Re: [asterisk-users] Realtime Hints

[Leandro Dardini]

Thank you again,
The problem in this setup is the inability to isolate a group of extensions
from others. I mean, if all hints are in the same context, each extension
can subscribe to any of the hints. The reason I prefer a completely
realtime hints was I'd like to dynamically create hints in dynamically
created context to jail an extension to only the hints I decide.

I have tried to contact Digium on this topic and I am ready to setup a
bounty to have a true realtime hints management coded if not already
available.

Leandro
Il giorno 26/set/2012 00:03, "Stephen Collier" 
ha scritto:

> I use the following in func_odbc.conf
>
> [FINDEXTN]
> dsn=asterisk
> readsql=SELECT ${ARG1} FROM extension_map as em left join sip_devices as
> sd on s
> d.id = em.name_id WHERE em.extension ='${ARG2}' and name_id IS NOT NULL
>
> this is for our own extension_map table which is part of our mapping to
> our Avaya users.
>
> A simple one would be
> [FINDEXTN]
> dsn=asterisk
> readsql=SELECT ${ARG1} FROM sip_devices as sd WHERE sd.name ='${ARG2}'
>
> This allows pulling any field from sip_devices which is our realtime sip
> table.
>
> You could pull some of the other data you are looking for.
>
> Cheers
> Stephen
>
>
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Re: [asterisk-users] asterisk-users Digest, Vol 98, Issue 38

[pankaj pandey]

Hi Danny,

Thank you for your prompt response.
The way you are suggesting is great . Infect asterisk have its own 
functionality that if user presses *1 during meetme conferencing asterisk 
automatically unmute that user and user comes in talking mode.But it is 
not fulfill my need.
There is and issue that if 3-4 user presses *1 at the same time than how can i 
decide that who is asking the question and how can we manage that situation.
Please suggest the another way to doing this.
  

-Thanks !!!

Message: 6

Date: Tue, 25 Sep 2012 13:15:32 -0500
From: "Danny Nicholas" 
Subject: Re: [asterisk-users] Hand Raise|Meetme Conf
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
    
Message-ID: <00df01cd9b49$c03b44c0$40b1ce40$@debsinc.com>
Content-Type: text/plain;    charset="iso-8859-1"

This might work:
[meetme-with-handraise]
Exten => s,1,meetme(1234,mX5)
Exten => s,n,hangup
Exten => 5,1,meetme(1234)
Exten => 5,2,goto(meetme-with-handraise,s,1)
Exten => I,1,playback(invalid)
Exten => I,n,goto(meetme-with-handraise,s,1)

According to the documentation, if the user presses 5, it should end their
muted session and put them back in a "talking" mode.  You could use X4 to
return them to muted mode by pressing 4.
Haven't tested it, but it's not a difficult thing to try.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of pankaj pandey
Sent: Tuesday, September 25, 2012 12:56 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hand Raise|Meetme Conf



Hi All,
I am am looking for the below feature of asterisk MEETME. 
I googled a lot but did not find any help.
Any body please suggest, how can we do it.  

Thanks!!! 
--
On Thu 20 Sep, 2012 3:11 PM EDT pankaj pandey wrote:

>Hi All,
>I have a?scenario where leader is giving a lecture and other participants
are on mute...
>
>At the end of conf , when Q&A session begins is there a way for
participants to raise hands if they have questions so Leader can unmute
them. Is this feature already there?in Meetme conf?? If there then how can
i?implement this.
>
>please?suggest...
>
>?
>Thanks & Regards,
>Pankaj Pandey
>+91-9990212758


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Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR

[Matt Hamilton]

Our top priority is the raw Write (INSERT) performance, Read (SELECT) 
performance is not important. Strict ACID compliance is not necessary either. 
MySQL (on a separate database server) should be able to handle inserting CDR 
records (approximately up to 10 records for each call) for about 1000 
concurrent calls coming from an Asterisk cluster. 
 
Matt


Date: Tue, 25 Sep 2012 18:19:50 -0500
From: lo...@keobi.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR

Very good point. For revenue critical data like CDRs, being ACID compliant is 
important.
MyISAM is compliant. And like InnoDB, can have the features making it compliant 
turned off. 
On Sep 25, 2012 6:12 PM, "Patrick Lists"  
wrote:

On 09/25/2012 11:18 PM, Logan Bibby wrote:


MyISAM would be best, in my opinion. The features that cause the little

bit of performance overhead in InnoDB wouldn't be necessary for CDR storage.




Iirc InnoDB is ACID compliant so might be preferable if MyISAM is not. More 
information here:



http://en.wikipedia.org/wiki/ACID



https://blogs.oracle.com/MySQL/entry/comparing_innodb_to_myisam_performance



Regards,

Patrick





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Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR

[Logan Bibby]

Very good point. For revenue critical data like CDRs, being ACID compliant
is important.

MyISAM is compliant. And like InnoDB, can have the features making it
compliant turned off.
On Sep 25, 2012 6:12 PM, "Patrick Lists" 
wrote:

> On 09/25/2012 11:18 PM, Logan Bibby wrote:
>
>> MyISAM would be best, in my opinion. The features that cause the little
>> bit of performance overhead in InnoDB wouldn't be necessary for CDR
>> storage.
>>
>
> Iirc InnoDB is ACID compliant so might be preferable if MyISAM is not.
> More information here:
>
> http://en.wikipedia.org/wiki/**ACID 
>
> https://blogs.oracle.com/**MySQL/entry/comparing_innodb_**
> to_myisam_performance
>
> Regards,
> Patrick
>
>
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Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR

[Patrick Lists]

On 09/25/2012 11:18 PM, Logan Bibby wrote:

MyISAM would be best, in my opinion. The features that cause the little
bit of performance overhead in InnoDB wouldn't be necessary for CDR storage.


Iirc InnoDB is ACID compliant so might be preferable if MyISAM is not. 
More information here:


http://en.wikipedia.org/wiki/ACID

https://blogs.oracle.com/MySQL/entry/comparing_innodb_to_myisam_performance

Regards,
Patrick


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Re: [asterisk-users] Realtime Hints

[Stephen Collier]

I use the following in func_odbc.conf

[FINDEXTN]
dsn=asterisk
readsql=SELECT ${ARG1} FROM extension_map as em left join sip_devices as sd on s
d.id = em.name_id WHERE em.extension ='${ARG2}' and name_id IS NOT NULL

this is for our own extension_map table which is part of our mapping to our 
Avaya users. 

A simple one would be 
[FINDEXTN]
dsn=asterisk
readsql=SELECT ${ARG1} FROM sip_devices as sd WHERE sd.name ='${ARG2}'

This allows pulling any field from sip_devices which is our realtime sip table.

