date:20190111

Re: [asterisk-users] Detecting a fax

2019-01-11 Thread Neil Youngman


On 11/01/2019 09:19, Administrator TOOTAI wrote:

Le 11/01/2019 à 10:12, Neil Youngman a écrit :
A while back, I posted about detecting when a call was picked up by a 
fax machine. It was suggested that having a "fax" extension and 
"faxdetect=yes" would cause it to jump to the "fax" extension. This 
was not something I could get to work.


I have now created a very simple example. In sip.conf I have 
"faxdetect = yes". My example extension is:


[test]
;
; Voice test extension
;
exten => voicetest,1,NoOp()
 same => n,LOG(Notice,${CHANNEL}: Extension voiceout starting)
 same => n,LOG(Notice,${CHANNEL}: Starting Answer Machine Detection)
 same => n,AMD()
 same => n,LOG(Notice,${CHANNEL}: Answer Machine Detection 
${AMDSTATUS}/${AMDCAUSE})

 same => n,Playback(/var/lib/asterisk/sounds/en/demo-congrats)
 same => n,LOG(Notice,${CHANNEL}: Voice out extension complete)
 same => n(hangup),Hangup()


;
; Fax detected extension
;
exten => fax,1,NoOp()
 same => n,LOG(Notice,${CHANNEL}: Extension fax starting)
 same => n,LOG(Notice,${CHANNEL}: Fax Machine Detected)
 same => n,Playback(/var/lib/asterisk/sounds/en/silence/2)
 same => n,LOG(Notice,${CHANNEL}: Fax extension complete)
 same => n(hangup),Hangup()

and the logs show that calling a fax using the voiceout extension in 
context test does not result in the fax extension being triggered.


[Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Extension voiceout starting
[Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Starting Answer Machine Detection
[Jan 11 08:55:13] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Answer Machine Detection 
MACHINE/LONGGREETING-1500-1500
[Jan 11 08:55:44] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Voice out extension complete


Just for completeness this is how the call is originated, with a 
different phone number:


Action: Originate
ActionId: 1234567W001-125
Context: test
Exten: voicetest
Priority: 1
Channel: SIP/+441632660987@31.13.156.183:5060
Timeout: 6
Async: True

Can anyone offer any insight into why this isn't working?

Neil Youngman




You didn't ANSWER() the call


It's an outgoing call. I wouldn't expect to answer an outgoing call?

Neil


Neil Youngman 
Developer

Wirefast Limited

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See our messaging solutions at  http://www.wirefast.com/ 
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[asterisk-users] Can SIP domain help to set multiple SIP trunks between two boxes ?

2019-01-11 Thread Olivier

Hello,

I've been asked if it is possible or not to set several (10 or so) SIP
trunks between two boxes, one beeing an Avaya IPBX, the other being an
Asterisk 13 or 16 box (with either chan_sip or pjsip).

The reason behind this question come from billing requirements.
I'm not convinced yet setting several trunks is the proper answer to
specific billing requirements but my above question remains.

Before officially answering the root question, I'm planning to set a
demonstration between two Asterisk boxes, leaving trials with an Avaya IPBX
for a later step.

>From previous trials with chan_sip years ago, the main issues was matching
incoming calls to appropriate trunk as calls may come from the same IP/port
combination.

Now, I read very few lines about SIP domains and wondered if this could be
a mean to set several trunks without touching IP/port settings.

What do you think of this ?
Is it possible for Asterisk to send outbound and receive inbound using SIP
domain syntax in SIP messages instead of IP values ?

Best regards
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Re: [asterisk-users] Detecting a fax

2019-01-11 Thread Administrator TOOTAI


Le 11/01/2019 à 10:12, Neil Youngman a écrit :
A while back, I posted about detecting when a call was picked up by a 
fax machine. It was suggested that having a "fax" extension and 
"faxdetect=yes" would cause it to jump to the "fax" extension. This was 
not something I could get to work.


