[asterisk-users] 2 pretty irritating issues....

[Gregory Malsack]

Hey All ~

1, queue records on fairly unreliable. I would say about 40 - 60 percent 
of the queue calls are not being recorded and I'm not sure why. I don't 
seem to see any kind of pattern to the failure. I've added a sample of 
our queue config at the bottom.


2, cel_pgsql module seems to crash regularly. It seems every time I look 
at our asterisk server, the cel_pgsql module is gone. A simple reload 
will not bring it back. I have to unload the module and load the module. 
I've seen this so many times that I have a cron job now in 
/etc/cron.hourly to perform this task. However, that doesn't seem to be 
helping because we are still missing a crap load of cel records from the 
db. I run the IT for a national call center and people are constantly 
asking what happened with a call. So I got sick of checking 
/var/log/asterisk/full to explain the entire call flow, so I built a web 
interface that connected to the cel database that they could use to find 
it for themselves. With the cel_pgsql mod crapping out all the time, I 
feel like that was a waste of my time


Thanks! Look forward to your input...
Gregory Malsack

Ok now for the meat

Asterisk Server
Dell 1950 Dual Quad Core Xeon 2.33ghz
16gb Ram
CentOS 5.8 x86_64
Asterisk 1.8.22.0

Recordings are stored on a separate server via NFS with a 1gb connection

SQL Server
Dell 2950 Dual Dual Core Xeon 2.66ghz
16gb Ram
CentOS 5.8 x86_64

One of our queues.

[Confirmation]
; extension  8666
announce-frequency=0
announce-holdtime=no
announce-position=no
eventmemberstatus=no
eventwhencalled=no
joinempty=yes
leavewhenempty=no
memberdelay=0
monitor-type=mixmonitor
monitor-format=gsm
penaltymemberslimit=0
periodic-announce-frequency=0
reportholdtime=no
retry=5
ringinuse=no
servicelevel=90
strategy=ringall
timeout=10
timeoutpriority=app
timeoutrestart=no
weight=0
wrapuptime=0


How a call gets to the queue.

exten = 8666,1,NoOp(Confirmation Queue <-> ${CALLERID(all)})
exten = 8666,n,GotoIfTime(${EH_HOURS}?:ehc_closed,s,1)
exten = 8666,n,Set(MONTH=${STRFTIME(${EPOCH},,%b)})
exten = 8666,n,Set(DAY=${STRFTIME(${EPOCH},,%-d)})
exten = 8666,n,Set(MONITOR_FILENAME=${MONTH}/${DAY}/8666-${UNIQUEID})
exten = 8666,n,Set(CDR(userfield)=href=http://pas.coastalacq.loc/recordings/${MONITOR_FILENAME}.gsm>8666-${UNIQUEID})

exten = 8666,n,Queue(Confirmation)



--


Greg


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Re: [asterisk-users] auto answer

[Steve Edwards]

Please don't top post.

On Thu, 18 Jul 2013, Gopalakrishnan N wrote:

If am not wrong even without doing any setting in asterisk side, if the 
phone has Auto Answer it works.. ! Correct me if am wrong. 


My experience is limited to Sipura, Polycom, and Cisco endpoints. All 
required configuration on the phone to enable the feature AND the addition 
of an 'endpoint specific' SIP header to ask the phone to auto-answer in 
the dialplan.


--
Thanks in advance,
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Newline  Fax: +1-760-731-3000--
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Re: [asterisk-users] auto answer

[Gopalakrishnan N]

If am not wrong even without doing any setting in asterisk side, if the
phone has Auto Answer it works.. !

Correct me if am wrong.