You could pull some of the other data you are looking for.

Cheers
Stephen


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Re: [asterisk-users] no audio while call forwarding, yes audio with followme

[Danny Nicholas]

Do the call both ways again and check(post) the CLI output.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bart Coninckx
Sent: Tuesday, September 25, 2012 4:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] no audio while call forwarding, yes audio with
followme

 

Hi all,

the subject says it all.
Technical details: 
- Asterisk 1.8.7.1 
- Behind NAT 
- Using external SIP provider

The call forwarding is tested both with this functionality on the phone and
with configuration in the dialplan. In the latter case a database variable
is set to the external number, if set a Dial command calls this number. So
really nothing fancy (actually I followed the example on
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding ).

sip.conf has nat=yes, externip= ... and I tried every setting of directmedia
in the providers configuration part. 

Followme works flawlessly, so I'm really wondering if this is a NAT issue. 


Can anyone point me into a certain direction?


Thx


BC

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[asterisk-users] no audio while call forwarding, yes audio with followme

[Bart Coninckx]

Hi all,

the subject says it all.
Technical details:
- Asterisk 1.8.7.1
- Behind NAT
- Using external SIP provider

The call forwarding is tested both with this functionality on the phone 
and with configuration in the dialplan. In the latter case a database 
variable is set to the external number, if set a Dial command calls this 
number. So really nothing fancy (actually I followed the example on 
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding ).


sip.conf has nat=yes, externip= ... and I tried every setting of 
directmedia in the providers configuration part.


Followme works flawlessly, so I'm really wondering if this is a NAT issue.


Can anyone point me into a certain direction?


Thx


BC

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Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR

[Logan Bibby]

MyISAM would be best, in my opinion. The features that cause the little bit
of performance overhead in InnoDB wouldn't be necessary for CDR storage.

- Logan
On Sep 25, 2012 4:15 PM, "Matt Hamilton"  wrote:

> Which one (InnoDB or MyISAM) is preferred for CDR as far as write
> performance is concerned?
>
> Thanks,
> Matt
>
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[asterisk-users] MySQL InnoDB or MyISAM for CDR

[Matt Hamilton]

Which one (InnoDB or MyISAM) is preferred for CDR as far as write performance 
is concerned?

Thanks,
Matt
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Re: [asterisk-users] confbridge video support

[Leif Madsen]

On 25/09/12 10:18 AM, Bryant Zimmerman wrote:

Where does video support for confbridge stand? I need to be able to take
in multiple video callers and have the active speaking caller displayed
to all participants. Are we there yet in Asterisk 10 or 11?


Yes, both Asterisk 10 and 11 support this in ConfBridge(). Look at 
video_mode=follow_talker



--
Leif Madsen
http://www.oreilly.com/catalog/asterisk

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Re: [asterisk-users] T.38 gateway ATA

[Jeff LaCoursiere]

On 09/25/2012 03:02 PM, Bryant Zimmerman wrote:

Jeff

The grandstream units support both Relay and passthru depending on how 
you set them up.


Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003


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Right -

So here is a summary so far:

Definitely supports "T.38 Fax Relay" to normal fax machines: OBI202, 
Cisco SPA8000, Cisco SPA 3102, Grandstream HT-701


There may be more.  I'm personally testing Cisco SPA122 with fingers 
crossed.  Hoping this information gets properly indexed, as I had a !@#@ 
of a time finding anything like this list myself.


Cheers,

j
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Re: [asterisk-users] T.38 gateway ATA

[Bryant Zimmerman]

Jeff

The grandstream units support both Relay and passthru depending on how you 
set them up. 

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
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[asterisk-users] asterisk ip authentication

[jin jan]

Hi all,
I've tried to sen calls to asterisk from different soft switch.
I want to define ip authentication(not register) to an extension for make
call through asterisk.
Is there any way to make call from asterisk  without register. Only ip
authentication.
Kind Regards.
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Re: [asterisk-users] T.38 gateway ATA

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Tuesday, September 25, 2012 2:49 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] T.38 gateway ATA

On 09/25/2012 10:44 AM, Danny Nicholas wrote:
> I would recommend the OBI110.
>
>
>
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Had a chat with their support - apparently the OBI202 supports "T.38 Fax
Relay", which is (I learned today) the term for the functionality I am
looking for.  The lower models do NOT support this, only passthru.  
Sadly the OBI202 is kind of pricy.  And has an odd USB port... what would
that be useful for I wonder?

Cheers,

j

Wondered about the USB port myself.  Looked this up -
http://www.obihai.com/docs/OBi202DS.pdf ; you can hook the device up to your
wifi network.


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Re: [asterisk-users] T.38 gateway ATA

[Jeff LaCoursiere]

On 09/25/2012 10:44 AM, Danny Nicholas wrote:

I would recommend the OBI110.



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Had a chat with their support - apparently the OBI202 supports "T.38 Fax 
Relay", which is (I learned today) the term for the functionality I am 
looking for.  The lower models do NOT support this, only passthru.  
Sadly the OBI202 is kind of pricy.  And has an odd USB port... what 
would that be useful for I wonder?


Cheers,

j

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Re: [asterisk-users] Realtime Hints

[Leandro Dardini]

Thank you,
I think I'll surrender in trying to use the realtime extension and use
instead the simple ODBC interface. However I'd like to access some channel
variables. Which ones are available inside the extension hint porcessing? I
tried ${CDR(accountcode)} and it is not available, nor the
${CHANNEL(peername)} ... what is the sd.name that you are referring?

Leandro

2012/9/25 Stephen Collier 

>
> We use something like below
>
> [blf]
> exten =>_ZXX!,hint,SIP/${ODBC_FINDEXTN(sd.name,${EXTEN})}
>
>
> This uses an odbc call to create the hint when the phone asks for it.
> Using snom 760 and 821
>
> Cheers
> Stephen
>
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Re: [asterisk-users] Hand Raise|Meetme Conf

[Danny Nicholas]

This might work:
[meetme-with-handraise]
Exten => s,1,meetme(1234,mX5)
Exten => s,n,hangup
Exten => 5,1,meetme(1234)
Exten => 5,2,goto(meetme-with-handraise,s,1)
Exten => I,1,playback(invalid)
Exten => I,n,goto(meetme-with-handraise,s,1)

According to the documentation, if the user presses 5, it should end their
muted session and put them back in a "talking" mode.  You could use X4 to
return them to muted mode by pressing 4.
Haven't tested it, but it's not a difficult thing to try.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of pankaj pandey
Sent: Tuesday, September 25, 2012 12:56 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hand Raise|Meetme Conf



Hi All,
I am am looking for the below feature of asterisk MEETME. 
I googled a lot but did not find any help.
Any body please suggest, how can we do it.   

