Adhearsion Recording Problem

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Adhearsion Recording Problem

Manjil shrestha
I have 3 SIP clients (3001, 3002, 3003).

Suppose, 3001 is the user, and 3002 & 3003 are the operators.

When 3001 dials 9801, the conversation between the user and operator has to be recorded ( which it records properly ).

The problem in my prototype is,,,
  1.     When it records including the ringback tone, it acts normally, i.e., the call disconnects properly. ( But, I don't need to record the ringback tone).
  2.     After I record only after the call has been received by the operator, the recording is done successfully, but the call won't be disconnected from the operator side. (The call only disconnects from the user side).

So, what I need is to make my prototype to disconnect the call from operator side as well and not include the ringback tone while recording the call conversation between user and operator. The call needs to be recorded only after the operator picks up the call.

Kindly, please help me in this!!

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Re: Adhearsion Recording Problem

Ben Langfeld-2
You should provide code and logs to support your report such that we might make informed and precise suggestions.

On Mon, 17 Oct 2016 at 09:40 Manjil shrestha <[hidden email]> wrote:
I have 3 SIP clients (3001, 3002, 3003).

Suppose, 3001 is the user, and 3002 & 3003 are the operators.

When 3001 dials 9801, the conversation between the user and operator has to be recorded ( which it records properly ).

The problem in my prototype is,,,
  1.     When it records including the ringback tone, it acts normally, i.e., the call disconnects properly. ( But, I don't need to record the ringback tone).
  2.     After I record only after the call has been received by the operator, the recording is done successfully, but the call won't be disconnected from the operator side. (The call only disconnects from the user side).

So, what I need is to make my prototype to disconnect the call from operator side as well and not include the ringback tone while recording the call conversation between user and operator. The call needs to be recorded only after the operator picks up the call.

Kindly, please help me in this!!

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Re: Adhearsion Recording Problem

Ben Langfeld-2
Note also that the logs should be at TRACE level and your code should have any non-essential pieces removed so that it is clear.

On Mon, 17 Oct 2016 at 10:18 Ben Langfeld <[hidden email]> wrote:
You should provide code and logs to support your report such that we might make informed and precise suggestions.

On Mon, 17 Oct 2016 at 09:40 Manjil shrestha <[hidden email]> wrote:
I have 3 SIP clients (3001, 3002, 3003).

Suppose, 3001 is the user, and 3002 & 3003 are the operators.

When 3001 dials 9801, the conversation between the user and operator has to be recorded ( which it records properly ).

The problem in my prototype is,,,
  1.     When it records including the ringback tone, it acts normally, i.e., the call disconnects properly. ( But, I don't need to record the ringback tone).
  2.     After I record only after the call has been received by the operator, the recording is done successfully, but the call won't be disconnected from the operator side. (The call only disconnects from the user side).

So, what I need is to make my prototype to disconnect the call from operator side as well and not include the ringback tone while recording the call conversation between user and operator. The call needs to be recorded only after the operator picks up the call.

Kindly, please help me in this!!

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Re: Adhearsion Recording Problem

Manjil shrestha
These are the trace level logs I receive while disconnecting from operator side:

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16

Event: Hangup
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16
Cause-txt: Normal Clearing

These are the trace level logs I receive when I disconnect from user side, after already disconnecting from operator:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Variable: AGISTATUS
Value: HANGUP

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Cause: 16


On Monday, 17 October 2016 18:04:07 UTC+5:45, Ben Langfeld wrote:
Note also that the logs should be at TRACE level and your code should have any non-essential pieces removed so that it is clear.

On Mon, 17 Oct 2016 at 10:18 Ben Langfeld <<a href="javascript:" target="_blank" gdf-obfuscated-mailto="1gQ6nS-DBAAJ" rel="nofollow" onmousedown="this.href=&#39;javascript:&#39;;return true;" onclick="this.href=&#39;javascript:&#39;;return true;">b...@...> wrote:
You should provide code and logs to support your report such that we might make informed and precise suggestions.