I have now created a very simple example. In sip.conf I have "faxdetect 
= yes". My example extension is:


[test]
;
; Voice test extension
;
exten => voicetest,1,NoOp()
     same => n,LOG(Notice,${CHANNEL}: Extension voiceout starting)
     same => n,LOG(Notice,${CHANNEL}: Starting Answer Machine Detection)
     same => n,AMD()
     same => n,LOG(Notice,${CHANNEL}: Answer Machine Detection 
${AMDSTATUS}/${AMDCAUSE})

     same => n,Playback(/var/lib/asterisk/sounds/en/demo-congrats)
     same => n,LOG(Notice,${CHANNEL}: Voice out extension complete)
     same => n(hangup),Hangup()


;
; Fax detected extension
;
exten => fax,1,NoOp()
     same => n,LOG(Notice,${CHANNEL}: Extension fax starting)
     same => n,LOG(Notice,${CHANNEL}: Fax Machine Detected)
     same => n,Playback(/var/lib/asterisk/sounds/en/silence/2)
     same => n,LOG(Notice,${CHANNEL}: Fax extension complete)
     same => n(hangup),Hangup()

and the logs show that calling a fax using the voiceout extension in 
context test does not result in the fax extension being triggered.


[Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Extension voiceout starting
[Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Starting Answer Machine Detection
[Jan 11 08:55:13] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Answer Machine Detection 
MACHINE/LONGGREETING-1500-1500
[Jan 11 08:55:44] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Voice out extension complete


Just for completeness this is how the call is originated, with a 
different phone number:


Action: Originate
ActionId: 1234567W001-125
Context: test
Exten: voicetest
Priority: 1
Channel: SIP/+441632660987@31.13.156.183:5060
Timeout: 6
Async: True

Can anyone offer any insight into why this isn't working?

Neil Youngman




You didn't ANSWER() the call

--
Daniel

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[asterisk-users] Detecting a fax

2019-01-11 Thread Neil Youngman

A while back, I posted about detecting when a call was picked up by a 
fax machine. It was suggested that having a "fax" extension and 
"faxdetect=yes" would cause it to jump to the "fax" extension. This was 
not something I could get to work.


I have now created a very simple example. In sip.conf I have "faxdetect 
= yes". My example extension is:


[test]
;
; Voice test extension
;
exten => voicetest,1,NoOp()
same => n,LOG(Notice,${CHANNEL}: Extension voiceout starting)
same => n,LOG(Notice,${CHANNEL}: Starting Answer Machine Detection)
same => n,AMD()
same => n,LOG(Notice,${CHANNEL}: Answer Machine Detection 
${AMDSTATUS}/${AMDCAUSE})

same => n,Playback(/var/lib/asterisk/sounds/en/demo-congrats)
same => n,LOG(Notice,${CHANNEL}: Voice out extension complete)
same => n(hangup),Hangup()


;
; Fax detected extension
;
exten => fax,1,NoOp()
same => n,LOG(Notice,${CHANNEL}: Extension fax starting)
same => n,LOG(Notice,${CHANNEL}: Fax Machine Detected)
same => n,Playback(/var/lib/asterisk/sounds/en/silence/2)
same => n,LOG(Notice,${CHANNEL}: Fax extension complete)
same => n(hangup),Hangup()

and the logs show that calling a fax using the voiceout extension in 
context test does not result in the fax extension being triggered.


[Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Extension voiceout starting
[Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Starting Answer Machine Detection
[Jan 11 08:55:13] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Answer Machine Detection 
MACHINE/LONGGREETING-1500-1500
[Jan 11 08:55:44] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Voice out extension complete


Just for completeness this is how the call is originated, with a 
different phone number:


Action: Originate
ActionId: 1234567W001-125
Context: test
Exten: voicetest
Priority: 1
Channel: SIP/+441632660987@31.13.156.183:5060
Timeout: 6
Async: True

Can anyone offer any insight into why this isn't working?

Neil Youngman



Neil Youngman 
Developer

Wirefast Limited

Wirefast provides secure corporate messaging services.
See our messaging solutions at  http://www.wirefast.com/ 
Please consider the environment.

Does this email or attachment need to be printed?
This message contains confidential information and is intended only 
for the individual named. If you are not the named addressee you 
should not disseminate, distribute or copy this email. Please 
notify the sender immediately by email if you have received this 
email by mistake and delete this email from your system.


Any views or opinions are solely those of the author
and do not necessarily represent those of Wirefast Limited

Email transmission cannot be guaranteed to be secure or error-free 
as information could be intercepted, corrupted, lost, destroyed, 
arrive late or incomplete, or contain viruses. The sender therefore 
does not accept liability for any errors or omissions in the contents 
of this message which arise as a result of email transmission.

