attachments, is intended only for the use of the named recipient(s) and may contain confidential and/or privileged information. If you are not one of the intended recipients, please contact the sender and delete this message. Any unauthorized use of the information it contains is prohibited. -- _ -- 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] Monitor Command Records separate channales
Try
# grep 'onitor' /var/log/asterisk/messages
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
but i tiried these commands and I didnt find anything about Monitor
[root@c-24-1-71-68 asterisk]# grep -R 'Monitor' *
[root@c-24-1-71-68 asterisk]# grep -R 'monitor' *
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 3:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales
I would wager that your setup dumps what would normally be in /v/l/a/full into
/v/l/a/messages
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
see attached ...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 3:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Even using Queue there should still be a /var/log/asterisk/full that records
the Monitor then the following Queue/Dial commands. What is in your
/var/log/asterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
My call happens with a queue , there is no full file but there is queue and
queue is useless, can u give me unix command to search all log files and print
moniter line?
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:
Asterisk -vvvrc
Is how you would get it live
After the fact you might find it in /var/log/asterisk/full
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 3:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Can u plz tell me how , I forgot how to run asterisk cli
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
Can you post a CLI output of the Monitor output? I'm supposing that
something in your $(STRFTIME) string might be eating the M option.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version,
my problem is not mixing It's the same file but inside that file
two seperate records first callers then reciever
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
According to the monitor documentation, the format you specified
should be calling SOX and mixing on call completion. What versions
of SOX and Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command this is how
I use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-
%
m
-%d_%H
:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Re: [asterisk-users] 1.6 and 1.8
I understand the end of life issue. What I fail to understand is that if 1.8 is the Cadillac of Asterisk, why did they make 10.0 and why does 1.8 have so many bugs (just what I read here, not from my actual experience)? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Wednesday, December 28, 2011 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.6 and 1.8 On 12/28/2011 03:10 PM, Danny Nicholas wrote: Can somebody point me to an explanation from Kevin or Tzafir or someone else up the food chain explaining the differences/benefits of 1.6/1.8 vs 1.4/10.0? Every branch (1.0, 1.2, 1.4, 1.6.0, 1.6.1, 1.6.2, 1.8, 10) of Asterisk contains new features that previous branches did not have. Many of these changes are documented in http://svnview.digium.com/svn/asterisk/branches/10/CHANGES Each branch of Asterisk has a lifecycle, which is documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions. As you can see, 1.8 and 10 are the currently supported branches. 1.4 and 1.6.2 are in security maintenance mode, which means that the only issues that will be fixed are security issues. They will both be EOL in April 2012, and will no longer receive any updates. Short version: If you aren't already using Asterisk 1.8 or higher, you really should be - and soon. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor Command Records separate channales
It got stuck ...
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:٢٩ م, Danny Nicholas da...@debsinc.com wrote:
Try
# grep 'onitor' /var/log/asterisk/messages
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
but i tiried these commands and I didnt find anything about Monitor
[root@c-24-1-71-68 asterisk]# grep -R 'Monitor' *
[root@c-24-1-71-68 asterisk]# grep -R 'monitor' *
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 3:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales
I would wager that your setup dumps what would normally be in /v/l/a/full
into /v/l/a/messages
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
see attached ...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 3:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Monitor Command Records separate channales
Even using Queue there should still be a /var/log/asterisk/full that records
the Monitor then the following Queue/Dial commands. What is in your
/var/log/asterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate channales
My call happens with a queue , there is no full file but there is queue and
queue is useless, can u give me unix command to search all log files and
print moniter line?
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:
Asterisk -vvvrc
Is how you would get it live
After the fact you might find it in /var/log/asterisk/full
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 3:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Can u plz tell me how , I forgot how to run asterisk cli
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
Can you post a CLI output of the Monitor output? I'm supposing that
something in your $(STRFTIME) string might be eating the M option.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version,
my problem is not mixing It's the same file but inside that file
two seperate records first callers then reciever
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
According to the monitor documentation, the format you specified
should be calling SOX and mixing on call completion. What versions
of SOX and Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Wednesday, December 28, 2011 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor Command Records separate
channales
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command this is how
I use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-
%
m
-%d_%H
:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the same?
Re: [asterisk-users] 1.6 and 1.8
The UPGRADE*.txt files included with the Asterisk tarballs give a nice summery of the major changes between each Asterisk verison. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Wednesday, December 28, 2011 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.6 and 1.8 On 12/28/2011 03:10 PM, Danny Nicholas wrote: Can somebody point me to an explanation from Kevin or Tzafir or someone else up the food chain explaining the differences/benefits of 1.6/1.8 vs 1.4/10.0? Every branch (1.0, 1.2, 1.4, 1.6.0, 1.6.1, 1.6.2, 1.8, 10) of Asterisk contains new features that previous branches did not have. Many of these changes are documented in http://svnview.digium.com/svn/asterisk/branches/10/CHANGES Each branch of Asterisk has a lifecycle, which is documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions. As you can see, 1.8 and 10 are the currently supported branches. 1.4 and 1.6.2 are in security maintenance mode, which means that the only issues that will be fixed are security issues. They will both be EOL in April 2012, and will no longer receive any updates. Short version: If you aren't already using Asterisk 1.8 or higher, you really should be - and soon. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Function TESTTIME example
Hi,
This function sets TESTTIME global variable and if TESTTIME variable is set,
then GoToIfTime use time from this variable.