On Mon, 17 Oct 2016 at 09:40 Manjil shrestha <<a href="javascript:" target="_blank" gdf-obfuscated-mailto="1gQ6nS-DBAAJ" rel="nofollow" onmousedown="this.href=&#39;javascript:&#39;;return true;" onclick="this.href=&#39;javascript:&#39;;return true;">shrestha...@...> wrote:
I have 3 SIP clients (3001, 3002, 3003).

Suppose, 3001 is the user, and 3002 & 3003 are the operators.

When 3001 dials 9801, the conversation between the user and operator has to be recorded ( which it records properly ).

The problem in my prototype is,,,
  1.     When it records including the ringback tone, it acts normally, i.e., the call disconnects properly. ( But, I don't need to record the ringback tone).
  2.     After I record only after the call has been received by the operator, the recording is done successfully, but the call won't be disconnected from the operator side. (The call only disconnects from the user side).

So, what I need is to make my prototype to disconnect the call from operator side as well and not include the ringback tone while recording the call conversation between user and operator. The call needs to be recorded only after the operator picks up the call.

Kindly, please help me in this!!

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Re: Adhearsion Recording Problem

Ben Langfeld
You have trimmed these logs. Please don't do that. Please provide full Adhearsion TRACE logs without editing. 

Enviado do meu iPhone

Em 18 de out de 2016, às 03:06, Manjil shrestha <[hidden email]> escreveu:

These are the trace level logs I receive while disconnecting from operator side:

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16

Event: Hangup
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16
Cause-txt: Normal Clearing

These are the trace level logs I receive when I disconnect from user side, after already disconnecting from operator:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Variable: AGISTATUS
Value: HANGUP

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Cause: 16


On Monday, 17 October 2016 18:04:07 UTC+5:45, Ben Langfeld wrote:
Note also that the logs should be at TRACE level and your code should have any non-essential pieces removed so that it is clear.

On Mon, 17 Oct 2016 at 10:18 Ben Langfeld <<a href="javascript:" target="_blank" gdf-obfuscated-mailto="1gQ6nS-DBAAJ" rel="nofollow" onmousedown="this.href='javascript:';return true;" onclick="this.href='javascript:';return true;">b...@...> wrote:
You should provide code and logs to support your report such that we might make informed and precise suggestions.

On Mon, 17 Oct 2016 at 09:40 Manjil shrestha <<a href="javascript:" target="_blank" gdf-obfuscated-mailto="1gQ6nS-DBAAJ" rel="nofollow" onmousedown="this.href='javascript:';return true;" onclick="this.href='javascript:';return true;">shrestha...@...> wrote:
I have 3 SIP clients (3001, 3002, 3003).

Suppose, 3001 is the user, and 3002 & 3003 are the operators.

When 3001 dials 9801, the conversation between the user and operator has to be recorded ( which it records properly ).

The problem in my prototype is,,,
  1.     When it records including the ringback tone, it acts normally, i.e., the call disconnects properly. ( But, I don't need to record the ringback tone).
  2.     After I record only after the call has been received by the operator, the recording is done successfully, but the call won't be disconnected from the operator side. (The call only disconnects from the user side).

So, what I need is to make my prototype to disconnect the call from operator side as well and not include the ringback tone while recording the call conversation between user and operator. The call needs to be recorded only after the operator picks up the call.

Kindly, please help me in this!!

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Re: Adhearsion Recording Problem

Manjil shrestha
Here is the full trace level log. Please have a look..
http://pastebin.com/y4uXY6wf

On Tuesday, 18 October 2016 17:40:24 UTC+5:45, Ben Langfeld wrote:
You have trimmed these logs. Please don't do that. Please provide full Adhearsion TRACE logs without editing. 