Thanks!!! 
--
On Thu 20 Sep, 2012 3:11 PM EDT pankaj pandey wrote:

>Hi All,
>I have a scenario where leader is giving a lecture and other participants
are on mute...
>
>At the end of conf , when Q&A session begins is there a way for
participants to raise hands if they have questions so Leader can unmute
them. Is this feature already there in Meetme conf ? If there then how can
i implement this.
>
>please suggest...
>
> 
>Thanks & Regards,
>Pankaj Pandey
>+91-9990212758

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Re: [asterisk-users] Help Required IVR

[Danny Nicholas]

You should be able to do a playback or background command before launching 
voicemail

 

[voicemail-connect]

Exten => s,1,playback(greeting)

Exten => s,n,voicemail(100@default,u)

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Tuesday, September 25, 2012 5:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help Required IVR

 

Are you programming the dialplan yourself or are you using a GUI? Also, are you 
sure that your greetings message is playable by Asterisk?

l.

 

 

2012/9/24 Farooq Hussain 

Hello everyone,

 

I stuck in problem I have creating a time based IVR and its working fine. If my 
IVR playing in office hour it would standard IVR and if not they we have play a 
greeting message and place that call to voice mail of a extension.

 

My problem is this I am able to transfer the call on voice mail but how to play 
greeting message first. I am using trixbox 2.2.8 anyone help is this regard 
would great full.


 

-- 
Thanks

Farooq Hussain


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-- 

Loway - home of QueueMetrics - http://queuemetrics.com

Test-drive WombatDialer beta @ http://wombatdialer.com 

 

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[asterisk-users] Hand Raise|Meetme Conf

[pankaj pandey]

Hi All,
I am am looking for the below feature of asterisk MEETME. 
I googled a lot but did not find any help.
Any body please suggest, how can we do it.   

Thanks!!! 
--
On Thu 20 Sep, 2012 3:11 PM EDT pankaj pandey wrote:

>Hi All,
>I have a scenario where leader is giving a lecture and other participants are 
>on mute...
>
>At the end of conf , when Q&A session begins is there a way for participants 
>to raise hands if they have questions so Leader can unmute them. Is this 
>feature already there in Meetme conf ? If there then how can i implement this.
>
>please suggest...  
>
> 
>Thanks & Regards,
>Pankaj Pandey
>+91-9990212758

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[asterisk-users] Hand Raise|Meetme Conf

[pankaj pandey]

Hi All,
I am am looking for the below feature of asterisk MEETME. 
I googled a lot but did not find any help.
Any body please suggest, how can we do it.   

Thanks!!! 
--
On Thu 20 Sep, 2012 3:11 PM EDT pankaj pandey wrote:

>Hi All,
>I have a scenario where leader is giving a lecture and other participants are 
>on mute...
>
>At the end of conf , when Q&A session begins is there a way for participants 
>to raise hands if they have questions so Leader can unmute them. Is this 
>feature already there in Meetme conf ? If there then how can i implement this.
>
>please suggest...  
>
> 
>Thanks & Regards,
>Pankaj Pandey
>+91-9990212758

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Re: [asterisk-users] T.38 gateway ATA

[Bryant Zimmerman]

 From: "Jeff LaCoursiere" 
Sent: Tuesday, September 25, 2012 11:40 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] T.38 gateway ATA

On 09/25/2012 10:34 AM, Sylvain Rochet wrote:
> Hi Jeff,
>
>
> On Tue, Sep 25, 2012 at 10:11:27AM -0500, Jeff LaCoursiere wrote:
>> Aha! That is what I am looking for. What firmware rev are you
>> using on the 3102?
> 5.1.12 on SPA8000, 5.1.10(GW) on SPA3102, there is probably new releases
> from Cisco, at the time I builded the T.38 network those were the latest
> releases.
>
> Here is my SPA3102 configuration that I changed or are relevant to 
FAX/T.38:
>
> Network Jitter Level: very high
> Jitter Buffer Adjustment: disable
> Call Waiting Serv: no
> Three Way Conf Serv: no
> Echo Canc Enable: no
> Echo Canc Adapt Enable: no
> Echo Supp Enable: no
> FAX CED Detect Enable: yes
> FAX CNG Detect Enable: yes
> FAX Passthru Codec: G711a
> FAX Codec Symmetric: yes
> FAX Passthru Method: NSE
> DTMF Tx Mode: Normal
> FAX Process NSE: yes
> FAX Disable ECAN: yes
> FAX Enable T38: yes
> FAX Tone Detect Mode: caller or callee
>
> Note that my Asterisk is also configured to do Fax Tone Detect and ask
> my ATA to switch to T.38 before my ATA ask Asterisk to do so, thereforce
> I don't know if ATA Fax Tone Detection work correctly.
>
>
>> So if I am parsing you correctly, the asterisk server the ATA is
>> registered to is *also* a T.38 gateway, as you have directly connected
>> ISDN?
> Yes, this is also a T.38 gateway, my goal was to use T.38 from my ATA to
> Asterisk over my non-isochronous IP network. T.38 gateway is supported
> since Asterisk >= 10 and you don't need it since you are doing T.38
> passthru, but you need a device somewhere on the path able to detect the
> FAX preamble and ask to switch the data channel from ulaw/alaw to T.38.
>
>
> Sylvain
>

Thanks for the info Sylvain, that is just what I was looking for. Hoping 
to find that a lower model has the same feature set (don't really need 
the FXO port)... I am playing with an SPA122 right now, which seems to 
have the same options available you mention above. I'll load the latest 
firmware and report my experiences for posterity.

Cheers,

j

Jeff

The Grandstream HT-701 (Single Port FXS ATA) is less than $30.00 from 
dstro. 

Working well right now. 