On 2011.12.28, at 17:28, Olivier oza_4...@yahoo.fr wrote:
Hi,
Thanks for replying.
I'm afraid this :
[foobar]
exten = 123,1,Verbose(0,Into context ${CONTEXT})
exten = 123,n,Verbose(0,Time is ${STRFTIME()})
exten = 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
exten = 123,n,Verbose(0,Time is ${STRFTIME()})
exten = 123,n,HangUp()
... gives this:
-- Executing [123@foobar:1] Verbose(SIP/7005-006b, 0,Into
context foobar) in new stack
Into context foobar
-- Executing [123@foobar:2] Verbose(SIP/7005-006b, 0,Time
is Wed Dec 28 16:25:59 2011) in new stack
Time is Wed Dec 28 16:25:59 2011
-- Executing [123@foobar:3] Set(SIP/7005-006b,
TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius) in new stack
-- Executing [123@foobar:4] Verbose(SIP/7005-006b, 0,Time
is Wed Dec 28 16:25:59 2011) in new stack
Time is Wed Dec 28 16:25:59 2011
-- Executing [123@foobar:5] Hangup(SIP/7005-006b, ) in new stack
Do you see the same behaviour ?
2011/12/28, Mindaugas Jasiulis mindaugas.jasiu...@mediafon.lt:
Hi,
I do not know, whether this is the best way to use TESTTIME function, but
for me it is working in that way:
exten = 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
OR
You can use this:
Set(__TESTTIME=${STRPTIME(2011-12-25 18:00:00,Europe/Vilnius,%Y-%m-%d
%H:%M:%S)})
Best regards,
Mindaugas
Hi,
Has someone a dialplan example using TESTTIME function (see core show
function TESTTIME) ?
I'm only getting replies such as Function TESTTIME cannot be read.
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Monitor Command Records separate channales
I attached log, but there is nothing unusual in it ...all normal ... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] Sent: Wednesday, December 28, 2011 4:06 PM To: Faraj Khasib Subject: Your message to asterisk-users awaits moderator approval Your mail to 'asterisk-users' with the subject RE: [asterisk-users] Monitor Command Records separate channales Is being held until the list moderator can review it for approval. The reason it is being held: Message body is too big: 1004233 bytes with a limit of 40 KB Either the message will get posted to the list, or you will receive notification of the moderator's decision. If you would like to cancel this posting, please visit the following URL: http://lists.digium.com/mailman/confirm/asterisk-users/7c086c1398347b43db2e2984127934cd8cbde5c4 -- _ -- 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] func_odbc not returning whole smalldatetime MS Sql field.
Hey All,
Odd thing. I am just trying to return the whole date time stamp from a
SMALLDATETIME field in a MS SQL server.
func_odbc.conf = readsql=SELECT DateCreated FROM [REDACTED] WHERE Code
= '${ARG1}'
Problem is I only get the first 15 back from the field. Like so...
Connected to Asterisk 1.8.6.0 currently running on [REDACTED]-dev (pid =
2240)
Verbosity is at least 3
[REDACTED]-dev*CLI odbc read ODBC_[REDACTED]-LOOKUP 104809 exec
DateCreated 2011-12-19 13:2
Returned 1 row. Query executed on handle 0 [asterisk-mssql-connector]
Notice how it only returns 2011-12-19 13:2 and not the rest of the
time... I have run the query on the SQL server and then from isql and it
works everytime leaving the only abstraction point Asterisk.
Any thoughts?
Thanks!!