Enviado do meu iPhone

Em 18 de out de 2016, às 03:06, Manjil shrestha <<a href="javascript:" target="_blank" gdf-obfuscated-mailto="UTGGv3jQBAAJ" rel="nofollow" onmousedown="this.href=&#39;javascript:&#39;;return true;" onclick="this.href=&#39;javascript:&#39;;return true;">shrestha...@...> escreveu:

These are the trace level logs I receive while disconnecting from operator side:

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16

Event: Hangup
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16
Cause-txt: Normal Clearing

These are the trace level logs I receive when I disconnect from user side, after already disconnecting from operator:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Variable: AGISTATUS
Value: HANGUP

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Cause: 16


On Monday, 17 October 2016 18:04:07 UTC+5:45, Ben Langfeld wrote:
Note also that the logs should be at TRACE level and your code should have any non-essential pieces removed so that it is clear.

On Mon, 17 Oct 2016 at 10:18 Ben Langfeld <[hidden email]> wrote:
You should provide code and logs to support your report such that we might make informed and precise suggestions.

On Mon, 17 Oct 2016 at 09:40 Manjil shrestha <[hidden email]> wrote:
I have 3 SIP clients (3001, 3002, 3003).

Suppose, 3001 is the user, and 3002 & 3003 are the operators.

When 3001 dials 9801, the conversation between the user and operator has to be recorded ( which it records properly ).

The problem in my prototype is,,,
  1.     When it records including the ringback tone, it acts normally, i.e., the call disconnects properly. ( But, I don't need to record the ringback tone).
  2.     After I record only after the call has been received by the operator, the recording is done successfully, but the call won't be disconnected from the operator side. (The call only disconnects from the user side).

So, what I need is to make my prototype to disconnect the call from operator side as well and not include the ringback tone while recording the call conversation between user and operator. The call needs to be recorded only after the operator picks up the call.

Kindly, please help me in this!!

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Re: Adhearsion Recording Problem

Ben Langfeld-2
You still havn't provided your code, but from the log I suspect your usage of record is incorrect and you're actually doing something like this:

dial "foo"
record

Here you are not recording after connection, you are recording after the joined calls are hung up. You would have to do an async recording, perhaps in a pre_join callback on your dial.

On Wed, 19 Oct 2016 at 02:45 Manjil shrestha <[hidden email]> wrote:
Here is the full trace level log. Please have a look..


On Tuesday, 18 October 2016 17:40:24 UTC+5:45, Ben Langfeld wrote:
You have trimmed these logs. Please don't do that. Please provide full Adhearsion TRACE logs without editing. 

Enviado do meu iPhone

Em 18 de out de 2016, às 03:06, Manjil shrestha <[hidden email]> escreveu:

These are the trace level logs I receive while disconnecting from operator side:

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: <a href="tel:(14)%207676-6302" value="+551476766302" class="gmail_msg" target="_blank">1476766302.10
Linkedid: <a href="tel:(14)%207676-6285" value="+551476766285" class="gmail_msg" target="_blank">1476766285.9
Cause: 16

Event: Hangup
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: <a href="tel:(14)%207676-6302" value="+551476766302" class="gmail_msg" target="_blank">1476766302.10
Linkedid: <a href="tel:(14)%207676-6285" value="+551476766285" class="gmail_msg" target="_blank">1476766285.9
Cause: 16
Cause-txt: Normal Clearing

These are the trace level logs I receive when I disconnect from user side, after already disconnecting from operator:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: <a href="tel:(14)%207676-6285" value="+551476766285" class="gmail_msg" target="_blank">1476766285.9
Linkedid: <a href="tel:(14)%207676-6285" value="+551476766285" class="gmail_msg" target="_blank">1476766285.9
Variable: AGISTATUS
Value: HANGUP

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: <a href="tel:(14)%207676-6285" value="+551476766285" class="gmail_msg" target="_blank">1476766285.9
Linkedid: <a href="tel:(14)%207676-6285" value="+551476766285" class="gmail_msg" target="_blank">1476766285.9
Cause: 16


On Monday, 17 October 2016 18:04:07 UTC+5:45, Ben Langfeld wrote:
Note also that the logs should be at TRACE level and your code should have any non-essential pieces removed so that it is clear.

