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Elvin G. Nodalo
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Subject: asterisk-users Digest, Vol 108, Issue 14
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Today's Topics:
1. analog phone digit delay (Justin Killen)
2. Re: analog phone digit delay (jg)
3. Re: analog phone digit delay (Justin Killen)
4. Re: analog phone digit delay (jg)
5. Re: analog phone digit delay (Steve Edwards)
6. Re: PCI Passthrough of T1 cards (Mauricio Tavares)
7. Re: PCI Passthrough of T1 cards (Nick Khamis)
8. Fwd: AQuA Meter ? waveform analysis to get continous MOS
scores for your network (Sevana Oy)
----------------------------------------------------------------------
Message: 1
Date: Mon, 8 Jul 2013 10:14:31 -0700
From: Justin Killen <jkillen at allamericanasphalt.com>
Subject: [asterisk-users] analog phone digit delay
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<55B5D66C43B57F44BC89CB4650FD32F80118FFC2B99F at
MAL.sg1.allamericanasphalt.com>
Content-Type: text/plain; charset="us-ascii"
I have an installation that has analog phones connected via T1 channel banks.
I'm getting complaints from users that they will enter a partial number (eg
91213), then turn away to get the next few digits, and the system will start
dialing before they have a chance to put in the rest of the dialing string. Is
there a way to increase this delay? The only use these 4 dialing patterns:
Internal 3 digit numbers
91 XXX XXX XXXX (for backwards compatibility)
9 XXX XXXX (also for compatibility)
XXX XXXX
I'm using the freepbx distro if that helps. Asterisk 11.2.
Thanks,
-Justin
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Message: 2
Date: Mon, 08 Jul 2013 19:21:10 +0200
From: jg <webaccounts at jgoettgens.de>
Subject: Re: [asterisk-users] analog phone digit delay
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <51DAF506.5070900 at jgoettgens.de>
Content-Type: text/plain; charset=UTF-8; format=flowed
Have a look at the documentation of the channel bank. I guess some kind of
overlap dialing is
enabled, which is typically associated with a timeout value. chan_dahdi.conf
also has entries
like this.
------------------------------
Message: 3
Date: Mon, 8 Jul 2013 10:45:52 -0700
From: Justin Killen <jkillen at allamericanasphalt.com>
Subject: Re: [asterisk-users] analog phone digit delay
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<55B5D66C43B57F44BC89CB4650FD32F80118FFC2B9BB at
MAL.sg1.allamericanasphalt.com>
Content-Type: text/plain; charset="us-ascii"
The channel banks are Adtran TA-624's using ESF/B8ZS. When a handset is
picked up, I can see the offhook in the asterisk console, so it looks that the
channel is immediately connected through the channel bank (not delayed until
after digits are dialed), so it looks that overlap dialing isn't a factor
and that asterisk has complete control.
As for options in chan_dahdi.conf, I simply can't find any that relate to
this problem. I have looked at the page here:
http://www.voip-info.org/wiki/view/chan_dahdi.conf and the closest thing I can
find is 'ringtimeout' which is obviously not what I want. I would
expect to see something like 'dialtimeout' or
'interdigittimeout'.
-Justin
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of jg
Sent: Monday, July 08, 2013 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
Have a look at the documentation of the channel bank. I guess some kind of
overlap dialing is
enabled, which is typically associated with a timeout value. chan_dahdi.conf
also has entries
like this.
--
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Message: 4
Date: Mon, 08 Jul 2013 20:38:20 +0200
From: jg <webaccounts at jgoettgens.de>
Subject: Re: [asterisk-users] analog phone digit delay
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <51DB071C.2030405 at jgoettgens.de>
Content-Type: text/plain; charset=UTF-8; format=flowed
> The channel banks are Adtran TA-624's using ESF/B8ZS. When a handset
is picked up, I can see the offhook in the asterisk console, so it looks that
the channel is immediately connected through the channel bank (not delayed until
after digits are dialed), so it looks that overlap dialing isn't a factor
and that asterisk has complete control.
This also means that you should see the digits as they are dialed. When
something times out you
should also see a message why there was a timeout.
I am using ISDN for PSTN connections and where I live there must be some kind of
overlap dialing
enabled, otherwise P2P configurations don't work. With current DAHDI drivers
I no longer need
special settings to make things work (Wanpipe/Woomera was different), so I guess
overlap dialing
is enabled. Some SIP phones distinguish between "Overlap Dialing" and
"Automatic Dialing", so
your channel bank might also have something like an Automatic Dialing option
with some timing
value.> As for options in chan_dahdi.conf, I simply can't find any that relate
to this problem. I have looked at the page here:
http://www.voip-info.org/wiki/view/chan_dahdi.conf and the closest thing I can
find is 'ringtimeout' which is obviously not what I want. I would
expect to see something like 'dialtimeout' or
'interdigittimeout'.