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Re: [asterisk-users] Monitor Command Records separate channales
Un-top-posting, snarky comments inline... On Wed, 28 Dec 2011, Faraj Khasib wrote: I am trying to record Call, but when the call is done I have one file but the conversation inside it is separate into calls conversation and receiver its single file but separate recording, How can I make it mixed together so the conversation will be normal? (I don't understand. How can you have separate recordings in a single file?) On Wed, 28 Dec 2011, Faraj Khasib wrote: I installed SOX( it was not installed before). Will that solve my problem? if not what are the parameter for the mixMonitor Command this is how I use Monitor exten=6500,2,Monitor(...) is Mix Monitor will have the same? (You want me to guess if installing sox will solve your problem?) (Too lazy to look up the mixmonitor() command?) On Wed, 28 Dec 2011, Faraj Khasib wrote: Asterisk 1.6.2 but sox I don't know but now it is the latest version, my problem is not mixing It's the same file but inside that file two seperate records first callers then reciever (Finally we get some details...) On Wed, 28 Dec 2011, Faraj Khasib wrote: Can u plz tell me how , I forgot how to run asterisk cli (Lazy or in over his head?) On Wed, 28 Dec 2011, Faraj Khasib wrote: My call happens with a queue , there is no full file but there is queue and queue is useless, can u give me unix command to search all log files and print moniter line? (Don't understand the question and seeing the lazy/unqualified thing again.) On Wed, 28 Dec 2011, Faraj Khasib wrote: I already searched using grep for the monitor word ... It doesn't exists (Don't have a lot of confidence in this statement.) On Wed, 28 Dec 2011, Faraj Khasib wrote: but i tiried these commands and I didnt find anything about Monitor [root@c-24-1-71-68 asterisk]# grep -R 'Monitor' * [root@c-24-1-71-68 asterisk]# grep -R 'monitor' * (Should brush up on grep's command line parameters. I wonder what '*' evaluates to. I hope he isn't really logged in as root) On Wed, 28 Dec 2011, Faraj Khasib wrote: It got stuck ... (I wonder what this means. What is he talking about?) On Wed, 28 Dec 2011, Faraj Khasib wrote: I attached log, but there is nothing unusual in it ...all normal ... (No file attached. Maybe he should read the error message the list manager returned. Little confidence in his assessment as 'normal') Please take a moment to learn list etiquette. 1) Please don't top post. 2) Please don't ask questions you could easily google yourself. 3) Please learn basic Unix commands like 'grep'. 4) Please take the time to form unambiguous questions. 5) Please include sufficient detail so we don't have to keep guessing what is going on. It appears (from your 3rd post) that your problem is that the monitor() application is concatenating both 'legs' of the call into a single file -- meaning that when you play the single recorded file you hear the entire conversation from the caller's side and then you hear the entire conversation from the callee's side. Kind of like: Callee) Hello? Callee) Fine, but I really have no clue what I'm doing. Callee) Never heard of it. Besides all these schmucks on the AU list like reading basic questions and spoon-feeding me the answers. Callee) There's a quota? Callee) How many questions do I have left? Callee) Steve? Callee) Hello? Callee) Hmmm. I must have a problem with my upstream provider... Caller) Hey Faraj, how ya doing? Caller) Sorry to hear that. Have you ever tried Google? Caller) Hmmm. Have you burned through your newbie question quota yet? Caller) Yep. It's not set in stone, but if you keep at it without showing you're putting in any effort, everybody will figure you out and ignore you. Instead of: Callee) Hello? Caller) Hey Faraj, how ya doing? Callee) Fine, but I really have no clue what I'm doing. Caller) Sorry to hear that. Have you ever tried Google? Callee) Never heard of it. Besides all these schmucks on the AU list like reading basic questions and spoon-feeding me the answers. Caller) Hmmm. Have you burned through your newbie question quota yet? Callee) There's a quota? Caller) Yep. It's not set in stone, but if you keep at it without showing you're putting in any effort, everybody will figure you out and ignore you. Callee) How many questions do I have left? Callee) Steve? Callee) Hello? Callee) Hmmm. I must have a problem with my upstream provider... This would be a novel problem since in over 8 years of reading this list I've never seen anybody else report it. Does this happen with all calls or only calls that are queued to an agent? Are you fiddling with MONITOR_EXEC or MONITOR_EXEC_ARGS? Can you post a link to a sample recorded file along with the console log for the call? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Re: [asterisk-users] Monitor Command Records separate channales
Hello, Do you use monitor?, because in asterisk 1.4 to new versions, It's use mixmonitor, in asterisk 1.2 had this mistake. Regards On Wed, Dec 28, 2011 at 10:11 PM, Steve Edwards asterisk@sedwards.comwrote: Un-top-posting, snarky comments inline... On Wed, 28 Dec 2011, Faraj Khasib wrote: I am trying to record Call, but when the call is done I have one file but the conversation inside it is separate into calls conversation and receiver its single file but separate recording, How can I make it mixed together so the conversation will be normal? (I don't understand. How can you have separate recordings in a single file?) On Wed, 28 Dec 2011, Faraj Khasib wrote: I installed SOX( it was not installed before). Will that solve my problem? if not what are the parameter for the mixMonitor Command this is how I use Monitor exten=6500,2,Monitor(...) is Mix Monitor will have the same? (You want me to guess if installing sox will solve your problem?) (Too lazy to look up the mixmonitor() command?) On Wed, 28 Dec 2011, Faraj Khasib