On Mon, 17 Oct 2016 at 10:18 Ben Langfeld <[hidden email]> wrote:
You should provide code and logs to support your report such that we might make informed and precise suggestions.

On Mon, 17 Oct 2016 at 09:40 Manjil shrestha <[hidden email]> wrote:
I have 3 SIP clients (3001, 3002, 3003).

Suppose, 3001 is the user, and 3002 & 3003 are the operators.

When 3001 dials 9801, the conversation between the user and operator has to be recorded ( which it records properly ).

The problem in my prototype is,,,
  1.     When it records including the ringback tone, it acts normally, i.e., the call disconnects properly. ( But, I don't need to record the ringback tone).
  2.     After I record only after the call has been received by the operator, the recording is done successfully, but the call won't be disconnected from the operator side. (The call only disconnects from the user side).

So, what I need is to make my prototype to disconnect the call from operator side as well and not include the ringback tone while recording the call conversation between user and operator. The call needs to be recorded only after the operator picks up the call.

Kindly, please help me in this!!

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Re: Adhearsion Recording Problem

Manjil shrestha
def dial_op(num)
@dial_num = num
@status = dial "SIP/#{@dial_num}", :async => true, :ringback => 'ring'

case @status.result
when :answer
puts 'Received!!!!'
record :async => true do |event|
puts 'Recording!!!!'
@rec = event.recording.uri
@fname = File.basename(@rec)
puts @fname, '--------------------------------------'
end
end

end

This is the part of code tha I used to dial and record only when the call is answered. The call goes well but the record does not operate.

On Thursday, 20 October 2016 04:12:59 UTC+5:45, Ben Langfeld wrote:
You still havn't provided your code, but from the log I suspect your usage of record is incorrect and you're actually doing something like this:

dial "foo"
record

Here you are not recording after connection, you are recording after the joined calls are hung up. You would have to do an async recording, perhaps in a pre_join callback on your dial.

On Wed, 19 Oct 2016 at 02:45 Manjil shrestha <<a href="javascript:" target="_blank" gdf-obfuscated-mailto="ESXmhJJBBQAJ" rel="nofollow" onmousedown="this.href=&#39;javascript:&#39;;return true;" onclick="this.href=&#39;javascript:&#39;;return true;">shrestha...@...> wrote:
Here is the full trace level log. Please have a look..
<a href="http://pastebin.com/y4uXY6wf" target="_blank" rel="nofollow" onmousedown="this.href=&#39;http://www.google.com/url?q\x3dhttp%3A%2F%2Fpastebin.com%2Fy4uXY6wf\x26sa\x3dD\x26sntz\x3d1\x26usg\x3dAFQjCNETK6gXdf4gUUSSoFTp57MX6WHMeg&#39;;return true;" onclick="this.href=&#39;http://www.google.com/url?q\x3dhttp%3A%2F%2Fpastebin.com%2Fy4uXY6wf\x26sa\x3dD\x26sntz\x3d1\x26usg\x3dAFQjCNETK6gXdf4gUUSSoFTp57MX6WHMeg&#39;;return true;">http://pastebin.com/y4uXY6wf


On Tuesday, 18 October 2016 17:40:24 UTC+5:45, Ben Langfeld wrote:
You have trimmed these logs. Please don't do that. Please provide full Adhearsion TRACE logs without editing. 

Enviado do meu iPhone

Em 18 de out de 2016, às 03:06, Manjil shrestha <[hidden email]> escreveu:

These are the trace level logs I receive while disconnecting from operator side:

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16

Event: Hangup
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16
Cause-txt: Normal Clearing

These are the trace level logs I receive when I disconnect from user side, after already disconnecting from operator:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Variable: AGISTATUS
Value: HANGUP

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Cause: 16


On Monday, 17 October 2016 18:04:07 UTC+5:45, Ben Langfeld wrote:
Note also that the logs should be at TRACE level and your code should have any non-essential pieces removed so that it is clear.