Bryant
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Re: [asterisk-users] T.38 gateway ATA

[Bryant Zimmerman]

 From: "Jeff LaCoursiere" 
Sent: Tuesday, September 25, 2012 11:35 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] T.38 gateway ATA

On 09/25/2012 10:29 AM, Bryant Zimmerman wrote:


 From: "Jeff LaCoursiere" 
Sent: Tuesday, September 25, 2012 11:05 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] T.38 gateway ATA

On 09/25/2012 09:26 AM, Bryant Zimmerman wrote:
 Jeff

Can you please clarify your layout? If you have an asterisk 1.8 (I would 
use 10 for this if possible) server why can't you just take the gateway 
call on that via a sip trunk. If you are coming in from and land line and 
want to do t.38 to the asterisk 1.8 server you would need a FXO t.38 
gateway. Based on your description I am not sure what your sources are and 
what your final desired destination is. Please be specific with your 
response.  We do t.38 all the time and have great success with it but the 
success is in the setup and control of the endpoints (gateways and ATAs)

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003 


 From: "Jeff LaCoursiere" 
Sent: Monday, September 24, 2012 9:20 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: [asterisk-users] T.38 gateway ATA

Hoping for some clarification. I would like to setup a NORMAL (not 
T.38) fax machine on an ATA, and have the ATA be a T.38 gateway to a 
remote asterisk (1.8) server, which is doing T.38 relay (passthru) to a 
provider.

Some amount of googling today seems to imply that most ATAs are just 
T.38 passthru devices, and expect a T.38 capable fax machine, otherwise 
just fallback to ulaw (and mostly fail, in my experience so far).

So does anyone use an ATA that actually does the gateway transcoding to 
a normal fax machine? Or am I barking up the wrong tree?

Thanks!

j

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Hi Bryan,

To follow an inbound fax call, our intended setup would be:

Our upstream sends a T.38 call to our border asterisk (1.8) server, which 
creates another T.38 call to the customer's hosted asterisk (1.8) instance, 
which creates another T.38 call to the ATA, which is over the Internet to 
their location.  The ATA would do the transcoding (is that even the 
appropriate term in this case?) to T.30, to the FXS connected normal fax 
machine.

Its that last bit that I am having trouble confirming is a feature of any 
mainstream ATA.  When I dug into it yesterday it seems that the mainstream 
ATAs will passthru T.38, expecting the connected fax machine to work with 
T.38 natively.  I can't depend on that.  The asterisk servers are all 
remote, and though I could presumably do the gateway on the asterisk server 
I would then have a ulaw fax call over the internet to the ATA, which in my 
experience has not been very reliable.

Of course I will need outbound faxing to follow the reverse path, letting 
the ATA turn the T.30 outbound fax call into T.38, which travels through 
our various asterisk servers to the upstream provider...

So in a nutshell, is there anyone using an ATA as the *gateway* rather than 
passthru?  I feel I am still not being clear... does that help?

Thanks,

j

Jeff

In your call stream you must make sure that all asterisk systems have the 
proper configurations for T.38 and that spandsp is loaded on the systems. 
Each peer entry must have the correct setup and the correct rtp port ranges 
must be set on all asterisk servers. Each server must have the correct 
configurations for their firewalls as well. I would flatten this to start. 
If possible connect the ATA's directly as a registration to your proxy. You 
will likely have better results. We have had the best luck with ATA's from 
audio codes and grandstream currently the HT-701, 702 have been working 
well. You must get the mix right and them keep it flat. Every time you 
allow user to drop an asterisk box between you and the ata you will give 
yourself a greater issues as they must become an asterisk t.38 expert to 
get the deployment to work or you must take it on. Hope this helps.  T.38 
works well when you get the right mix and control it. 

Bryant 

Hmm, I would think I would 

Re: [asterisk-users] is silk included in asterisk 11?

[sean darcy]

On 09/25/2012 11:49 AM, Jonathan Rose wrote:

Jonathan Rose wrote:


Sean Darcy wrote:


I'm building asterisk 11 beta 2. I've been using silk a lot. I
don't
see
silk listed in menuselect as a codec. But I also don't see an
asterisk
11 silk codec on
http://downloads.digium.com/pub/telephony/codec_silk.

Do we use the asterisk 10 codec_silk.so ?

sean


https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats

See the bottom blurb in particular --

* The SILK licensing, like the licensing for Polycom's Siren 7
G.722.1 and Siren 14 G.722.1C codecs, requires that the distribution
of binary codec modules that can be used by Asterisk. To download
the SILK codec module for Asterisk, browse to
http://downloads.digium.com/pub/telephony/codec_silk/unsupported/asterisk-10.0/
and drop the untar'd .so file into /usr/lib/asterisk/modules and
issue an Asterisk restart, or simply load the codec module from the
Asterisk CLI

Actually, it seems that link provided is a little incorrect and the
actual codec modules are available here:
http://downloads.digium.com/pub/telephony/codec_silk/asterisk-10.0/

These modules are actually compiled for Asterisk 10 rather than 11,
but the architecture for codec translators remains largely
unchanged, so I would guess it'll probably work. Probably.


I've been informed that this probably won't actually work due to other
changes that were probably made in resources that codecs tend to rely
on, so take the above with a grain of salt. However, I'm also hearing
that a SILK codec for 11 may be released once 11 is out of beta.



OK. I'll wait on 11 for the silk codec.

Thanks for the quick response.

sean


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Re: [asterisk-users] is silk included in asterisk 11?

[Jonathan Rose]

Jonathan Rose wrote:

> Sean Darcy wrote:
> 
> > I'm building asterisk 11 beta 2. I've been using silk a lot. I
> > don't
> > see
> > silk listed in menuselect as a codec. But I also don't see an
> > asterisk
> > 11 silk codec on
> > http://downloads.digium.com/pub/telephony/codec_silk.
> > 
> > Do we use the asterisk 10 codec_silk.so ?
> > 
> > sean
> 
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats
> 
> See the bottom blurb in particular --
> 
> * The SILK licensing, like the licensing for Polycom's Siren 7
> G.722.1 and Siren 14 G.722.1C codecs, requires that the distribution
> of binary codec modules that can be used by Asterisk. To download
> the SILK codec module for Asterisk, browse to
> http://downloads.digium.com/pub/telephony/codec_silk/unsupported/asterisk-10.0/
> and drop the untar'd .so file into /usr/lib/asterisk/modules and
> issue an Asterisk restart, or simply load the codec module from the
> Asterisk CLI
> 
> Actually, it seems that link provided is a little incorrect and the
> actual codec modules are available here:
> http://downloads.digium.com/pub/telephony/codec_silk/asterisk-10.0/
> 
> These modules are actually compiled for Asterisk 10 rather than 11,
> but the architecture for codec translators remains largely
> unchanged, so I would guess it'll probably work. Probably.

I've been informed that this probably won't actually work due to other
changes that were probably made in resources that codecs tend to rely
on, so take the above with a grain of salt. However, I'm also hearing
that a SILK codec for 11 may be released once 11 is out of beta.

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Re: [asterisk-users] T.38 gateway ATA

[Danny Nicholas]

I would recommend the OBI110.