wrote: Asterisk 1.6.2 but sox I don't know but now it is the latest version, my problem is not mixing It's the same file but inside that file two seperate records first callers then reciever (Finally we get some details...) On Wed, 28 Dec 2011, Faraj Khasib wrote: Can u plz tell me how , I forgot how to run asterisk cli (Lazy or in over his head?) On Wed, 28 Dec 2011, Faraj Khasib wrote: My call happens with a queue , there is no full file but there is queue and queue is useless, can u give me unix command to search all log files and print moniter line? (Don't understand the question and seeing the lazy/unqualified thing again.) On Wed, 28 Dec 2011, Faraj Khasib wrote: I already searched using grep for the monitor word ... It doesn't exists (Don't have a lot of confidence in this statement.) On Wed, 28 Dec 2011, Faraj Khasib wrote: but i tiried these commands and I didnt find anything about Monitor [root@c-24-1-71-68 asterisk]# grep -R 'Monitor' * [root@c-24-1-71-68 asterisk]# grep -R 'monitor' * (Should brush up on grep's command line parameters. I wonder what '*' evaluates to. I hope he isn't really logged in as root) On Wed, 28 Dec 2011, Faraj Khasib wrote: It got stuck ... (I wonder what this means. What is he talking about?) On Wed, 28 Dec 2011, Faraj Khasib wrote: I attached log, but there is nothing unusual in it ...all normal ... (No file attached. Maybe he should read the error message the list manager returned. Little confidence in his assessment as 'normal') Please take a moment to learn list etiquette. 1) Please don't top post. 2) Please don't ask questions you could easily google yourself. 3) Please learn basic Unix commands like 'grep'. 4) Please take the time to form unambiguous questions. 5) Please include sufficient detail so we don't have to keep guessing what is going on. It appears (from your 3rd post) that your problem is that the monitor() application is concatenating both 'legs' of the call into a single file -- meaning that when you play the single recorded file you hear the entire conversation from the caller's side and then you hear the entire conversation from the callee's side. Kind of like: Callee) Hello? Callee) Fine, but I really have no clue what I'm doing. Callee) Never heard of it. Besides all these schmucks on the AU list like reading basic questions and spoon-feeding me the answers. Callee) There's a quota? Callee) How many questions do I have left? Callee) Steve? Callee) Hello? Callee) Hmmm. I must have a problem with my upstream provider... Caller) Hey Faraj, how ya doing? Caller) Sorry to hear that. Have you ever tried Google? Caller) Hmmm. Have you burned through your newbie question quota yet? Caller) Yep. It's not set in stone, but if you keep at it without showing you're putting in any effort, everybody will figure you out and ignore you. Instead of: Callee) Hello? Caller) Hey Faraj, how ya doing? Callee) Fine, but I really have no clue what I'm doing. Caller) Sorry to hear that. Have you ever tried Google? Callee) Never heard of it. Besides all these schmucks on the AU list like reading basic questions and spoon-feeding me the answers. Caller) Hmmm. Have you burned through your newbie question quota yet? Callee) There's a quota? Caller) Yep. It's not set in stone, but if you keep at it without showing you're putting in any effort, everybody will figure you out and ignore you. Callee) How many questions do I have left? Callee) Steve? Callee) Hello? Callee) Hmmm. I must have a problem with my upstream provider... This would be a novel problem since in over 8 years of reading this list I've never seen anybody else report it. Does this happen with all calls or only calls that are queued to an agent? Are you fiddling with MONITOR_EXEC or
Re: [asterisk-users] Interesting attack tonight fail2ban them
I just realized there is no IP (host) in the message line, so no way for
fail2ban to catch it.
Other suggestions? Or will I have to code something into my dialplan
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Furey
[andrew.fu...@gmail.com]
Sent: Wednesday, December 28, 2011 11:37 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Interesting attack tonight fail2ban them
On 29 December 2011 12:07, Michelle Dupuis mdup...@ocg.ca wrote:
I thought that it might be worth adding a line to my fail2ban filter, but am
looking for a hand with the regex. I have come up with:
NOTICE.* .*: Call from '' to extension '.*' rejected because
extension not found
but I realize that anyone misdialling a valid extension a few times gets cut
off. Can someone suggest an improvement? (How could I limit this to 4 or
more digits dialled for example?)
[ Caveat - I have never used fail2ban ]
If it supports Perl-style regexps, you could do:
NOTICE.* .*: Call from '' to extension '[0-9]{4,}' rejected because
extension not found
That will do at least 4 digits.
Or the long way (Bash-style etc):
NOTICE.* .*: Call from '' to extension '[0-9][0-9][0-9][0-9][0-9]*'
rejected because extension not found
HTH,
Andrew
--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
-- Bill Garrett
--
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Re: [asterisk-users] Client - registers but unreachable