Wirefast Limited is registered in England & Wales
Company number: 03865860
Registered Office: 7/10 Chandos Street, Cavendish Square, London, W1G 9DQ

Wirefast definitions of classification can be found here: 
www.wirefast.com/classifications
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Re: [asterisk-users] Detecting a fax

2019-01-11 Thread Neil Youngman


On 11/01/2019 10:08, Administrator TOOTAI wrote:

I don't understand your goal. You want to send or receive fax?


I'm attempting to deliver a voice message. If the message has been 
mistakenly sent to a fax I want to detect that it's a fax and report that.


Neil Youngman


Neil Youngman 
Developer

Wirefast Limited

Wirefast provides secure corporate messaging services.
See our messaging solutions at  http://www.wirefast.com/ 
Please consider the environment.

Does this email or attachment need to be printed?
This message contains confidential information and is intended only 
for the individual named. If you are not the named addressee you 
should not disseminate, distribute or copy this email. Please 
notify the sender immediately by email if you have received this 
email by mistake and delete this email from your system.


Any views or opinions are solely those of the author
and do not necessarily represent those of Wirefast Limited

Email transmission cannot be guaranteed to be secure or error-free 
as information could be intercepted, corrupted, lost, destroyed, 
arrive late or incomplete, or contain viruses. The sender therefore 
does not accept liability for any errors or omissions in the contents 
of this message which arise as a result of email transmission.

Wirefast Limited is registered in England & Wales
Company number: 03865860
Registered Office: 7/10 Chandos Street, Cavendish Square, London, W1G 9DQ

Wirefast definitions of classification can be found here: 
www.wirefast.com/classifications
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 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Detecting a fax

2019-01-11 Thread Neil Youngman


On 11/01/2019 10:04, Floimair Florian wrote:

I would guess from your explanation that the "outgoing" call somehow ends up in 
your Asterisk machine again, either at the voicetest or fax extension.
You don't answer it in either of the extensions.
That's what TOOTAI meant.


No, it's not looped back to the same asterisk extension. For this 
particular test I'm testing against a hardware fax machine plugged into 
a standard phone line.



If this is done in another extension, than this part of the Dialplan is missing 
in your post. So we can only guess what's going on.
Either post the full dialplan or add some log or cli output (with verbose set 
to at least 1) so we can have a chance of seeing what's going on.


I attach some CLI output.

Neil


Neil Youngman 
Developer

Wirefast Limited

Wirefast provides secure corporate messaging services.
See our messaging solutions at  http://www.wirefast.com/ 
Please consider the environment.

Does this email or attachment need to be printed?
This message contains confidential information and is intended only 
for the individual named. If you are not the named addressee you 
should not disseminate, distribute or copy this email. Please 
notify the sender immediately by email if you have received this 
email by mistake and delete this email from your system.


Any views or opinions are solely those of the author
and do not necessarily represent those of Wirefast Limited

Email transmission cannot be guaranteed to be secure or error-free 
as information could be intercepted, corrupted, lost, destroyed, 
arrive late or incomplete, or contain viruses. The sender therefore 
does not accept liability for any errors or omissions in the contents 
of this message which arise as a result of email transmission.

Wirefast Limited is registered in England & Wales
Company number: 03865860
Registered Office: 7/10 Chandos Street, Cavendish Square, London, W1G 9DQ

Wirefast definitions of classification can be found here: 
www.wirefast.com/classifications
[Kswasp6*CLI> 
[0K[1;30m  == [0mUsing SIP RTP CoS mark 5

[Kswasp6*CLI> 
[0K[1;30m-- [0mCalled +441632900987@10.11.12.13:5060

[Kswasp6*CLI> 
[0K[1;30m-- [0mSIP/10.11.12.13:5060-00f8 is making progress

[Kswasp6*CLI> 
[0K[1;30m-- [0mSIP/10.11.12.13:5060-00f8 answered

[Kswasp6*CLI> 
[0K[1;30m-- [0mExecuting [voicetest@test:1] 
[1;36mNoOp[0m("[1;35mSIP/10.11.12.13:5060-00f8[0m", "[1;35m[0m") in 
new stack

[Kswasp6*CLI> 
[0K[1;30m-- [0mExecuting [voicetest@test:2] 
[1;36mLog[0m("[1;35mSIP/10.11.12.13:5060-00f8[0m", 
"[1;35mNotice,SIP/10.11.12.13:5060-00f8: Extension voiceout starting[0m") 
in new stack

