Re: [asterisk-users] Asterisk CPU use

[Tiago Geada]

I recently had high load average due to disk usage (IO) . I use
mixmonitor() to record to tmpfs and moved mysql to a different disk
(realtime, cdr etc).
Load average is now better.


On 31 July 2013 19:45, Paul Belanger  wrote:

> On 13-07-29 10:22 AM, Eduardo Leones wrote:
>
>> Hello, working in a call center where we set up a structure in asterisk.
>> When my voip reaches 150 calls are with bad quality. We do not transcode
>> codec. What I realized using the top command server (CentOS) processing is
>> too high for the asterisk. But the general processor server is down. Would
>> any limitation of Asterisk to use more hardware resources?
>>
>>
> Your load average is insane.  Time to off load resources from your PBX,
> for example why are you running httpd?  You need to figure out where your
> bottleneck is and then adjust it.
>
> Using something like iotop, netstat and see what your system is doing.
>
> I doubt this is a CPU issue.
>
> --
> Paul Belanger | PolyBeacon, Inc.
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> Github: https://github.com/pabelanger | Twitter:
> https://twitter.com/pabelanger
>
>
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Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Tiago Geada]

Hi,

You just said you use Local channels. Local channel is a dialplan that has
a Dial() to a sip device?

We use queues, and have a queue-macro that sends the UserEvent upon
bridging the call...


On 4 August 2013 16:41, Timothy Smith  wrote:

> Dear Tiago,
>
> Thanks for your answer, but I have a few questions.
>
> Do you use queues? We are operating a call centre with several queues,
> so I don't see how we would use the Dial command. When a call comes
> in, we enter the caller (depending on what options he has selected)
> into a queue. Do you have any alternative method, which would involve
> dialling the agent directly as you described below?
>
> regards,
> T
>
> On Sun, Aug 4, 2013 at 3:47 PM, Tiago Geada  wrote:
> > Hi,
> >
> > Our queue members are Local channels, thus when dialing the agent, the
> > dialplan will do several stuff including:
> >
> > Set(CALLERID(name)=${CALLERID(name)}:Sales)
> > UserEvent(something,data: ${bunch-of-data-in-some-format})
> > Dial(SIP/final-agent-phone,timeout,A(Sales))
> >
> > The UserEvent will be picked up by our client-register-ticket-stuff
> software
> >
> > The announcement A() will be heard by the agent upon answering the call
> like
> > "sales call"
> >
> >
> > On 4 August 2013 02:59, Mitch Claborn  wrote:
> >>
> >> We do something very similar.
> >>
> >> Use the gosub parameter of the Queue application to call a subroutine in
> >> the dial plan when the agent answers the call.
> >>
> >> same =>n,Queue(sales,tc,,sub-QueueConnected)
> >>
> >> [sub-QueueConnected]
> >> ; this runs on the agent/member's channel
> >> exten =>s,1,NoOp()
> >>   ; whatever you need to do here
> >>   same =>n,Return()
> >>
> >> See
> >>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue
> >>
> >>
> >> Mitch
> >>
> >>
> >> On 08/03/2013 12:45 PM, Timothy Smith wrote:
> >>>
> >>> Hello Folks,
> >>>
> >>> I am setting up a call center but we have few agents so one agent is
> >>> able to handle calls of different languages and different queues. For
> >>> the agent to identify the caller, I want a popup to appear as the
> >>> phone starts to ring with the caller's number, language (selected in
> >>> the IVR), Queue (sales, support etc) and any other information (e.g a
> >>> URL with parameters)
> >>>
> >>> I can send this information either via netcat (to a client such as
> >>> yac) to a Windows PC but the problem is I do not know when the caller
> >>> is about to be connected to the agent, so that I run the command. If I
> >>> wasn't using queues, it would be easy because  I would run the netcat
> >>> command and then dial the user's extension.
> >>>
> >>> My Question is: Is there a way I can know when the caller is just
> >>> about to be connected to an agent (when the agent's SIP extension
> >>> starts ringing)?
> >>>
> >>> There are these settings setinterfacevar, setqueueentryvar,
> >>> setqueuevar in queues.conf but when can I use them?
> >>>
> >>> Have you guys been in this situation before? Any alternative solutions
> >>> (sending caller info to an agent)?
> >>>
> >>> I am using Asterisk 11 and Windows 7 PCs for agents.
> >>>
> >>> Thank you!
> >>>
> >>> Kind Regards,
> >>> Wilson
> >>>
> >>> --
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> >>> http://www.asterisk.org/hello
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> >>>
> >>
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> >
> >
> >
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Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Timothy Smith]

Dear Tiago,

Thanks for your answer, but I have a few questions.