Here is more of a SIP debug log: As you can see Asterisk retries four times but I assume the softphone is not responding? --- Really destroying SIP dialog '637b0e9777c88caa16a5a70b5a8984fe@172.31.253.4'mailto:'637b0e9777c88caa16a5a70b5a8984fe@172.31.253.4' Method: OPTIONS Reliably Transmitting (no NAT) to 172.31.254.53:9653: OPTIONS sip:230bb@172.31.254.53:9653 SIP/2.0 Via: SIP/2.0/UDP 172.31.253.4:5060;branch=z9hG4bK6953e396;rport From: asterisk sip:asterisk@172.31.253.4;tag=as78f74756 To: sip:230bb@172.31.254.53:9653 Contact: sip:asterisk@172.31.253.4 Call-ID: 1173f2d330ff00674ef754f863f9cbae@172.31.253.4mailto:1173f2d330ff00674ef754f863f9cbae@172.31.253.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Dec 2011 04:22:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #1 (no NAT) to 172.31.254.53:9653: OPTIONS sip:230bb@172.31.254.53:9653 SIP/2.0 Via: SIP/2.0/UDP 172.31.253.4:5060;branch=z9hG4bK6953e396;rport From: asterisk sip:asterisk@172.31.253.4;tag=as78f74756 To: sip:230bb@172.31.254.53:9653 Contact: sip:asterisk@172.31.253.4 Call-ID: 1173f2d330ff00674ef754f863f9cbae@172.31.253.4mailto:1173f2d330ff00674ef754f863f9cbae@172.31.253.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Dec 2011 04:22:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #2 (no NAT) to 172.31.254.53:9653: OPTIONS sip:230bb@172.31.254.53:9653 SIP/2.0 Via: SIP/2.0/UDP 172.31.253.4:5060;branch=z9hG4bK6953e396;rport From: asterisk sip:asterisk@172.31.253.4;tag=as78f74756 To: sip:230bb@172.31.254.53:9653 Contact: sip:asterisk@172.31.253.4 Call-ID: 1173f2d330ff00674ef754f863f9cbae@172.31.253.4mailto:1173f2d330ff00674ef754f863f9cbae@172.31.253.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Dec 2011 04:22:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- -- Remote UNIX connection -- Remote UNIX connection disconnected Retransmitting #3 (no NAT) to 172.31.254.53:9653: OPTIONS sip:230bb@172.31.254.53:9653 SIP/2.0 Via: SIP/2.0/UDP 172.31.253.4:5060;branch=z9hG4bK6953e396;rport From: asterisk sip:asterisk@172.31.253.4;tag=as78f74756 To: sip:230bb@172.31.254.53:9653 Contact: sip:asterisk@172.31.253.4 Call-ID: 1173f2d330ff00674ef754f863f9cbae@172.31.253.4mailto:1173f2d330ff00674ef754f863f9cbae@172.31.253.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Dec 2011 04:22:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #4 (no NAT) to 172.31.254.53:9653: OPTIONS sip:230bb@172.31.254.53:9653 SIP/2.0 Via: SIP/2.0/UDP 172.31.253.4:5060;branch=z9hG4bK6953e396;rport From: asterisk sip:asterisk@172.31.253.4;tag=as78f74756 To: sip:230bb@172.31.254.53:9653 Contact: sip:asterisk@172.31.253.4 Call-ID: 1173f2d330ff00674ef754f863f9cbae@172.31.253.4mailto:1173f2d330ff00674ef754f863f9cbae@172.31.253.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Dec 2011 04:22:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 -- _ -- 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] Client - registers but unreachable
The BB is using wifi, on the same subnet as the asterisk server so no need for NAT. There is no keep alive option on the softphone (very simplistic settings) Thanks -- _ -- 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] Interesting attack tonight fail2ban them
I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '6442032987219' rejected because extension not found. [2011-12-28 22:53:44] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '7442032987216' rejected because extension not found. [2011-12-28 22:53:46] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '8442032987216' rejected because extension not found. [2011-12-28 22:53:48] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '008442032987215' rejected because extension not found. [2011-12-28 22:53:50] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '007442032987218' rejected because extension not found. [2011-12-28 22:53:52] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '006442032987219' rejected because extension not found. [2011-12-28 22:53:54] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '005442032987216' rejected because extension not found. [2011-12-28 22:53:56] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '004442032987250' rejected because extension not found. I thought that it might be worth adding a line to my fail2ban filter, but am looking for a hand with the regex. I have come up with: NOTICE.* .*: Call from '' to extension '.*' rejected because extension not found but I realize that anyone misdialling a valid extension a few times gets cut off. Can someone suggest an improvement? (How could I limit this to 4 or more digits dialled for example?) Thanks! -- _ -- 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] Interesting attack tonight fail2ban them
Yes fail2ban is working fine. I did NOT have a filter for the rejected because extension not found line yet (I'm still working on it). Hoping for input on the regex. Thanks From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Rojas [crt.ro...@gmail.com] Sent: Wednesday, December 28, 2011 11:11 PM To: Asterisk Users List Subject: Re: [asterisk-users] Interesting attack tonight fail2ban them Hello, Do you set up, your logrotate in /etc/asterisk ? Do you test that your fail2ban work fine? Regards On Wed, Dec 28, 2011 at 11:07 PM, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28tel:%5B2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '6442032987219' rejected because extension not found. [2011-12-28tel:%5B2011-12-28 22:53:44] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '7442032987216' rejected because extension not found. [2011-12-28tel:%5B2011-12-28 22:53:46] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '8442032987216' rejected because extension not found. [2011-12-28tel:%5B2011-12-28 22:53:48] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '008442032987215' rejected because extension not found. [2011-12-28tel:%5B2011-12-28 22:53:50] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '007442032987218' rejected because extension not found. [2011-12-28tel:%5B2011-12-28 22:53:52] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '006442032987219' rejected because extension not found. [2011-12-28tel:%5B2011-12-28 22:53:54] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '005442032987216' rejected because extension not found. [2011-12-28tel:%5B2011-12-28 22:53:56] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '004442032987250' rejected because extension not found. I thought that it might be worth adding a line to my fail2ban filter, but am looking for a hand with the regex. I have come up with: NOTICE.* .*: Call from '' to extension '.*' rejected because extension not found but I realize that anyone misdialling a valid extension a few times gets cut off. Can someone suggest an improvement? (How could I limit this to 4 or more digits dialled for example?) Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6 and 1.8