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Re: [asterisk-users] T.38 gateway ATA

[Jeff LaCoursiere]

On 09/25/2012 10:34 AM, Sylvain Rochet wrote:

Hi Jeff,


On Tue, Sep 25, 2012 at 10:11:27AM -0500, Jeff LaCoursiere wrote:

Aha!  That is what I am looking for.  What firmware rev are you
using on the 3102?

5.1.12 on SPA8000, 5.1.10(GW) on SPA3102, there is probably new releases
from Cisco, at the time I builded the T.38 network those were the latest
releases.

Here is my SPA3102 configuration that I changed or are relevant to FAX/T.38:

Network Jitter Level: very high
Jitter Buffer Adjustment: disable
Call Waiting Serv: no
Three Way Conf Serv: no
Echo Canc Enable: no
Echo Canc Adapt Enable: no
Echo Supp Enable: no
FAX CED Detect Enable: yes
FAX CNG Detect Enable: yes
FAX Passthru Codec: G711a
FAX Codec Symmetric: yes
FAX Passthru Method: NSE
DTMF Tx Mode: Normal
FAX Process NSE: yes
FAX Disable ECAN: yes
FAX Enable T38: yes
FAX Tone Detect Mode: caller or callee

Note that my Asterisk is also configured to do Fax Tone Detect and ask
my ATA to switch to T.38 before my ATA ask Asterisk to do so, thereforce
I don't know if ATA Fax Tone Detection work correctly.



So if I am parsing you correctly, the asterisk server the ATA is
registered to is *also* a T.38 gateway, as you have directly connected
ISDN?

Yes, this is also a T.38 gateway, my goal was to use T.38 from my ATA to
Asterisk over my non-isochronous IP network. T.38 gateway is supported
since Asterisk >= 10 and you don't need it since you are doing T.38
passthru, but you need a device somewhere on the path able to detect the
FAX preamble and ask to switch the data channel from ulaw/alaw to T.38.


Sylvain



Thanks for the info Sylvain, that is just what I was looking for. Hoping 
to find that a lower model has the same feature set (don't really need 
the FXO port)... I am playing with an SPA122 right now, which seems to 
have the same options available you mention above. I'll load the latest 
firmware and report my experiences for posterity.


Cheers,

j



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Re: [asterisk-users] T.38 gateway ATA

[Jeff LaCoursiere]

On 09/25/2012 10:29 AM, Bryant Zimmerman wrote:


*From*: "Jeff LaCoursiere" 
*Sent*: Tuesday, September 25, 2012 11:05 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] T.38 gateway ATA

On 09/25/2012 09:26 AM, Bryant Zimmerman wrote:

Jeff

Can you please clarify your layout? If you have an asterisk 1.8 (I 
would use 10 for this if possible) server why can't you just take the 
gateway call on that via a sip trunk. If you are coming in from and 
land line and want to do t.38 to the asterisk 1.8 server you would 
need a FXO t.38 gateway. Based on your description I am not sure what 
your sources are and what your final desired destination is. Please 
be specific with your response.  We do t.38 all the time and have 
great success with it but the success is in the setup and control of 
the endpoints (gateways and ATAs)


Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003



*From*: "Jeff LaCoursiere" 
*Sent*: Monday, September 24, 2012 9:20 PM
*To*: "Asterisk Users Mailing List - Non-Commercial Discussion" 


*Subject*: [asterisk-users] T.38 gateway ATA


Hoping for some clarification. I would like to setup a NORMAL (not
T.38) fax machine on an ATA, and have the ATA be a T.38 gateway to a
remote asterisk (1.8) server, which is doing T.38 relay (passthru) to a
provider.

Some amount of googling today seems to imply that most ATAs are just
T.38 passthru devices, and expect a T.38 capable fax machine, otherwise
just fallback to ulaw (and mostly fail, in my experience so far).

So does anyone use an ATA that actually does the gateway transcoding to
a normal fax machine? Or am I barking up the wrong tree?

Thanks!

j

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Hi Bryan,

To follow an inbound fax call, our intended setup would be:

Our upstream sends a T.38 call to our border asterisk (1.8) server, 
which creates another T.38 call to the customer's hosted asterisk 
(1.8) instance, which creates another T.38 call to the ATA, which is 
over the Internet to their location.  The ATA would do the transcoding 
(is that even the appropriate term in this case?) to T.30, to the FXS 
connected normal fax machine.


Its that last bit that I am having trouble confirming is a feature of 
any mainstream ATA.  When I dug into it yesterday it seems that the 
mainstream ATAs will passthru T.38, expecting the connected fax 
machine to work with T.38 natively.  I can't depend on that.  The 
asterisk servers are all remote, and though I could presumably do the 
gateway on the asterisk server I would then have a ulaw fax call over 
the internet to the ATA, which in my experience has not been very 
reliable.


Of course I will need outbound faxing to follow the reverse path, 
letting the ATA turn the T.30 outbound fax call into T.38, which 
travels through our various asterisk servers to the upstream provider...


So in a nutshell, is there anyone using an ATA as the *gateway* rather 
than passthru?  I feel I am still not being clear... does that help?


Thanks,

j

Jeff

In your call stream you must make sure that all asterisk systems have 
the proper configurations for T.38 and that spandsp is loaded on the 
systems. Each peer entry must have the correct setup and the correct 
rtp port ranges must be set on all asterisk servers. Each server must 
have the correct configurations for their firewalls as well. I would 
flatten this to start. If possible connect the ATA's directly as a 
registration to your proxy. You will likely have better results. We 
have had the best luck with ATA's from audio codes and grandstream 
currently the HT-701, 702 have been working well. You must get the mix 
right and them keep it flat. Every time you allow user to drop an 
asterisk box between you and the ata you will give yourself a greater 
issues as they must become an asterisk t.38 expert to get the 
deployment to work or you must take it on. Hope this helps.  T.38 
works well when you get the right mix and control it.


Bryant


Hmm, I would think I would need spandsp only if the asterisk server(s) 
would be gateways... is that not the case?  If all I want the asteris

Re: [asterisk-users] T.38 gateway ATA

[Sylvain Rochet]

Hi Jeff,


On Tue, Sep 25, 2012 at 10:11:27AM -0500, Jeff LaCoursiere wrote:
> 
> Aha!  That is what I am looking for.  What firmware rev are you
> using on the 3102?

5.1.12 on SPA8000, 5.1.10(GW) on SPA3102, there is probably new releases 
from Cisco, at the time I builded the T.38 network those were the latest 
releases.