On Mon, 17 Oct 2016 at 10:18 Ben Langfeld <[hidden email]> wrote:
You should provide code and logs to support your report such that we might make informed and precise suggestions.

On Mon, 17 Oct 2016 at 09:40 Manjil shrestha <[hidden email]> wrote:
I have 3 SIP clients (3001, 3002, 3003).

Suppose, 3001 is the user, and 3002 & 3003 are the operators.

When 3001 dials 9801, the conversation between the user and operator has to be recorded ( which it records properly ).

The problem in my prototype is,,,
  1.     When it records including the ringback tone, it acts normally, i.e., the call disconnects properly. ( But, I don't need to record the ringback tone).
  2.     After I record only after the call has been received by the operator, the recording is done successfully, but the call won't be disconnected from the operator side. (The call only disconnects from the user side).

So, what I need is to make my prototype to disconnect the call from operator side as well and not include the ringback tone while recording the call conversation between user and operator. The call needs to be recorded only after the operator picks up the call.

Kindly, please help me in this!!

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Re: Adhearsion Recording Problem

Ben Langfeld-2
Async #dial is not a thing. My previous explanation holds. http://www.rubydoc.info/github/adhearsion/adhearsion/Adhearsion/CallController/Dial

On Thu, 20 Oct 2016 at 01:17 Manjil shrestha <[hidden email]> wrote:
def dial_op(num)
@dial_num = num
@status = dial "SIP/#{@dial_num}", :async => true, :ringback => 'ring'

case @status.result
when :answer
puts 'Received!!!!'
record :async => true do |event|
puts 'Recording!!!!'
@rec = event.recording.uri
@fname = File.basename(@rec)
puts @fname, '--------------------------------------'
end
end

end

This is the part of code tha I used to dial and record only when the call is answered. The call goes well but the record does not operate.

On Thursday, 20 October 2016 04:12:59 UTC+5:45, Ben Langfeld wrote:
You still havn't provided your code, but from the log I suspect your usage of record is incorrect and you're actually doing something like this:

dial "foo"
record

Here you are not recording after connection, you are recording after the joined calls are hung up. You would have to do an async recording, perhaps in a pre_join callback on your dial.

On Wed, 19 Oct 2016 at 02:45 Manjil shrestha <[hidden email]> wrote:
Here is the full trace level log. Please have a look..


On Tuesday, 18 October 2016 17:40:24 UTC+5:45, Ben Langfeld wrote:
You have trimmed these logs. Please don't do that. Please provide full Adhearsion TRACE logs without editing. 

Enviado do meu iPhone

Em 18 de out de 2016, às 03:06, Manjil shrestha <[hidden email]> escreveu:

These are the trace level logs I receive while disconnecting from operator side:

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16

Event: Hangup
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16
Cause-txt: Normal Clearing

These are the trace level logs I receive when I disconnect from user side, after already disconnecting from operator:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Variable: AGISTATUS
Value: HANGUP

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Cause: 16


On Monday, 17 October 2016 18:04:07 UTC+5:45, Ben Langfeld wrote:
Note also that the logs should be at TRACE level and your code should have any non-essential pieces removed so that it is clear.

On Mon, 17 Oct 2016 at 10:18 Ben Langfeld <[hidden email]> wrote:
You should provide code and logs to support your report such that we might make informed and precise suggestions.

On Mon, 17 Oct 2016 at 09:40 Manjil shrestha <[hidden email]> wrote:
I have 3 SIP clients (3001, 3002, 3003).

Suppose, 3001 is the user, and 3002 & 3003 are the operators.

When 3001 dials 9801, the conversation between the user and operator has to be recorded ( which it records properly ).

The problem in my prototype is,,,
  1.     When it records including the ringback tone, it acts normally, i.e., the call disconnects properly. ( But, I don't need to record the ringback tone).
  2.     After I record only after the call has been received by the operator, the recording is done successfully, but the call won't be disconnected from the operator side. (The call only disconnects from the user side).