There is an "overlap" option in configs/chan_dahdi.conf.sample.
I am currently assembling an Asterisk box that has 48+2 analog channels (+ SIP +
ISDN). If your
problem doesn't go away I could tell next week what my system is doing.
jg
------------------------------
Message: 5
Date: Mon, 8 Jul 2013 11:55:21 -0700 (PDT)
From: Steve Edwards <asterisk.org at sedwards.com>
Subject: Re: [asterisk-users] analog phone digit delay
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <alpine.DEB.2.02.1307081154360.13329 at ws>
Content-Type: text/plain; charset="iso-8859-7";
Format="flowed"
On Mon, 8 Jul 2013, Justin Killen wrote:
> I have an installation that has analog phones connected via T1 channel
> banks. ?I?m getting complaints from users that they will enter a partial
> number (eg 91213), then turn away to get the next few digits, and the
> system will start dialing before they have a chance to put in the rest
> of the dialing string. ?Is there a way to increase this delay?? The only
> use these 4 dialing patterns:
Will 'show function TIMEOUT' help?
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
------------------------------
Message: 6
Date: Mon, 8 Jul 2013 17:07:53 -0400
From: Mauricio Tavares <raubvogel at gmail.com>
Subject: Re: [asterisk-users] PCI Passthrough of T1 cards
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<CAHEKYV60YsVa76GJ+TX2ToVA1w=AV2gi+=F4GHdz8cAnmd25XA at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
On Wed, Jun 19, 2013 at 11:52 AM, Nick Khamis <symack at gmail.com>
wrote:> Hello James,
>
> Thank you so much for your response. I should have chose my words
> carefully. PCI pass-through in terms of virtualization of devices and
> it's draw back are well know. I was leaning more towards near host
> performance virtualization using SR-IOV.
>
I know I am late in the show, but what are the drawbacks as far
as using Asterisk is concerned?
> This moves emphasis back to the production drivers of the interface
> card using virtual functions etc., and can provide near host
> performance. Rephrasing my question, are any of the T1 pci
> manufactures providing support for virtualization using SR-IOV and
> virutal functions?
>
> Kind Regards,
>
> Nick
>
> --
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------------------------------
Message: 7
Date: Mon, 8 Jul 2013 19:11:36 -0400
From: Nick Khamis <symack at gmail.com>
Subject: Re: [asterisk-users] PCI Passthrough of T1 cards
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<CAGWRaZY0Q_jAGjLPGzqF6X=utMcTKRrwBBH5ZRLFyJo=UZ_sCw at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Asterisk does fine in a virtual instance. The key is finding hardware that
would
support more than just virtualization (i.e., SR-IOV).... Not sure if such a
card
exist.
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Message: 8
Date: Tue, 9 Jul 2013 19:34:34 +0400
From: Sevana Oy <sales at sevana.fi>
Subject: [asterisk-users] Fwd: AQuA Meter ? waveform analysis to get
continous MOS scores for your network
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<CAMyj0v=ez4dkj9hPi205QQ9ySQs-wbV_pHq8b4uJRVeDhJfbMA at mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"
Hi,
Although this is a repost from Asterisk biz, we would like to ask if
somebody may help us to develop a native Asterisk module using AQuA
technology for voice quality monitoring using the same web service AQuA
Meter is using.
Thanks,
Sevana Finland/Estonia
---------- Forwarded message ----------
From: Sevana Oy <sales at sevana.fi>
Date: Mon, Jun 17, 2013 at 7:30 PM
Subject: AQuA Meter ? waveform analysis to get continous MOS scores for
your network
To: asterisk-biz at lists.digium.com
[image: AQuA
Meter]<http://blog.sevana.fi/wp-content/uploads/2013/03/screenshot.png>
Hi,
We would like to offer you to learn about our new application that performs
scheduled voice test calls to a predefined
echo server and then uses our AQuA web service to evaluate the call quality.
We developed it because several VoIP service providers have inquired us for
a possibility to make test calls from local machines within
their customers? network.
A typical example is when you provide VoIP communications to a company that
rents its premises (including an Internet connection) in a
business center. In this case it is quite important to monitor voice call
quality from different computers in the office space to the
service provider?s server.
This is a cross platform (Windows, Linux, MAC) Java application and uses
our latest developments in waveform analysis to evaluate voice call
quality: http://www.sevana.fi/aquameter.zip
The setup is simple: our application calls the echo server (apparently
provided by the VoIP service provider), plays a reference audio and records
the playback from the echo server and can thus provide overall (both ways)
call quality analysis.
We are very interested to receive your feedback and feature wishlist. The
application is free.
Best Regards,
Sevana Oy/O?
Finland/Estonia
http://blog.sevana.fi/aqua-meter-waveform-analysis-to-get-continous-mos-scores-for-your-network/
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