Here is my SPA3102 configuration that I changed or are relevant to FAX/T.38:

Network Jitter Level: very high
Jitter Buffer Adjustment: disable
Call Waiting Serv: no
Three Way Conf Serv: no
Echo Canc Enable: no
Echo Canc Adapt Enable: no
Echo Supp Enable: no
FAX CED Detect Enable: yes
FAX CNG Detect Enable: yes
FAX Passthru Codec: G711a
FAX Codec Symmetric: yes
FAX Passthru Method: NSE
DTMF Tx Mode: Normal
FAX Process NSE: yes
FAX Disable ECAN: yes
FAX Enable T38: yes
FAX Tone Detect Mode: caller or callee

Note that my Asterisk is also configured to do Fax Tone Detect and ask 
my ATA to switch to T.38 before my ATA ask Asterisk to do so, thereforce 
I don't know if ATA Fax Tone Detection work correctly.


> So if I am parsing you correctly, the asterisk server the ATA is 
> registered to is *also* a T.38 gateway, as you have directly connected 
> ISDN?

Yes, this is also a T.38 gateway, my goal was to use T.38 from my ATA to 
Asterisk over my non-isochronous IP network. T.38 gateway is supported 
since Asterisk >= 10 and you don't need it since you are doing T.38 
passthru, but you need a device somewhere on the path able to detect the 
FAX preamble and ask to switch the data channel from ulaw/alaw to T.38.


Sylvain


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Re: [asterisk-users] T.38 gateway ATA

[Bryant Zimmerman]

 

 From: "Jeff LaCoursiere" 
Sent: Tuesday, September 25, 2012 11:05 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] T.38 gateway ATA

On 09/25/2012 09:26 AM, Bryant Zimmerman wrote:
 Jeff

Can you please clarify your layout? If you have an asterisk 1.8 (I would 
use 10 for this if possible) server why can't you just take the gateway 
call on that via a sip trunk. If you are coming in from and land line and 
want to do t.38 to the asterisk 1.8 server you would need a FXO t.38 
gateway. Based on your description I am not sure what your sources are and 
what your final desired destination is. Please be specific with your 
response.  We do t.38 all the time and have great success with it but the 
success is in the setup and control of the endpoints (gateways and ATAs)

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003 


 From: "Jeff LaCoursiere" 
Sent: Monday, September 24, 2012 9:20 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: [asterisk-users] T.38 gateway ATA

Hoping for some clarification. I would like to setup a NORMAL (not 
T.38) fax machine on an ATA, and have the ATA be a T.38 gateway to a 
remote asterisk (1.8) server, which is doing T.38 relay (passthru) to a 
provider.

Some amount of googling today seems to imply that most ATAs are just 
T.38 passthru devices, and expect a T.38 capable fax machine, otherwise 
just fallback to ulaw (and mostly fail, in my experience so far).

So does anyone use an ATA that actually does the gateway transcoding to 
a normal fax machine? Or am I barking up the wrong tree?

Thanks!

j

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Hi Bryan,

To follow an inbound fax call, our intended setup would be:

Our upstream sends a T.38 call to our border asterisk (1.8) server, which 
creates another T.38 call to the customer's hosted asterisk (1.8) instance, 
which creates another T.38 call to the ATA, which is over the Internet to 
their location.  The ATA would do the transcoding (is that even the 
appropriate term in this case?) to T.30, to the FXS connected normal fax 
machine.

Its that last bit that I am having trouble confirming is a feature of any 
mainstream ATA.  When I dug into it yesterday it seems that the mainstream 
ATAs will passthru T.38, expecting the connected fax machine to work with 
T.38 natively.  I can't depend on that.  The asterisk servers are all 
remote, and though I could presumably do the gateway on the asterisk server 
I would then have a ulaw fax call over the internet to the ATA, which in my 
experience has not been very reliable.

Of course I will need outbound faxing to follow the reverse path, letting 
the ATA turn the T.30 outbound fax call into T.38, which travels through 
our various asterisk servers to the upstream provider...

So in a nutshell, is there anyone using an ATA as the *gateway* rather than 
passthru?  I feel I am still not being clear... does that help?

Thanks,

j

Jeff

In your call stream you must make sure that all asterisk systems have the 
proper configurations for T.38 and that spandsp is loaded on the systems. 
Each peer entry must have the correct setup and the correct rtp port ranges 
must be set on all asterisk servers. Each server must have the correct 
configurations for their firewalls as well. I would flatten this to start. 
If possible connect the ATA's directly as a registration to your proxy. You 
will likely have better results. We have had the best luck with ATA's from 
audio codes and grandstream currently the HT-701, 702 have been working 
well. You must get the mix right and them keep it flat. Every time you 
allow user to drop an asterisk box between you and the ata you will give 
yourself a greater issues as they must become an asterisk t.38 expert to 
get the deployment to work or you must take it on. Hope this helps.  T.38 
works well when you get the right mix and control it. 

Bryant
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Re: [asterisk-users] is silk included in asterisk 11?

[Jonathan Rose]

Sean Darcy wrote:

> I'm building asterisk 11 beta 2. I've been using silk a lot. I don't
> see
> silk listed in menuselect as a codec. But I also don't see an
> asterisk
> 11 silk codec on
> http://downloads.digium.com/pub/telephony/codec_silk.
> 
> Do we use the asterisk 10 codec_silk.so ?
> 
> sean

https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats

See the bottom blurb in particular --

* The SILK licensing, like the licensing for Polycom's Siren 7 G.722.1 and 
Siren 14 G.722.1C codecs, requires that the distribution of binary codec 
modules that can be used by Asterisk. To download the SILK codec module for 
Asterisk, browse to 
http://downloads.digium.com/pub/telephony/codec_silk/unsupported/asterisk-10.0/ 
and drop the untar'd .so file into /usr/lib/asterisk/modules and issue an 
Asterisk restart, or simply load the codec module from the Asterisk CLI

Actually, it seems that link provided is a little incorrect and the actual 
codec modules are available here:
http://downloads.digium.com/pub/telephony/codec_silk/asterisk-10.0/

These modules are actually compiled for Asterisk 10 rather than 11, but the 
architecture for codec translators remains largely unchanged, so I would guess 
it'll probably work. Probably.

--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct +1 256 428 6139 

Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] T.38 gateway ATA

[Jeff LaCoursiere]

On 09/25/2012 03:54 AM, Sylvain Rochet wrote:

Hi Jeff,

On Mon, Sep 24, 2012 at 08:18:53PM -0500, Jeff LaCoursiere wrote:

Hoping for some clarification.  I would like to setup a NORMAL (not
T.38) fax machine on an ATA, and have the ATA be a T.38 gateway to a
remote asterisk (1.8) server, which is doing T.38 relay (passthru)
to a provider.