On Wed, Dec 28, 2011 at 4:33 PM, Danny Nicholas da...@debsinc.com wrote: I understand the end of life issue. What I fail to understand is that if 1.8 is the Cadillac of Asterisk, why did they make 10.0 and why does 1.8 have so many bugs (just what I read here, not from my actual experience)? Once released a version will only have bug and security fixes. New features go into trunk to be included in the next version. Asterisk has long term support releases like 1.4 and 1.8 and standard releases like 1.6 and 10. This model is no different than other software like Ubuntu. Even though a series only has bug and security fixes I have found regressions occur between point releases. Just make sure to test thoroughly before putting a system in production. I tend to stick with a version until I need the features in a newer version or back porting a security fix becomes overly involved. I have been running 1.8.7 with a few fixes back ported from the 1.8.8 release candidate for the last 2.5 months. The system processes around 4,000 calls per day over PRIs for 250 Polycom phones. Previously I was running 1.6.1.18 with a bunch of back ports for fixes and features. Overall it was stable but every few months I had an issue where a channel would get hung. When this happened core show channels would crash the console and I would eventually have to restart Asterisk. Ryan -- _ -- 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] Interesting attack tonight fail2ban them
On 29 December 2011 12:07, Michelle Dupuis mdup...@ocg.ca wrote:
I thought that it might be worth adding a line to my fail2ban filter, but am
looking for a hand with the regex. I have come up with:
NOTICE.* .*: Call from '' to extension '.*' rejected because
extension not found
but I realize that anyone misdialling a valid extension a few times gets cut
off. Can someone suggest an improvement? (How could I limit this to 4 or
more digits dialled for example?)
[ Caveat - I have never used fail2ban ]
If it supports Perl-style regexps, you could do:
NOTICE.* .*: Call from '' to extension '[0-9]{4,}' rejected because
extension not found
That will do at least 4 digits.
Or the long way (Bash-style etc):
NOTICE.* .*: Call from '' to extension '[0-9][0-9][0-9][0-9][0-9]*'
rejected because extension not found
HTH,
Andrew
--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
-- Bill Garrett
--
_
-- 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:
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Re: [asterisk-users] Interesting attack tonight fail2ban them
Hello, Do you set up, your logrotate in /etc/asterisk ? Do you test that your fail2ban work fine? Regards On Wed, Dec 28, 2011 at 11:07 PM, Michelle Dupuis mdup...@ocg.ca wrote: I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '6442032987219' rejected because extension not found. [2011-12-28 22:53:44] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '7442032987216' rejected because extension not found. [2011-12-28 22:53:46] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '8442032987216' rejected because extension not found. [2011-12-28 22:53:48] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '008442032987215' rejected because extension not found. [2011-12-28 22:53:50] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '007442032987218' rejected because extension not found. [2011-12-28 22:53:52] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '006442032987219' rejected because extension not found. [2011-12-28 22:53:54] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '005442032987216' rejected because extension not found. [2011-12-28 22:53:56] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '004442032987250' rejected because extension not found. I thought that it might be worth adding a line to my fail2ban filter, but am looking for a hand with the regex. I have come up with: NOTICE.* .*: Call from '' to extension '.*' rejected because extension not found but I realize that anyone misdialling a valid extension a few times gets cut off. Can someone suggest an improvement? (How could I limit this to 4 or more digits dialled for example?) Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Client - registers but unreachable
I have a softphone I'm trying on a blackberry, that registers on my Asterisk, can make outgoing calls, but can't receive calls. There is very little traffic with this phone (see debug below - as the phone registers), and sip show peers confirms it is unreachable. Any suggestions? Is this just a dumb client or do I need to tweak an asterisk setting? Thanks pbx*CLI sip debug peer 230bb Unable to get IP address of peer '230bb' The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. pbx*CLI -- Registered SIP '230bb' at 172.31.254.53 port 9653 expires 1800 [2011-12-28 23:11:09] NOTICE[9635]: chan_sip.c:15851 sip_poke_noanswer: Peer '230bb' is now UNREACHABLE! Last qualify: 0 pbx*CLI sip show peers Name/username HostDyn Nat ACL Port Status 230bb/bob 172.31.254.53D 9653 UNREACHABLE -- _ -- 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] Client - registers but unreachable
Hello, try to configure keep alive option on Softphone if there is. Regards -- _ -- 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] Client - registers but unreachable