So, what I need is to make my prototype to disconnect the call from operator side as well and not include the ringback tone while recording the call conversation between user and operator. The call needs to be recorded only after the operator picks up the call.

Kindly, please help me in this!!

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Re: Adhearsion Recording Problem

Ben Klang-2
In reply to this post by Manjil shrestha
What Ben Langfeld is trying to explain to you is that you can’t do the record after the dial, because the dial doesn’t return until the call disconnects.  You’re effectively trying to record *after* the call has hung up.

Try something like this:

record :async => true do |event|
  # This code will execute when the recording is complete (when the call ends)
  logger.info “Your recording is at #{event.recording,uri}”
end

dial “SIP/#{@dial_num}”


Let us know if that helps.
/BAK/


-- 
Ben Klang
Principal/Technology Strategist, Mojo Lingo
+1.404.475.4841

Mojo Lingo -- Voice applications that work like magic
Twitter: @MojoLingo

Il giorno 19 ott 2016, alle ore 23:17, Manjil shrestha <[hidden email]> ha scritto:

def dial_op(num)
@dial_num = num
@status = dial "SIP/#{@dial_num}", :async => true, :ringback => 'ring'

case @status.result
when :answer
puts 'Received!!!!'
record :async => true do |event|
puts 'Recording!!!!'
@rec = event.recording.uri
@fname = File.basename(@rec)
puts @fname, '--------------------------------------'
end
end

end

This is the part of code tha I used to dial and record only when the call is answered. The call goes well but the record does not operate.

On Thursday, 20 October 2016 04:12:59 UTC+5:45, Ben Langfeld wrote:
You still havn't provided your code, but from the log I suspect your usage of record is incorrect and you're actually doing something like this:

dial "foo"
record

Here you are not recording after connection, you are recording after the joined calls are hung up. You would have to do an async recording, perhaps in a pre_join callback on your dial.

On Wed, 19 Oct 2016 at 02:45 Manjil shrestha <<a href="javascript:" target="_blank" gdf-obfuscated-mailto="ESXmhJJBBQAJ" rel="nofollow" onmousedown="this.href='javascript:';return true;" onclick="this.href='javascript:';return true;" class="">shrestha...@...> wrote:
Here is the full trace level log. Please have a look..
<a href="http://pastebin.com/y4uXY6wf" target="_blank" rel="nofollow" onmousedown="this.href='http://www.google.com/url?q\x3dhttp%3A%2F%2Fpastebin.com%2Fy4uXY6wf\x26sa\x3dD\x26sntz\x3d1\x26usg\x3dAFQjCNETK6gXdf4gUUSSoFTp57MX6WHMeg';return true;" onclick="this.href='http://www.google.com/url?q\x3dhttp%3A%2F%2Fpastebin.com%2Fy4uXY6wf\x26sa\x3dD\x26sntz\x3d1\x26usg\x3dAFQjCNETK6gXdf4gUUSSoFTp57MX6WHMeg';return true;" class="">http://pastebin.com/y4uXY6wf


On Tuesday, 18 October 2016 17:40:24 UTC+5:45, Ben Langfeld wrote:
You have trimmed these logs. Please don't do that. Please provide full Adhearsion TRACE logs without editing. 

Enviado do meu iPhone

Em 18 de out de 2016, às 03:06, Manjil shrestha <[hidden email]> escreveu:

These are the trace level logs I receive while disconnecting from operator side:

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16

Event: Hangup
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16
Cause-txt: Normal Clearing

These are the trace level logs I receive when I disconnect from user side, after already disconnecting from operator:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Variable: AGISTATUS
Value: HANGUP

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Cause: 16


On Monday, 17 October 2016 18:04:07 UTC+5:45, Ben Langfeld wrote:
Note also that the logs should be at TRACE level and your code should have any non-essential pieces removed so that it is clear.

On Mon, 17 Oct 2016 at 10:18 Ben Langfeld <[hidden email]> wrote:
You should provide code and logs to support your report such that we might make informed and precise suggestions.