Some amount of googling today seems to imply that most ATAs are just
T.38 passthru devices, and expect a T.38 capable fax machine,
otherwise just fallback to ulaw (and mostly fail, in my experience
so far).

So does anyone use an ATA that actually does the gateway transcoding
to a normal fax machine?  Or am I barking up the wrong tree?

I am using Cisco SPA-8000 and Cisco SPA-3102 as POTS(analog)<>T.38
gateway to an Asterisk 10 doing T.38<>RNIS gateway and it works fine (at
least better than ulaw/alaw over the whole IP network).

However I had to upgrade all ATA firmwares in order to get something
working correctly.

Sylvain



Aha!  That is what I am looking for.  What firmware rev are you using on 
the 3102?  So if I am parsing you correctly, the asterisk server the ATA 
is registered to is *also* a T.38 gateway, as you have directly 
connected ISDN?


Cheers,

j


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Re: [asterisk-users] T.38 gateway ATA

[Jeff LaCoursiere]

On 09/25/2012 09:26 AM, Bryant Zimmerman wrote:

Jeff

Can you please clarify your layout? If you have an asterisk 1.8 (I 
would use 10 for this if possible) server why can't you just take the 
gateway call on that via a sip trunk. If you are coming in from and 
land line and want to do t.38 to the asterisk 1.8 server you would 
need a FXO t.38 gateway. Based on your description I am not sure what 
your sources are and what your final desired destination is. Please be 
specific with your response.  We do t.38 all the time and have great 
success with it but the success is in the setup and control of the 
endpoints (gateways and ATAs)


Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003



*From*: "Jeff LaCoursiere" 
*Sent*: Monday, September 24, 2012 9:20 PM
*To*: "Asterisk Users Mailing List - Non-Commercial Discussion" 


*Subject*: [asterisk-users] T.38 gateway ATA


Hoping for some clarification. I would like to setup a NORMAL (not
T.38) fax machine on an ATA, and have the ATA be a T.38 gateway to a
remote asterisk (1.8) server, which is doing T.38 relay (passthru) to a
provider.

Some amount of googling today seems to imply that most ATAs are just
T.38 passthru devices, and expect a T.38 capable fax machine, otherwise
just fallback to ulaw (and mostly fail, in my experience so far).

So does anyone use an ATA that actually does the gateway transcoding to
a normal fax machine? Or am I barking up the wrong tree?

Thanks!

j

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Hi Bryan,

To follow an inbound fax call, our intended setup would be:

Our upstream sends a T.38 call to our border asterisk (1.8) server, 
which creates another T.38 call to the customer's hosted asterisk (1.8) 
instance, which creates another T.38 call to the ATA, which is over the 
Internet to their location.  The ATA would do the transcoding (is that 
even the appropriate term in this case?) to T.30, to the FXS connected 
normal fax machine.


Its that last bit that I am having trouble confirming is a feature of 
any mainstream ATA.  When I dug into it yesterday it seems that the 
mainstream ATAs will passthru T.38, expecting the connected fax machine 
to work with T.38 natively.  I can't depend on that.  The asterisk 
servers are all remote, and though I could presumably do the gateway on 
the asterisk server I would then have a ulaw fax call over the internet 
to the ATA, which in my experience has not been very reliable.


Of course I will need outbound faxing to follow the reverse path, 
letting the ATA turn the T.30 outbound fax call into T.38, which travels 
through our various asterisk servers to the upstream provider...


So in a nutshell, is there anyone using an ATA as the *gateway* rather 
than passthru?  I feel I am still not being clear... does that help?


Thanks,

j
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[asterisk-users] Question about async channel or macro for monitoring a call

[Sebastian Gutierrez]

Hi,

 Im trying to do this:  


1) Originate a call between an external number and a ivr that do some things in 
background

2) after the originate I bridge the person that dial that extent with the 
external number

I would like to have the ivr in background while the bridge is up for 
monitoring porpoises, but seems to stop processing when the local bridge is done



other possibility could be having a Macro async??   when I make a dial and 
execute a macro I would like to put a while there and control some stuff, but 
until the macro is over I can´t have audio pass between the 2 channels that 
where dialled.


any hint on this??


thanks


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[asterisk-users] is silk included in asterisk 11?

[sean darcy]

I'm building asterisk 11 beta 2. I've been using silk a lot. I don't see 
silk listed in menuselect as a codec. But I also don't see an asterisk 
11 silk codec on http://downloads.digium.com/pub/telephony/codec_silk.


Do we use the asterisk 10 codec_silk.so ?

sean


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Re: [asterisk-users] T.38 gateway ATA

[Bryant Zimmerman]

Jeff

Can you please clarify your layout? If you have an asterisk 1.8 (I would 
use 10 for this if possible) server why can't you just take the gateway 
call on that via a sip trunk. If you are coming in from and land line and 
want to do t.38 to the asterisk 1.8 server you would need a FXO t.38 
gateway. Based on your description I am not sure what your sources are and 
what your final desired destination is. Please be specific with your 
response.  We do t.38 all the time and have great success with it but the 
success is in the setup and control of the endpoints (gateways and ATAs)

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003 


 From: "Jeff LaCoursiere" 
Sent: Monday, September 24, 2012 9:20 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: [asterisk-users] T.38 gateway ATA

Hoping for some clarification. I would like to setup a NORMAL (not 
T.38) fax machine on an ATA, and have the ATA be a T.38 gateway to a 
remote asterisk (1.8) server, which is doing T.38 relay (passthru) to a 
provider.

Some amount of googling today seems to imply that most ATAs are just 
T.38 passthru devices, and expect a T.38 capable fax machine, otherwise 
just fallback to ulaw (and mostly fail, in my experience so far).

So does anyone use an ATA that actually does the gateway transcoding to 
a normal fax machine? Or am I barking up the wrong tree?

Thanks!

j

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[asterisk-users] confbridge video support

[Bryant Zimmerman]

Where does video support for confbridge stand? I need to be able to take in 
multiple video callers and have the active speaking caller displayed to all 
participants. Are we there yet in Asterisk 10 or 11? 

Thanks

Bryant
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Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message

[equis software]

I understand, I'll contact Digium technical support.
Thanks a lot!