[Kswasp6*CLI> 
[0K[Jan 11 10:36:35] [1;33mNOTICE[0m[9709][C-0117]: [1;37mExt. 
voicetest[0m:[1;37m2[0m [1;37m@ test[0m: SIP/10.11.12.13:5060-00f8: 
Extension voiceout starting
[1;30m-- [0mExecuting [voicetest@test:3] 
[1;36mLog[0m("[1;35mSIP/10.11.12.13:5060-00f8[0m", 
"[1;35mNotice,SIP/10.11.12.13:5060-00f8: Starting Answer Machine 
Detection[0m") in new stack
[Jan 11 10:36:35] [1;33mNOTICE[0m[9709][C-0117]: [1;37mExt. 
voicetest[0m:[1;37m3[0m [1;37m@ test[0m: SIP/10.11.12.13:5060-00f8: 
Starting Answer Machine Detection
[1;30m-- [0mExecuting [voicetest@test:4] 
[1;36mAMD[0m("[1;35mSIP/10.11.12.13:5060-00f8[0m", "[1;35m[0m") in 
new stack
[1;30m-- [0mAMD: SIP/10.11.12.13:5060-00f8 (N/A) (N/A) (Fmt: alaw)
[1;30m-- [0mAMD: initialSilence [2500] greeting [1500] 
afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] 
betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256] 
maximumWordLength [5000] 

[Kswasp6*CLI> 
[0K[1;30m-- [0mAMD: Channel [SIP/10.11.12.13:5060-00f8]. Changed 
state to STATE_IN_SILENCE

[Kswasp6*CLI> 
[0K[1;30m-- [0mAMD: Channel [SIP/10.11.12.13:5060-00f8]. Word 
detected. iWordsCount:1
[1;30m-- [0mAMD: Channel [SIP/10.11.12.13:5060-00f8]. Detected Talk, 
previous silence duration: 1980

[Kswasp6*CLI> 
[0K[1;30m-- [0mAMD: Channel [SIP/10.11.12.13:5060-00f8]. Changed 
state to STATE_IN_SILENCE

[Kswasp6*CLI> 
[0K[1;30m-- [0mAMD: Channel [SIP/10.11.12.13:5060-00f8]. Detected 
Talk, previous silence duration: 100

[Kswasp6*CLI> 
[0K[1;30m-- [0mAMD: Channel [SIP/10.11.12.13:5060-00f8]. Word 
detected. iWordsCount:2

[Kswasp6*CLI> 
[0K[1;30m-- [0mAMD: Channel [SIP/10.11.12.13:5060-00f8]. Changed 
state to STATE_IN_SILENCE

[Kswasp6*CLI> 
[0K[1;30m-- [0mAMD: Channel [SIP/10.11.12.13:5060-00f8]. Detected 
Talk, previous silence duration: 100

[Kswasp6*CLI> 
[0K[1;30m-- [0mAMD: Channel [SIP/10.11.12.13:5060-00f8]. Word 
detected. iWordsCount:3
[1;30m-- [0mAMD: Channel [SIP/10.11.12.13:5060-00f8]. ANSWERING 
MACHINE: iWordsCount:3

[Kswasp6*CLI> 
[0K[1;30m-- [0mExecuting [voicetest@test:5]

Re: [asterisk-users] Detecting a fax

2019-01-11 Thread Administrator TOOTAI


Le 11/01/2019 à 10:23, Neil Youngman a écrit :

On 11/01/2019 09:19, Administrator TOOTAI wrote:

Le 11/01/2019 à 10:12, Neil Youngman a écrit :
A while back, I posted about detecting when a call was picked up by 
a fax machine. It was suggested that having a "fax" extension and 
"faxdetect=yes" would cause it to jump to the "fax" extension. This 
was not something I could get to work.