Do you use queues? We are operating a call centre with several queues,
so I don't see how we would use the Dial command. When a call comes
in, we enter the caller (depending on what options he has selected)
into a queue. Do you have any alternative method, which would involve
dialling the agent directly as you described below?

regards,
T

On Sun, Aug 4, 2013 at 3:47 PM, Tiago Geada  wrote:
> Hi,
>
> Our queue members are Local channels, thus when dialing the agent, the
> dialplan will do several stuff including:
>
> Set(CALLERID(name)=${CALLERID(name)}:Sales)
> UserEvent(something,data: ${bunch-of-data-in-some-format})
> Dial(SIP/final-agent-phone,timeout,A(Sales))
>
> The UserEvent will be picked up by our client-register-ticket-stuff software
>
> The announcement A() will be heard by the agent upon answering the call like
> "sales call"
>
>
> On 4 August 2013 02:59, Mitch Claborn  wrote:
>>
>> We do something very similar.
>>
>> Use the gosub parameter of the Queue application to call a subroutine in
>> the dial plan when the agent answers the call.
>>
>> same =>n,Queue(sales,tc,,sub-QueueConnected)
>>
>> [sub-QueueConnected]
>> ; this runs on the agent/member's channel
>> exten =>s,1,NoOp()
>>   ; whatever you need to do here
>>   same =>n,Return()
>>
>> See
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue
>>
>>
>> Mitch
>>
>>
>> On 08/03/2013 12:45 PM, Timothy Smith wrote:
>>>
>>> Hello Folks,
>>>
>>> I am setting up a call center but we have few agents so one agent is
>>> able to handle calls of different languages and different queues. For
>>> the agent to identify the caller, I want a popup to appear as the
>>> phone starts to ring with the caller's number, language (selected in
>>> the IVR), Queue (sales, support etc) and any other information (e.g a
>>> URL with parameters)
>>>
>>> I can send this information either via netcat (to a client such as
>>> yac) to a Windows PC but the problem is I do not know when the caller
>>> is about to be connected to the agent, so that I run the command. If I
>>> wasn't using queues, it would be easy because  I would run the netcat
>>> command and then dial the user's extension.
>>>
>>> My Question is: Is there a way I can know when the caller is just
>>> about to be connected to an agent (when the agent's SIP extension
>>> starts ringing)?
>>>
>>> There are these settings setinterfacevar, setqueueentryvar,
>>> setqueuevar in queues.conf but when can I use them?
>>>
>>> Have you guys been in this situation before? Any alternative solutions
>>> (sending caller info to an agent)?
>>>
>>> I am using Asterisk 11 and Windows 7 PCs for agents.
>>>
>>> Thank you!
>>>
>>> Kind Regards,
>>> Wilson
>>>
>>> --
>>> _
>>> -- 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 --
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>>   http://www.asterisk.org/hello
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Timothy Smith]

Dear Mitch,

Thank you so much. This partly solves my problem by a great deal, as
we'll send a message to the agent immediately on picking the call. As
the agents are local SIP channels, I will attempt looking up the
caller's name (if it exists in our database) and set it prior to
entering the queue.

Is there any way of informing the agent (just) before they pick up?
e.g when their phone starts ringing, so that they prepare accordingly?

Regards,
Wilson

On Sun, Aug 4, 2013 at 4:59 AM, Mitch Claborn  wrote:
> We do something very similar.
>
> Use the gosub parameter of the Queue application to call a subroutine in the
> dial plan when the agent answers the call.
>
> same =>n,Queue(sales,tc,,sub-QueueConnected)
>
> [sub-QueueConnected]
> ; this runs on the agent/member's channel
> exten =>s,1,NoOp()
>   ; whatever you need to do here
>   same =>n,Return()
>
> See https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue
>
>
> Mitch
>
>
> On 08/03/2013 12:45 PM, Timothy Smith wrote:
>>
>> Hello Folks,
>>
>> I am setting up a call center but we have few agents so one agent is
>> able to handle calls of different languages and different queues. For
>> the agent to identify the caller, I want a popup to appear as the
>> phone starts to ring with the caller's number, language (selected in
>> the IVR), Queue (sales, support etc) and any other information (e.g a
>> URL with parameters)
>>
>> I can send this information either via netcat (to a client such as
>> yac) to a Windows PC but the problem is I do not know when the caller
>> is about to be connected to the agent, so that I run the command. If I
>> wasn't using queues, it would be easy because  I would run the netcat
>> command and then dial the user's extension.
>>
>> My Question is: Is there a way I can know when the caller is just
>> about to be connected to an agent (when the agent's SIP extension
>> starts ringing)?
>>
>> There are these settings setinterfacevar, setqueueentryvar,
>> setqueuevar in queues.conf but when can I use them?
>>
>> Have you guys been in this situation before? Any alternative solutions
>> (sending caller info to an agent)?
>>
>> I am using Asterisk 11 and Windows 7 PCs for agents.
>>
>> Thank you!
>>
>> Kind Regards,
>> Wilson
>>
>> --
>> _
>> -- 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
>>
>
> --
> _
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Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Tiago Geada]

Hi,

Our queue members are Local channels, thus when dialing the agent, the
dialplan will do several stuff including:

Set(CALLERID(name)=${CALLERID(name)}:Sales)
UserEvent(something,data: ${bunch-of-data-in-some-format})
Dial(SIP/final-agent-phone,timeout,A(Sales))

The UserEvent will be picked up by our client-register-ticket-stuff software

The announcement A() will be heard by the agent upon answering the call
like "sales call"


On 4 August 2013 02:59, Mitch Claborn  wrote:

> We do something very similar.
>
> Use the gosub parameter of the Queue application to call a subroutine in
> the dial plan when the agent answers the call.
>
> same =>n,Queue(sales,tc,,sub-**QueueConnected)
>
> [sub-QueueConnected]
> ; this runs on the agent/member's channel
> exten =>s,1,NoOp()
>   ; whatever you need to do here
>   same =>n,Return()
>
> See https://wiki.asterisk.org/**wiki/display/AST/Asterisk+11+**
> Application_Queue
>
>
> Mitch
>
>
> On 08/03/2013 12:45 PM, Timothy Smith wrote:
>
>> Hello Folks,
>>
>> I am setting up a call center but we have few agents so one agent is
>> able to handle calls of different languages and different queues. For
>> the agent to identify the caller, I want a popup to appear as the
>> phone starts to ring with the caller's number, language (selected in
>> the IVR), Queue (sales, support etc) and any other information (e.g a
>> URL with parameters)
>>
>> I can send this information either via netcat (to a client such as
>> yac) to a Windows PC but the problem is I do not know when the caller
>> is about to be connected to the agent, so that I run the command. If I
>> wasn't using queues, it would be easy because  I would run the netcat
>> command and then dial the user's extension.
>>
>> My Question is: Is there a way I can know when the caller is just
>> about to be connected to an agent (when the agent's SIP extension
>> starts ringing)?
>>
>> There are these settings setinterfacevar, setqueueentryvar,
>> setqueuevar in queues.conf but when can I use them?
>>
>> Have you guys been in this situation before? Any alternative solutions
>> (sending caller info to an agent)?
>>
>> I am using Asterisk 11 and Windows 7 PCs for agents.
>>
>> Thank you!
>>
>> Kind Regards,
>> Wilson
>>
>> --
>> __**__**_
>> -- 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
>>
>>
> --
> __**__**_
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>   
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Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

[Larry Moore]

On 01/08/2013, at 2:20 PM, Zoltán Fekete  wrote:

> 
> 2013/8/1 Joshua Colp 
> Larry Moore wrote:
> On 31/07/2013 8:08 PM, Joshua Colp wrote:
> Zoltán Fekete wrote:
> Thank You Larry!
> 
> I have discussed with my provider. They are not able to insert the
> T38MaxBitRate value into the sip answer. :(
> https://gist.github.com/anonymous/6120148 (line 559)
> 
> That means we are not able to passtrough T38 Faxes with any asterisk
> version at all?
> What do you mean? Am I able to modify and compile the source? Is it
> compicated? (I'm not a developer)
> 
> Based on the SDP in your gist the remote implementation has given no
> attributes with the T.38 stream which makes it pretty broken
> (T38FaxRateManagement is mandatory) and fun. The two hard parts really
> would be 1. Modifying Asterisk in a sane fashion to cope and 2.
> Determining the exact settings to make the implementation happy.
> Defaults as defined in the spec are fine and good, but my experience has
> taught me to throw those out the window when it comes to actual
> implementations.
> 
> 
> It would seem that having a configurable option would be an idea for
> this scenario.
> 
> That implies it would solve the problem, which my gut and experience tells 
> me... it wouldn't. I think the T.38 implementation is just cobbled together 
> and without knowing exactly how it behaves getting it to work would likely be 
> a nightmare (trust me, I've spent time in those deep dark reaches). Throwing 
> assumptions and defaults at it to try to make it work is of course an option.
> 
> My testing with Asterisk 1.8 and T.38, I obserevd that setting
> FAXOPT(minrate) or FAXOPT(maxrate)had no effect, I concluded that when
> Astrerisk is receiving it uses hard coded values - is this a sane thing
> to do?!
> 
> When Asterisk is receiving the stack implementation offers what it wants, 
> with the ability to override. So Asterisk doesn't hard code those values, the 
> stack provides them. What is hard coded is the default values if none are 
> received.
> 
> I would even say it's a bug that the negotiation doesn't fail, since the 
> remote side isn't providing a mandatory attribute.
> 
> -- 
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
> 
> --
> _
> 
> 
> Yes you're right! As I know FAXOPT() value affect only when asterisk woks as 
> gateway. 
> We need passtrouh because my endpoints and also my provider supports T38.
> 
> https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
> "Using T.38 Gateway mode
> T.38 Gateway mode should be used when one leg of a call is not capable of 
> T.38 mode. In the event that both legs are capable and Gateway mode is 
> configured, then the Gateway will step out of the way, allowing transparent 
> T.38 passthrough."
> 
> The main problem is that I can't use G711 for the entire fax session because 
> the endpoints has 20-30ms response time.
> 
> When I try to use my Asterisk as FAXOPT gateway (endpoint leg T38 and 
> provider leg G711) can I force somehow to not accept the T38 re-INVITEs from 
> the provider? 
> They have ~1ms response time, so G711 on that leg would be fine but they also 
> detect fax CED tones and sends the re-INVITEs.
> 

Have you tried setting in your sip.conf for your provider t38pt_udptl=no whilst 
having the gateway option enabled?

Sorry I cant test this myself.

Cheers,

Larry.--
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