On Fri, Sep 21, 2012 at 2:15 PM, Shaun Ruffell  wrote:

> On Fri, Sep 21, 2012 at 01:59:28PM -0300, equis software wrote:
> > Shaun, here is the last test that I made.
> >
> >
> > Time 0:
> > IVR1 (with digiumA) have erros (Master changed to TE2/0/1)
> > IVR2 (with digiumB) OK
> >
> > Time 1: (just swap spams)
> > IVR1 (with digiumB) OK
> > IVR2 (with digiumA) have erros (Master changed to TE2/0/1)
> >
> > I'm seriously thinking that the problem is in the digiumA
> >
> > What you think?
>
> Honestly, it's hard to say without really looking at your complete
> configuration. At this point it might be quickest if you contact
> Digium technical support since they are in a position to collect
> complete information about your setup.
>
> Cheers,
> Shaun
>
> --
> Shaun Ruffell
> Digium, Inc. | Linux Kernel Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org
>
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[asterisk-users] white noise on conference

[Gary Carr]

I am trying to track down a white noise problem we are having in our conference 
rooms. If there are 3 or 4 users in the conference the quality is good. After 
we get more users in the conference we develop a white noise that gets louder 
as more users come online. I have tried both meetme and confbridge. I am 
running 1.8.16.0 compiled from source.

Can anyone provide some insight on where to look or anything to tweak to 
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Re: [asterisk-users] BLF and Call Queues

[Alec Davis]

>> 2012/8/21 Olivier 
>> 
>> 
>>  Hi,
>>  
>> 
>>  What about Queue logs ? How is a picked-up call logged ?
>>  

The member that was being tried is recorded in queue_log. Not the device
that pickedup the call.
This has been queried on issues.asterisk.org with a result of 'will not
fix'.

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
> Lenz Emilitri
> Sent: Tuesday, 25 September 2012 10:48 p.m.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] BLF and Call Queues
> 
> In general I would not use this for a "true" call-center with 
> hundreds of agents, where it is the ACD's responsibility to 
> route calls to agents and there are strict policies on agent 
> behavior, but I'm sure there are a number of cases where this 
> could be useful (eg small call centers, internal service 
> desks, receptionists, etc...).
> 
> Just my two cents,
> l.
> 

Picking up a queue can be done without the use of a BLF hint, a ringing
phone next to you is another hint.
But I agree, in an organisation where the ACD is in total control, the
dialplan code shouldn't exist to allow pickup of a queue.

Queue hints code has now been submitted to asterisk-11-beta2 and trunk.

Alec Davis
 


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Re: [asterisk-users] undefined symbols

[Vieri]

Never mind.

In order to fix those undefined symbols at startup, I needed to preload the 
following modules:

[modules]
autoload=yes
preload => res_ael_share.so
preload => res_speech.so
preload => res_agi.so
preload => res_smdi.so
preload => res_odbc.so
preload => res_fax.so
preload => res_pktccops.so
preload => res_jabber.so

Vieri


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Re: [asterisk-users] BLF and Call Queues

[Lenz Emilitri]

In general I would not use this for a "true" call-center with hundreds of
agents, where it is the ACD's responsibility to route calls to agents and
there are strict policies on agent behavior, but I'm sure there are a
number of cases where this could be useful (eg small call centers, internal
service desks, receptionists, etc...).
Just my two cents,
l.



2012/8/21 Olivier 

> Hi,
>
> What about Queue logs ? How is a picked-up call logged ?
>
> Giving agents the capability to easily pickup a call, without beeing
> logged-in, is a big change with both positive and negative side effects.
> I would be curious to read opinions about that.
>
>
-- 
Loway - home of QueueMetrics - http://queuemetrics.com
Test-drive WombatDialer beta @ http://wombatdialer.com
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Re: [asterisk-users] Help Required IVR

[Lenz Emilitri]

Are you programming the dialplan yourself or are you using a GUI? Also, are
you sure that your greetings message is playable by Asterisk?
l.


2012/9/24 Farooq Hussain 

> Hello everyone,
>
> I stuck in problem I have creating a time based IVR and its working fine.
> If my IVR playing in office hour it would standard IVR and if not they we
> have play a greeting message and place that call to voice mail of
> a extension.
>
> My problem is this I am able to transfer the call on voice mail but how to
> play greeting message first. I am using trixbox 2.2.8 anyone help is this
> regard would great full.
>
> --
> Thanks
>
> Farooq Hussain
>
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Re: [asterisk-users] T.38 gateway ATA

[Sylvain Rochet]

Hi Jeff,

On Mon, Sep 24, 2012 at 08:18:53PM -0500, Jeff LaCoursiere wrote:
> 
> Hoping for some clarification.  I would like to setup a NORMAL (not
> T.38) fax machine on an ATA, and have the ATA be a T.38 gateway to a
> remote asterisk (1.8) server, which is doing T.38 relay (passthru)
> to a provider.
> 
> Some amount of googling today seems to imply that most ATAs are just
> T.38 passthru devices, and expect a T.38 capable fax machine,
> otherwise just fallback to ulaw (and mostly fail, in my experience
> so far).
> 
> So does anyone use an ATA that actually does the gateway transcoding
> to a normal fax machine?  Or am I barking up the wrong tree?

I am using Cisco SPA-8000 and Cisco SPA-3102 as POTS(analog)<>T.38 
gateway to an Asterisk 10 doing T.38<>RNIS gateway and it works fine (at 
least better than ulaw/alaw over the whole IP network).

However I had to upgrade all ATA firmwares in order to get something 
working correctly.

Sylvain


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Re: [asterisk-users] undefined symbols

[A J Stiles]

On Tuesday 25 September 2012, Vieri wrote:
> Hi,
> 
> I compiled Asterisk 10.7.0 with gcc-4.5.3 and at runtime I'm getting these
> warnings:

Usually, this is an indication that you have a missing  (or out-of-date)  
developer's package for some required main package.  Has any version of 
Asterisk ever built successfully on this machine?

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Asterisk as a E1/T1 FXO ?

[A J Stiles]

On Monday 24 September 2012, hbk wrote:
> Hi,
> 
> Does anybody here have experience using Asterisk as an FXO to emulate a
> E1/T1/PRI line for test purpose?
> 
> Gateway or PCI card?

Yes, we use a TE410P to connect to an E1  (ISDN-30)  line,  and also to an 
Eicon Diva card  (which itself is meant to connect to an E1)  in a server 
running Hylafax.  It all works beautifully, with outgoing faxes being routed 
over the phone network and incoming calls on the fax number directed to the 
server.

You just need to specify

signalling = pri_net

and a suitable default context in the appropriate span in your 
chan_dahdi.conf.  (You might want to use NoOp(${EXTEN}) in your first iteration 
of this default context, just to see exactly how the called number is getting 
sent through from the CPE.)

(pri_cpe is Customer Premises Equipment -- i.e., something such as you would 
plug into the box provided by your telco; pri_net is NETwork -- i.e., a box 
such as a telco might provide.)

-- 
AJS

Answers come *after* questions.

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