Hello, Your blackberry sip client, works in your wifi network? or by blackberry internet? do you set nat=yes if your phone, register by internet? What is your sip.conf? Regards On Wed, Dec 28, 2011 at 11:16 PM, Michelle Dupuis mdup...@ocg.ca wrote: I have a softphone I'm trying on a blackberry, that registers on my Asterisk, can make outgoing calls, but can't receive calls. There is very little traffic with this phone (see debug below - as the phone registers), and sip show peers confirms it is unreachable. Any suggestions? Is this just a dumb client or do I need to tweak an asterisk setting? Thanks pbx*CLI sip debug peer 230bb Unable to get IP address of peer '230bb' The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. pbx*CLI -- Registered SIP '230bb' at 172.31.254.53 port 9653 expires 1800 [2011-12-28 23:11:09] NOTICE[9635]: chan_sip.c:15851 sip_poke_noanswer: Peer '230bb' is now UNREACHABLE! Last qualify: 0 pbx*CLI sip show peers Name/username HostDyn Nat ACL Port Status 230bb/bob 172.31.254.53D 9653 UNREACHABLE -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Client - registers but unreachable
On Wed, 2011-12-28 at 23:16 -0500, Michelle Dupuis wrote: I have a softphone I'm trying on a blackberry, that registers on my Asterisk, can make outgoing calls, but can't receive calls. There is very little traffic with this phone (see debug below - as the phone registers), and sip show peers confirms it is unreachable. Any suggestions? Is this just a dumb client or do I need to tweak an asterisk setting? Thanks pbx*CLI sip debug peer 230bb Unable to get IP address of peer '230bb' The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. pbx*CLI -- Registered SIP '230bb' at 172.31.254.53 port 9653 expires 1800 [2011-12-28 23:11:09] NOTICE[9635]: chan_sip.c:15851 sip_poke_noanswer: Peer '230bb' is now UNREACHABLE! Last qualify: 0 pbx*CLI sip show peers Name/username HostDyn Nat ACL Port Status 230bb/bob 172.31.254.53D 9653 UNREACHABLE -- 172.31.254.53 is an RFC1918 address. You need to enable NAT on this client. j -- _ -- 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] 1.6 and 1.8
I have been running 1.8.7 with a few fixes back ported from the 1.8.8 release candidate for the last 2.5 months. The system processes around 4,000 calls per day over PRIs for 250 Polycom phones. Previously I was running 1.6.1.18 with a bunch of back ports for fixes and features. Overall it was stable but every few months I had an issue where a channel would get hung. When this happened core show channels would crash the console and I would eventually have to restart Asterisk. Ryan What od you mean by, been running 1.8.7 with a few fixes back ported from the 1.8.8 release candidate. So, this is a version 1.8.7 release that you are using or a 1.8.8 or is this a mix of both that you come up with? Can you please be specific with fixes? Thanks -- _ -- 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] Interesting attack tonight fail2ban them
Hi Michelle, I just realized there is no IP (host) in the message line, so no way for fail2ban to catch it. Probably my understanding is limited, but it seems to me that they have already 'access' to your Asterisk for them to be able to try to make outgoing calls. Wouldn't it be better to make sure they get the usual errors like Registration from failed - no matching peer found? In other words, how did they get this far in the first place? Best regards, Jeroen Eeuwes -- _ -- 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] Interesting attack tonight fail2ban them
You mentioned the IP, 208.122.57.58, where did you get that from?
Following are the default for Asterisk 1.8 (It would be great to have
others input on this to strengthen this part of the filter):
failregex = Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Wrong password
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - No
matching peer found
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Device does not match ACL
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Username/auth name mismatch
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Peer
is not supposed to register
NOTICE.* HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' (from HOST)
NOTICE.* .*: Host HOST failed MD5 authentication for '.*' (.*)
VERBOSE.* logger.c: -- .*IP/HOST-.* Playing 'ss-noservice'
(language '.*')
Regards,
On Wed, Dec 28, 2011 at 11:50 PM, Michelle Dupuis mdup...@ocg.ca wrote:
I just realized there is no IP (host) in the message line, so no way for
fail2ban to catch it.
Other suggestions? Or will I have to code something into my dialplan
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Re: [asterisk-users] Function TESTTIME example
OK !
But AEL2's ifTime keyword do not use it, does it ?
2011/12/28, Mindaugas Jasiulis mindaugas.jasiu...@mediafon.lt:
Hi,
This function sets TESTTIME global variable and if TESTTIME variable is set,
then GoToIfTime use time from this variable.
On 2011.12.28, at 17:28, Olivier oza_4...@yahoo.fr wrote:
Hi,
Thanks for replying.
I'm afraid this :
[foobar]
exten = 123,1,Verbose(0,Into context ${CONTEXT})
exten = 123,n,Verbose(0,Time is ${STRFTIME()})
exten = 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
exten = 123,n,Verbose(0,Time is ${STRFTIME()})
exten = 123,n,HangUp()
... gives this:
-- Executing [123@foobar:1] Verbose(SIP/7005-006b, 0,Into
context foobar) in new stack
Into context foobar
-- Executing [123@foobar:2] Verbose(SIP/7005-006b, 0,Time
is Wed Dec 28 16:25:59 2011) in new stack
Time is Wed Dec 28 16:25:59 2011
-- Executing [123@foobar:3] Set(SIP/7005-006b,
TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius) in new stack
-- Executing [123@foobar:4] Verbose(SIP/7005-006b, 0,Time
is Wed Dec 28 16:25:59 2011) in new stack
Time is Wed Dec 28 16:25:59 2011
-- Executing [123@foobar:5] Hangup(SIP/7005-006b, ) in new
stack
Do you see the same behaviour ?
2011/12/28, Mindaugas Jasiulis mindaugas.jasiu...@mediafon.lt:
Hi,
I do not know, whether this is the best way to use TESTTIME function, but
for me it is working in that way:
exten = 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
OR
You can use this:
Set(__TESTTIME=${STRPTIME(2011-12-25 18:00:00,Europe/Vilnius,%Y-%m-%d
%H:%M:%S)})
Best regards,
Mindaugas
Hi,
Has someone a dialplan example using TESTTIME function (see core show
function TESTTIME) ?