I have now created a very simple example. In sip.conf I have 
"faxdetect = yes". My example extension is:


[test]
;
; Voice test extension
;
exten => voicetest,1,NoOp()
 same => n,LOG(Notice,${CHANNEL}: Extension voiceout starting)
 same => n,LOG(Notice,${CHANNEL}: Starting Answer Machine Detection)
 same => n,AMD()
 same => n,LOG(Notice,${CHANNEL}: Answer Machine Detection 
${AMDSTATUS}/${AMDCAUSE})

 same => n,Playback(/var/lib/asterisk/sounds/en/demo-congrats)
 same => n,LOG(Notice,${CHANNEL}: Voice out extension complete)
 same => n(hangup),Hangup()


;
; Fax detected extension
;
exten => fax,1,NoOp()
 same => n,LOG(Notice,${CHANNEL}: Extension fax starting)
 same => n,LOG(Notice,${CHANNEL}: Fax Machine Detected)
 same => n,Playback(/var/lib/asterisk/sounds/en/silence/2)
 same => n,LOG(Notice,${CHANNEL}: Fax extension complete)
 same => n(hangup),Hangup()

and the logs show that calling a fax using the voiceout extension in 
context test does not result in the fax extension being triggered.


[Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Extension voiceout starting
[Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Starting Answer Machine Detection
[Jan 11 08:55:13] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Answer Machine Detection 
MACHINE/LONGGREETING-1500-1500
[Jan 11 08:55:44] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Voice out extension complete


Just for completeness this is how the call is originated, with a 
different phone number:


Action: Originate
ActionId: 1234567W001-125
Context: test
Exten: voicetest
Priority: 1
Channel: SIP/+441632660987@31.13.156.183:5060
Timeout: 6
Async: True

Can anyone offer any insight into why this isn't working?

Neil Youngman




You didn't ANSWER() the call


It's an outgoing call. I wouldn't expect to answer an outgoing call?


I don't understand your goal. You want to send or receive fax?

On our side, dialplan very simplified on incoming calls (receive fax):

. in CLI> fax show settings ; output FAX For Asterisk Settings: blabla
. Set(FAXOPT(faxdetect)=yes); or faxdetect=yes in sip.conf
. Answer()
; if a fax is detected, go to fax extension
. exten => fax,n,ReceiveFAX(${FAXPATHTOFILE}/${FAXFILE},fs)

and done.
--
Daniel

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Re: [asterisk-users] Detecting a fax

2019-01-11 Thread Floimair Florian

I would guess from your explanation that the "outgoing" call somehow ends up in 
your Asterisk machine again, either at the voicetest or fax extension.
You don't answer it in either of the extensions. 
That's what TOOTAI meant.

If this is done in another extension, than this part of the Dialplan is missing 
in your post. So we can only guess what's going on.
Either post the full dialplan or add some log or cli output (with verbose set 
to at least 1) so we can have a chance of seeing what's going on.

With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com 

Security and Communication by Commend

FN 178618z | LG Salzburg

Am 11.01.19, 10:23 schrieb "asterisk-users im Auftrag von Neil Youngman" 
:

On 11/01/2019 09:19, Administrator TOOTAI wrote:
> Le 11/01/2019 à 10:12, Neil Youngman a écrit :
>> A while back, I posted about detecting when a call was picked up by a 
>> fax machine. It was suggested that having a "fax" extension and 
>> "faxdetect=yes" would cause it to jump to the "fax" extension. This 
>> was not something I could get to work.
>>
>> I have now created a very simple example. In sip.conf I have 
>> "faxdetect = yes". My example extension is:
>>
>> [test]
>> ;
>> ; Voice test extension
>> ;
>> exten => voicetest,1,NoOp()
>>  same => n,LOG(Notice,${CHANNEL}: Extension voiceout starting)
>>  same => n,LOG(Notice,${CHANNEL}: Starting Answer Machine Detection)
>>  same => n,AMD()
>>  same => n,LOG(Notice,${CHANNEL}: Answer Machine Detection 
>> ${AMDSTATUS}/${AMDCAUSE})
>>  same => n,Playback(/var/lib/asterisk/sounds/en/demo-congrats)
>>  same => n,LOG(Notice,${CHANNEL}: Voice out extension complete)
>>  same => n(hangup),Hangup()
>>
>>
>> ;
>> ; Fax detected extension
>> ;
>> exten => fax,1,NoOp()
>>  same => n,LOG(Notice,${CHANNEL}: Extension fax starting)
>>  same => n,LOG(Notice,${CHANNEL}: Fax Machine Detected)
>>  same => n,Playback(/var/lib/asterisk/sounds/en/silence/2)
>>  same => n,LOG(Notice,${CHANNEL}: Fax extension complete)
>>  same => n(hangup),Hangup()
>>
>> and the logs show that calling a fax using the voiceout extension in 
>> context test does not result in the fax extension being triggered.
>>
>> [Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
>> SIP/31.13.156.183:5060-00f4: Extension voiceout starting
>> [Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
>> SIP/31.13.156.183:5060-00f4: Starting Answer Machine Detection
>> [Jan 11 08:55:13] NOTICE[18073][C-0115] Ext. voicetest: 
>> SIP/31.13.156.183:5060-00f4: Answer Machine Detection 
>> MACHINE/LONGGREETING-1500-1500
>> [Jan 11 08:55:44] NOTICE[18073][C-0115] Ext. voicetest: 
>> SIP/31.13.156.183:5060-00f4: Voice out extension complete
>>
>> Just for completeness this is how the call is originated, with a 
>> different phone number:
>>
>> Action: Originate
>> ActionId: 1234567W001-125
>> Context: test
>> Exten: voicetest
>> Priority: 1
>> Channel: SIP/+441632660987@31.13.156.183:5060
>> Timeout: 6
>> Async: True
>>
>> Can anyone offer any insight into why this isn't working?
>>
>> Neil Youngman
>>
>>
> 
> You didn't ANSWER() the call