On Mon, 17 Oct 2016 at 09:40 Manjil shrestha <[hidden email]> wrote:
I have 3 SIP clients (3001, 3002, 3003).

Suppose, 3001 is the user, and 3002 & 3003 are the operators.

When 3001 dials 9801, the conversation between the user and operator has to be recorded ( which it records properly ).

The problem in my prototype is,,,
  1.     When it records including the ringback tone, it acts normally, i.e., the call disconnects properly. ( But, I don't need to record the ringback tone).
  2.     After I record only after the call has been received by the operator, the recording is done successfully, but the call won't be disconnected from the operator side. (The call only disconnects from the user side).

So, what I need is to make my prototype to disconnect the call from operator side as well and not include the ringback tone while recording the call conversation between user and operator. The call needs to be recorded only after the operator picks up the call.

Kindly, please help me in this!!

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Re: Adhearsion Recording Problem

Manjil shrestha
Yes, Thank You,, it does the recording but still it records the ringback tone too which I don't want to !! Any help on this!!

On Thursday, 20 October 2016 22:58:07 UTC+5:45, Ben Klang wrote:
What Ben Langfeld is trying to explain to you is that you can’t do the record after the dial, because the dial doesn’t return until the call disconnects.  You’re effectively trying to record *after* the call has hung up.

Try something like this:

record :async => true do |event|
  # This code will execute when the recording is complete (when the call ends)
  <a href="http://logger.info" target="_blank" rel="nofollow" onmousedown="this.href=&#39;http://www.google.com/url?q\x3dhttp%3A%2F%2Flogger.info\x26sa\x3dD\x26sntz\x3d1\x26usg\x3dAFQjCNGOfWV_MqSEYOUyz77vBSYyM6smOg&#39;;return true;" onclick="this.href=&#39;http://www.google.com/url?q\x3dhttp%3A%2F%2Flogger.info\x26sa\x3dD\x26sntz\x3d1\x26usg\x3dAFQjCNGOfWV_MqSEYOUyz77vBSYyM6smOg&#39;;return true;">logger.info “Your recording is at #{event.recording,uri}”
end

dial “SIP/#{@dial_num}”


Let us know if that helps.
/BAK/


-- 
Ben Klang
Principal/Technology Strategist, Mojo Lingo
<a href="javascript:" target="_blank" gdf-obfuscated-mailto="2vRoS_h-BQAJ" rel="nofollow" onmousedown="this.href=&#39;javascript:&#39;;return true;" onclick="this.href=&#39;javascript:&#39;;return true;">bkl...@...
+1.404.475.4841

Mojo Lingo -- Voice applications that work like magic
<a href="http://mojolingo.com/" target="_blank" rel="nofollow" onmousedown="this.href=&#39;http://www.google.com/url?q\x3dhttp%3A%2F%2Fmojolingo.com%2F\x26sa\x3dD\x26sntz\x3d1\x26usg\x3dAFQjCNEdz1VI9Wu7PPmqtzifmF6XsODIog&#39;;return true;" onclick="this.href=&#39;http://www.google.com/url?q\x3dhttp%3A%2F%2Fmojolingo.com%2F\x26sa\x3dD\x26sntz\x3d1\x26usg\x3dAFQjCNEdz1VI9Wu7PPmqtzifmF6XsODIog&#39;;return true;">http://mojolingo.com
Twitter: @MojoLingo

Il giorno 19 ott 2016, alle ore 23:17, Manjil shrestha <<a href="javascript:" target="_blank" gdf-obfuscated-mailto="2vRoS_h-BQAJ" rel="nofollow" onmousedown="this.href=&#39;javascript:&#39;;return true;" onclick="this.href=&#39;javascript:&#39;;return true;">shrestha...@...> ha scritto:

def dial_op(num)
@dial_num = num
@status = dial "SIP/#{@dial_num}", :async => true, :ringback => 'ring'

case @status.result
when :answer
puts 'Received!!!!'
record :async => true do |event|
puts 'Recording!!!!'
@rec = event.recording.uri
@fname = File.basename(@rec)
puts @fname, '--------------------------------------'
end
end

end

This is the part of code tha I used to dial and record only when the call is answered. The call goes well but the record does not operate.