On Wed, Jul 17, 2013 at 9:14 PM, Steve Edwards wrote:

> Please don't top post.
>
>
> On Wed, 17 Jul 2013, bilal ghayyad wrote:
>
>  So it is not at asterisk configuration?
>>
>
> 1) The phone has to be configured to allow it.
>
> 2) Asterisk has to set the appropriate SIP header for your specific model
> phone prior to 'dialing' the phone for each call. I.e. the added SIP header
> for a Cisco is different than for a Polycom.
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
>
> --
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[asterisk-users] Dial problem with Asterisk 1.8.4.4

[Kevin Larsen]

One of my sites asked for a way to identify if the person they are calling 
on another extension is already on another call. To that end, I wrote a 
bit of code in the dialplan for my extensions that checks to see if the 
extension they are dialing has a device status that is anything other than 
NOT_INUSE. If the device is NOT_INUSE, then it dials the call normally. If 
it has a different status, then it dials using the option m(ringing) where 
ringing is a musiconhold class that plays a standard ring with a beep at 
the end so that they have an audible clue that the other person is on a 
call. This works well in most cases, but we have found one case that seems 
to cause an issue. If they try to blind transfer a call to an extension 
that is on another call, then the person being transferred does not get 
the ringing musiconhold played to them. 

The basic call flow is that a call comes in from an outside trunk and is 
answered by person A. They then hit the transfer button on their phone 
(Polycom IP 450) and dial person B. During this time, the caller hears the 
standard musiconhold. Person A hears the audio they should hear based on 
the m(ringing) option for the Dial application. Person A then hits 
transfer again to finish a blind transfer. At this point, the musiconhold 
that the caller hears cuts out and is not replaced by the m(ringing) 
audio. Any thoughts on if it is possible to make this work?

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208--
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Re: [asterisk-users] auto answer

[Steve Edwards]

Please don't top post.

On Wed, 17 Jul 2013, bilal ghayyad wrote:


So it is not at asterisk configuration?


1) The phone has to be configured to allow it.

2) Asterisk has to set the appropriate SIP header for your specific model 
phone prior to 'dialing' the phone for each call. I.e. the added SIP 
header for a Cisco is different than for a Polycom.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] auto answer

[Pat Collins]

Dialplan auto answer

; intercom

exten=_*,1,SIPAddHeader(Call-Info:
\;answer-after=0) ;xxx.xxx.xxx.xxx is the address of
your asterisk box

exten=_*,n,Dial(SIP/${EXTEN:1})

 

As long as your phones are compatible, this MIGHT work.

Worked for me.  Sadly, I cannot recall which phones we were using.  Long
time ago.

Hope it helps,

Pat

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan
N
Sent: Wednesday, July 17, 2013 9:02 AM
To: bilal ghayyad; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] auto answer

 

yes its not asterisk configuration, its phone feature and phone
configuration. 

 

On Wed, Jul 17, 2013 at 3:27 PM, bilal ghayyad  wrote:

So it is not at asterisk configuration?

 

Regards

Bilal

 

  _  

From: A J Stiles 


To: bilal ghayyad ; Asterisk Users Mailing List -
Non-Commercial Discussion  

Sent: Wednesday, July 17, 2013 12:57 PM


Subject: Re: [asterisk-users] auto answer


On Wednesday 17 July 2013, bilal ghayyad wrote:
> But this not in the sip.conf, this in the extensions.conf, right?
> 
> Regards
> Bilal

No.  This would be set up in the phone's own configuration file, which in
turn 
depends on the make and model of phone  (and its location depends on your
site 
setup).

-- 
AJS

Answers come *after* questions.




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Re: [asterisk-users] auto answer

[Gopalakrishnan N]

yes its not asterisk configuration, its phone feature and phone
configuration.


On Wed, Jul 17, 2013 at 3:27 PM, bilal ghayyad  wrote:

> So it is not at asterisk configuration?
>
> Regards
> Bilal
>
>   --
>  *From:* A J Stiles 
>
> *To:* bilal ghayyad ; Asterisk Users Mailing List -
> Non-Commercial Discussion 
> *Sent:* Wednesday, July 17, 2013 12:57 PM
>
> *Subject:* Re: [asterisk-users] auto answer
>
> On Wednesday 17 July 2013, bilal ghayyad wrote:
> > But this not in the sip.conf, this in the extensions.conf, right?
> >
> > Regards
> > Bilal
>
> No.  This would be set up in the phone's own configuration file, which in
> turn
> depends on the make and model of phone  (and its location depends on your
> site
> setup).
>
> --
> AJS
>
> Answers come *after* questions.
>
>
>
> --
> _
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Re: [asterisk-users] auto answer

[bilal ghayyad]

So it is not at asterisk configuration?