It's an outgoing call. I wouldn't expect to answer an outgoing call?

Neil

Neil Youngman 
Developer
Wirefast Limited

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Re: [asterisk-users] [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

2019-01-11 Thread Jean Aunis


Le 11/01/2019 à 16:47, Matt Riddell a écrit :

Hiya,

When I hang up on a call to my stasis app I’m getting multiple channelDestroyed 
events for the same channel:

app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer was not talking to anyone
app.js:1126:13) 2019-01-11 10:28:29
app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer was not talking to anyone
app.js:1126:13) 2019-01-11 10:28:29
app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer was not talking to anyone

I’m only registering once to the event:

ari.on('ChannelDestroyed', channelDestroyed);

Is this normal?

I’m writing like a CDR on channel destroyed so don’t want to write it multiple 
times.

Should I keep an array of channels and only write if I haven’t seen the event 
for that channel before?

Cheers,

Matt Riddell
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Hello,

It may happen if several applications subscribed to the channel. In this 
case you can discriminate the events on the basis of application name.


Regards

Jean Aunis


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[asterisk-users] chan_sip bind port

2019-01-11 Thread Dan Cropp

Did something recently change for the chan_sip bindport setting?

I know I had this working with the previous version of asterisk.  Can't 
remember if it was an earlier 16.x version or 13.x
I was running chan_sip (binding to port 5061) and PJSIP using the default port 
of 5060.

I recently upgraded to asterisk 16.1.1 and don't recall changing anything 
specific for the sip.conf

My sip.conf ...
[general]
context = ABC
bindaddr = 0.0.0.0
bindport = 5061
websocket_enabled = false
srvlookup = no
sipdebug = yes
callcounter = yes

My pjsip.conf transports...
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[transport2]
type = transport
bind = 0.0.0.0
protocol = wss

Every time I start up, asterisk logging is showing the following, as if it's 
not reading in the bindport setting.

[01/11 09:43:51.059] WARNING[1658] chan_sip.c: Failed to bind to 0.0.0.0:5060: 
Address already in use

Any thoughts?

Dan
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Re: [asterisk-users] chan_sip bind port

2019-01-11 Thread Dan Cropp

Please disregard.

I just found my problem.  In my sip.conf, I had an include statement for 
another file at the top.  The included file also had a general section.  Once I 
deleted that general section and reloaded everything worked as expected, now 
using port 5061.

From: asterisk-users  On Behalf Of Dan 
Cropp
Sent: Friday, January 11, 2019 11:54 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] chan_sip bind port

Did something recently change for the chan_sip bindport setting?

I know I had this working with the previous version of asterisk.  Can't 
remember if it was an earlier 16.x version or 13.x
I was running chan_sip (binding to port 5061) and PJSIP using the default port 
of 5060.

I recently upgraded to asterisk 16.1.1 and don't recall changing anything 
specific for the sip.conf

My sip.conf ...
[general]
context = ABC
bindaddr = 0.0.0.0
bindport = 5061
websocket_enabled = false
srvlookup = no
sipdebug = yes
callcounter = yes

My pjsip.conf transports...
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[transport2]
type = transport
bind = 0.0.0.0
protocol = wss

Every time I start up, asterisk logging is showing the following, as if it's 
not reading in the bindport setting.