On Thursday, 20 October 2016 04:12:59 UTC+5:45, Ben Langfeld wrote:
You still havn't provided your code, but from the log I suspect your usage of record is incorrect and you're actually doing something like this:

dial "foo"
record

Here you are not recording after connection, you are recording after the joined calls are hung up. You would have to do an async recording, perhaps in a pre_join callback on your dial.

On Wed, 19 Oct 2016 at 02:45 Manjil shrestha <[hidden email]> wrote:
Here is the full trace level log. Please have a look..
<a href="http://pastebin.com/y4uXY6wf" rel="nofollow" target="_blank" onmousedown="this.href=&#39;http://www.google.com/url?q\x3dhttp%3A%2F%2Fpastebin.com%2Fy4uXY6wf\x26sa\x3dD\x26sntz\x3d1\x26usg\x3dAFQjCNETK6gXdf4gUUSSoFTp57MX6WHMeg&#39;;return true;" onclick="this.href=&#39;http://www.google.com/url?q\x3dhttp%3A%2F%2Fpastebin.com%2Fy4uXY6wf\x26sa\x3dD\x26sntz\x3d1\x26usg\x3dAFQjCNETK6gXdf4gUUSSoFTp57MX6WHMeg&#39;;return true;">http://pastebin.com/y4uXY6wf


On Tuesday, 18 October 2016 17:40:24 UTC+5:45, Ben Langfeld wrote:
You have trimmed these logs. Please don't do that. Please provide full Adhearsion TRACE logs without editing. 

Enviado do meu iPhone

Em 18 de out de 2016, às 03:06, Manjil shrestha <[hidden email]> escreveu:

These are the trace level logs I receive while disconnecting from operator side:

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16

Event: Hangup
Privilege: call,all
Channel: SIP/3001-00000005
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: SIP/3002
CallerIDName: unknown
ConnectedLineNum: 3002
ConnectedLineName: <unknown>
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766302.10
Linkedid: 1476766285.9
Cause: 16
Cause-txt: Normal Clearing

These are the trace level logs I receive when I disconnect from user side, after already disconnecting from operator:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Variable: AGISTATUS
Value: HANGUP

Event: SoftHangupRequest
Privilege: call,all
Channel: SIP/3002-00000004
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 3002
CallerIDName: <unknown>
ConnectedLineNum: SIP/3002
ConnectedLineName: unknown
Language: en
AccountCode: 
Context: adhearsion-redirect
Exten: 1
Priority: 1
Uniqueid: 1476766285.9
Linkedid: 1476766285.9
Cause: 16


On Monday, 17 October 2016 18:04:07 UTC+5:45, Ben Langfeld wrote:
Note also that the logs should be at TRACE level and your code should have any non-essential pieces removed so that it is clear.

On Mon, 17 Oct 2016 at 10:18 Ben Langfeld <[hidden email]> wrote:
You should provide code and logs to support your report such that we might make informed and precise suggestions.

On Mon, 17 Oct 2016 at 09:40 Manjil shrestha <[hidden email]> wrote:
I have 3 SIP clients (3001, 3002, 3003).

Suppose, 3001 is the user, and 3002 & 3003 are the operators.

When 3001 dials 9801, the conversation between the user and operator has to be recorded ( which it records properly ).

The problem in my prototype is,,,
  1.     When it records including the ringback tone, it acts normally, i.e., the call disconnects properly. ( But, I don't need to record the ringback tone).
  2.     After I record only after the call has been received by the operator, the recording is done successfully, but the call won't be disconnected from the operator side. (The call only disconnects from the user side).

So, what I need is to make my prototype to disconnect the call from operator side as well and not include the ringback tone while recording the call conversation between user and operator. The call needs to be recorded only after the operator picks up the call.

Kindly, please help me in this!!

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