I'm only getting replies such as Function TESTTIME cannot be read.
Regards
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Re: [asterisk-users] Function TESTTIME example
AEL2 ifTime use it too. AEL2 ifTime become the same GoToIfTime in the dialplan
:)
On Dec 29, 2011, at 8:40 AM, Olivier wrote:
OK !
But AEL2's ifTime keyword do not use it, does it ?
2011/12/28, Mindaugas Jasiulis mindaugas.jasiu...@mediafon.lt:
Hi,
This function sets TESTTIME global variable and if TESTTIME variable is set,
then GoToIfTime use time from this variable.
On 2011.12.28, at 17:28, Olivier oza_4...@yahoo.fr wrote:
Hi,
Thanks for replying.
I'm afraid this :
[foobar]
exten = 123,1,Verbose(0,Into context ${CONTEXT})
exten = 123,n,Verbose(0,Time is ${STRFTIME()})
exten = 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
exten = 123,n,Verbose(0,Time is ${STRFTIME()})
exten = 123,n,HangUp()
... gives this:
-- Executing [123@foobar:1] Verbose(SIP/7005-006b, 0,Into
context foobar) in new stack
Into context foobar
-- Executing [123@foobar:2] Verbose(SIP/7005-006b, 0,Time
is Wed Dec 28 16:25:59 2011) in new stack
Time is Wed Dec 28 16:25:59 2011
-- Executing [123@foobar:3] Set(SIP/7005-006b,
TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius) in new stack
-- Executing [123@foobar:4] Verbose(SIP/7005-006b, 0,Time
is Wed Dec 28 16:25:59 2011) in new stack
Time is Wed Dec 28 16:25:59 2011
-- Executing [123@foobar:5] Hangup(SIP/7005-006b, ) in new
stack
Do you see the same behaviour ?
2011/12/28, Mindaugas Jasiulis mindaugas.jasiu...@mediafon.lt:
Hi,
I do not know, whether this is the best way to use TESTTIME function, but
for me it is working in that way:
exten = 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
OR
You can use this:
Set(__TESTTIME=${STRPTIME(2011-12-25 18:00:00,Europe/Vilnius,%Y-%m-%d
%H:%M:%S)})
Best regards,
Mindaugas
Hi,
Has someone a dialplan example using TESTTIME function (see core show
function TESTTIME) ?
I'm only getting replies such as Function TESTTIME cannot be read.
Regards
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Re: [asterisk-users] DTMF Testing software to test IVR system
I originate calls from .call file and 1 channel I have at A server A and
another channel at B server.
*A server code is below:-*
exten = 43689956,1,Answer()
same = n,Wait(5)
same = n,SendDTMF(1)
same = n,NoOp(== ${CHANNEL(state)}== state)
same = n,wait(2)
same = n,SendDTMF(123456789012345#)
same = n,NoOp(== ${CHANNEL(state)}== state)
same = n,Hangup()
_ _
| A server | ___DTMF Send_= | B server |
|_| =--- Responce - |_|
*B server code is below:-*
At B server call come to 201 extension which is mention here..
exten = _20[1-6],1,Answer()
same = n,Ringing()
same = n,wait(2)
same = n,ExecIf($[$[${EXTEN}=201] || $[${EXTEN}=203]] ?*
AGI(/home/applications/ivr/nono2ivr/ivr/index_mvl.php))*
same = n,ExecIf($[${EXTEN}=202 || $[${EXTEN}=204] ||
$[${EXTEN}=205] ||
$[${EXTEN}=206]]?AGI(/home/applications/ivr/nono2ivr/ivr/index.php))
same = n,Hangup()
Now I can send the DTMF from A to B. But How I will get the responce at
server A. I checked all the channels variable but they didn't reply status
of B server channel. All information I will get of server A. Main problem
is that control reach to AGI and then I don't have any rights to do any
update or modification on AGI. So if I can work on request and responce
then it will be the last solution as per my knowledge.
Is this possible with the dialplan or I am just westing time?
On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger pabelan...@digium.comwrote:
On 11-12-28 03:25 AM, virendra bhati wrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can used
it?
like A script make the call by using localdail, .call file or any method.
And after landing the call we send dtmf to IVR system automatically as per
my script..
*extensions.conf:-*
exten = 1234,1,Answer()
same = n,Read(value,**pleasePress1forSupportPress2fo**
rHelp,1,,10)
same = n,NoOp(${value})
same = n,ExecIf($[${value}=1]?Goto(**suppot,1))
same = n,ExecIf($[${value}=2]?Goto(**help,1))
same = n,Hangup()
exten= support,1,Answer()
same = n,NoOp(you are at support section)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
We have DTMF based tests for the testsuite[1] that you could use.
[1]
http://svn.asterisk.org/svn/**testsuite/asterisk/trunk/http://svn.asterisk.org/svn/testsuite/asterisk/trunk/
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+91-8885268942
Software Engineer
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