[01/11 09:43:51.059] WARNING[1658] chan_sip.c: Failed to bind to 0.0.0.0:5060: 
Address already in use

Any thoughts?

Dan
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Re: [asterisk-users] [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

2019-01-11 Thread Matt Riddell


> On Jan 11, 2019, at 11:14, Jean Aunis  wrote:
> 
> Le 11/01/2019 à 16:47, Matt Riddell a écrit :
>> Hiya,
>> 
>> When I hang up on a call to my stasis app I’m getting multiple 
>> channelDestroyed events for the same channel:
> 
> It may happen if several applications subscribed to the channel. In this case 
> you can discriminate the events on the basis of application name.

Yeah you’re right - I was registering 3 applications so get three events - even 
though the channel was only in one application.

Thanks,

Matt Riddell
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Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joel)

2019-01-11 Thread Stefan Viljoen

>Hi,

>On the other side.. There is a specific note regarding CDR behavior changes 
>from asterisk 12 onwards. So going from 1.8 to 13 means it applies to you. 

>Have a look at:

>https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12 

>And 

>https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification

Hi Joel

I've read the above in detail, and studied them, but I cannot find anything 
that explains the CDR behaviour I'm seeing in Asterisk 13.

The only thing relevant is the following in the Upgrade Notes:

---
The duration, billsec, start, answer, and end times now reflect the times 
associated with the current CDR for the channel, as opposed to a cumulative 
measurement of all CDRs for that channel.
---

That is exactly where the problem lies, e. g. the above behaviour is exactly 
what is NOT happening...

E. g. instead of the origination channel created by the AMI call we make as 
previously described, having its CDR written and done with, its CDR data for 
the originating call leg apparently gets propagated into the resulting outgoing 
call leg of the call to the AMI originate command.

E. g. my origination channel's CDR is

Start: 2019-01-11 08:22:07
Answer: 2019-01-11 08:22:12
End: 2019-01-11 08:24:09

and the associated outgoin channel's CDR is

Start: 2019-01-11 08:22:12
Answer: 2019-01-11 08:22:12
End: 2019-01-11 08:24:09

e.g. it appears that the origination channel's pickup time of Answer: 
2019-01-11 08:22:12 gets propagated into the outgoing channel's answer time of 
Answer: 2019-01-11 08:22:12.

E. g. the above in the release notes is 180 degrees the opposite of how 
Asterisk 13 actually behaves in practice?

Despite the above release note, it DOES appear in fact that CDR measurement is 
cumulative... very definitely not distinct

Thanks for the reply!

Regards

Stefan



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[asterisk-users] [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

2019-01-11 Thread Matt Riddell

Hiya,

When I hang up on a call to my stasis app I’m getting multiple channelDestroyed 
events for the same channel:

app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer was not talking to anyone
app.js:1126:13) 2019-01-11 10:28:29
app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer was not talking to anyone
app.js:1126:13) 2019-01-11 10:28:29
app.js:985:13) Channel was destroyed: 1547220509.77
app.js:1029:17) This was a customer
app.js:1030:17) Checking if this was a customer talking to an agent
app.js:1043:21) Customer was not talking to anyone

I’m only registering once to the event:

ari.on('ChannelDestroyed', channelDestroyed);

Is this normal?

I’m writing like a CDR on channel destroyed so don’t want to write it multiple 
times.

Should I keep an array of channels and only write if I haven’t seen the event 
for that channel before?

Cheers,

Matt Riddell
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Re: [asterisk-users] Detecting a fax

[asterisk-users] Can SIP domain help to set multiple SIP trunks between two boxes ?

Re: [asterisk-users] Detecting a fax

[asterisk-users] Detecting a fax

Re: [asterisk-users] Detecting a fax

Re: [asterisk-users] Detecting a fax

Re: [asterisk-users] Detecting a fax

Re: [asterisk-users] Detecting a fax

Re: [asterisk-users] [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

[asterisk-users] chan_sip bind port

Re: [asterisk-users] chan_sip bind port

Re: [asterisk-users] [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joel)

[asterisk-users] [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel

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