Regards
Bilal



 From: A J Stiles 
To: bilal ghayyad ; Asterisk Users Mailing List - 
Non-Commercial Discussion  
Sent: Wednesday, July 17, 2013 12:57 PM
Subject: Re: [asterisk-users] auto answer
 

On Wednesday 17 July 2013, bilal ghayyad wrote:
> But this not in the sip.conf, this in the extensions.conf, right?
> 
> Regards
> Bilal

No.  This would be set up in the phone's own configuration file, which in turn 
depends on the make and model of phone  (and its location depends on your site 
setup).

-- 
AJS

Answers come *after* questions.--
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Re: [asterisk-users] auto answer

[A J Stiles]

On Wednesday 17 July 2013, bilal ghayyad wrote:
> But this not in the sip.conf, this in the extensions.conf, right?
> 
> Regards
> Bilal

No.  This would be set up in the phone's own configuration file, which in turn 
depends on the make and model of phone  (and its location depends on your site 
setup).

-- 
AJS

Answers come *after* questions.

--
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Re: [asterisk-users] auto answer

[bilal ghayyad]

But this not in the sip.conf, this in the extensions.conf, right?

Regards
Bilal



 From: Yasin Suluhan 
To: bilal ghayyad ; Asterisk Users Mailing List - 
Non-Commercial Discussion  
Sent: Wednesday, July 17, 2013 12:21 PM
Subject: Re: [asterisk-users] auto answer
 


Hello, 

You could use Answer-After for that. But, afaik there is no definitive 
description in the RFCs on how it is used. 

You would have to enable such features on the telephones too. And I would 
expect that different phone manufacturers would probably use different 
mechanisms to enable such an option. 

Furthermore, considering the security issues this would create i wouldn' t 
recommend taking such a path. 



On Wed, Jul 17, 2013 at 12:04 PM, bilal ghayyad  wrote:

Hello;
>
>
>Is it possible to configure in the sip.conf for the Phone to be auto answer?
>
>
>RegardsBilal
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-- 

Best Regards.


Yasin SULUHAN
Contact Information
Mobile: +90 535 656 35 55--
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Re: [asterisk-users] auto answer

[A J Stiles]

On Wednesday 17 July 2013, bilal ghayyad wrote:
> Hello;
> 
> Is it possible to configure in the sip.conf for the Phone to be auto
> answer?

That is a phone feature, not an Asterisk feature  (in fact, not all phones 
even support it.  A GPO 746 plugged into a Grandstream HandyTone 286, for 
instance -- well, OK, that's a bit of an extreme case).  You would have to put 
it in the phone's configuration file  (which is usually sent to the phone by 
TFTP or HTTP, after it obtains an IP address by DHCP).  The config file usually 
is named after the MAC address of the phone, with a manufacturer-dependent 
extension.

You really need to refer to the manual for the phones you are using.

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] auto answer

[Yasin Suluhan]

Hello,

You could use Answer-After for that. But, afaik there is no definitive
description in the RFCs on how it is used.

You would have to enable such features on the telephones too. And I would
expect that different phone manufacturers would probably use different
mechanisms to enable such an option.

Furthermore, considering the security issues this would create i wouldn' t
recommend taking such a path.


On Wed, Jul 17, 2013 at 12:04 PM, bilal ghayyad  wrote:

> Hello;
>
> Is it possible to configure in the sip.conf for the Phone to be auto
> answer?
>
> Regards
> Bilal
>
> --
> _
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>



-- 
Best Regards.

Yasin SULUHAN
Contact Information
Mobile: +90 535 656 35 55


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[asterisk-users] auto answer

[bilal ghayyad]

Hello;

Is it possible to configure in the sip.conf for the Phone to be auto answer?

Regards
Bilal--
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