Re: [asterisk-users] Upgrading to 1.4

[Rob Hillis]

As is just about always the case, posting twice to the list within three 
hours is not only unlikely to get a faster response, I would in fact 
imagine it would /reduce/ your chances of getting a response at all.


lotusscript wrote:

A good while back when installing 1.2 there were major issues with UK
callerid.  Asterisk 1.2 didn't recognise the callerid correctly because
of the way BT sent the information.  Sometimes before the first ring or
just after.  After applying a third party patch we got it to work.  We
were afraid to touch it after that  :-)  Has this problem now gone away
with 1.4?
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Re: [asterisk-users] Quality problems with ISDN PRI

[Rob Hillis]

Steve Totaro wrote:

On Fri, Apr 25, 2008 at 11:10 AM, Doug Lytle <[EMAIL PROTECTED]> wrote:
  

Steve Totaro wrote:
 > That is interesting.  I have an intel C2D and I can only see two
 > procs, not four, is that normal?  Are you sure what you are saying is
 >

 I believe Intel removed HyperThreading after it moved over to dual cores.

 Doug

 --

 Ben Franklin quote:

 "Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."




My dual proc, dual core AMD boxen show as four procs.  I guess the AMD
architecture uses Hypertheading (or whatever the equivalent is for
AMD, I assume Intel owns the rights to the name Hyperthreading).

  
Two dual core processors /would/ should four processors - each processor 
has two virtual processors for a total of four.


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Re: [asterisk-users] Graphing Jitter Packet loss and Latency for SIP Calls

[Grey Man]

On Fri, Apr 25, 2008 at 10:55 PM, Vikas <[EMAIL PROTECTED]> wrote:
> Requirement:  Monitor the QOS for the SIP phones connecting to the voip 
> server.
>
>  Ideal solution: Browder based reporting software that I can install on
>  the asterisk server (I use freepbx) and when I connect to this
>  reporting engine it gives me the Jitter loss, packet loss and latency
>  for each of the calls that the extensions connecting to this asterisk
>  server make and receive.
>
>  Network design:
>  A. The sip endpoints: 6 polycom 650 phones in India connecting to an
>  VOIP server.
>  B. Network between the SIP endpoints and VOIP server: The Indian
>  office has 5 different ISPs giving the internet connection. Each ISP
>  has a different packet loss latnecy and Jitter and these change over
>  time. So I want a way to be able to select the best ISP on a given
>  day.
>  C. VOIP server: hosted at he.net datacenter and acts as the gateway
>  between the sip endpoints and the PSTN gateway.
>  D. PSTN gateway: Broadvoice for outgoing calls and exgn.net for
>  incoming calls on the 800 number
>
>  Things I have looked at:
>  1. Wireshark -> I did not find a good reporting engine which I can
>  automate to collect data and then graph it.
>  2. Endian 2.2
>  3. IPCop
>
>  I would really appreciate any insights on how to monitor the QOS.
>
>  Thanks for your time,
>

I doubt you'll find a good solution for free (if you do I for one
would love to hear about it).

My company looked at monitoring QoS about 18 months ago. We ended up
evaluating on of the Hammer products from Empirix. At the time of the
eval the product couldn't do much in real-time with QoS stats such as
jitter and you could only collate general statistical information at
the end of the call. Subsequent to our eval the product was enhanced
to provide better real-time reporting and in the end I don't think
there was too much it couldn't do. The drawback then came down to
price which is hefty. I suspect the Empirix range of products won't
suit your needs due to price but they could be worth checking out to
give you a guide as to how QoS monitoring could be done.

As an aside I beleive Digium are using the Empirix load tools in some
kind of partnership arrangement to stress test Asterisk these days.

Regards,

Greyman.

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Re: [asterisk-users] choopy audio when both side talk at the same time

[Matt Watson]

You might want to begin with tuning your rxgain and txgain settings... there 
are a few methods for doing this on the internet, unfortunatly nobody can give 
you exactly values to use for tx/rxgain as they will be not only specific to 
your install, but specific to every single analog line you have... you can 
probably get away with setting it once for all of your lines, but i'd recomend 
setting it for every one.

http://lists.digium.com/pipermail/asterisk-users/2004-November/064312.html

That is the best guide i;ve seen to doing it, but there may be better ones.

After you get rxgain and txgain tuned for every one of your lines, you'll 
probably notice a dramatic decrease in echo right away, but you can also tune 
your echocancel= and echotraining= after that.

You can set these values on a per-channel basis by doing it like:

rxgain=6.3
txgain=-1.0
channel => 9

rxgain=7.595
txgain=-2.0
channel => 10

rxgain=6.3
txgain=-1.136
channel => 11

etc.

First step IMHO is getting your rx/txgain set properly... don't underestimate 
how important those values are... I learned that the hard way.

--
Matt

From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Ruben Zamora [EMAIL 
PROTECTED]
Sent: Friday, April 25, 2008 7:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] choopy audio when both side talk at the same time

Hi

I have a server with the last version of asterisk branches, zaptel
branches, 2 Digium Card with TDM800P
16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10
Grandstream GXP2000.

zapata.conf
echocancel=64
rxgain=0
txgain=0

when i place a call o receive a call, I finish a sentence i hear a
,AND  when the both side talks at
the same time i have choppy audio.

Any help i appreciate.

Thanks Ruben

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[asterisk-users] choopy audio when both side talk at the same time

[Ruben Zamora]

Hi

I have a server with the last version of asterisk branches, zaptel 
branches, 2 Digium Card with TDM800P
16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10 
Grandstream GXP2000. 

zapata.conf
echocancel=64
rxgain=0
txgain=0

when i place a call o receive a call, I finish a sentence i hear a 
,AND  when the both side talks at
the same time i have choppy audio.

Any help i appreciate.

Thanks Ruben

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Re: [asterisk-users] Graphing Jitter Packet loss and Latency for SIP Calls

[Andrew Matthews]

On Fri, Apr 25, 2008 at 2:55 PM, Vikas <[EMAIL PROTECTED]> wrote:

>  B. Network between the SIP endpoints and VOIP server: The Indian
>  office has 5 different ISPs giving the internet connection. Each ISP
>  has a different packet loss latnecy and Jitter and these change over
>  time. So I want a way to be able to select the best ISP on a given
>  day.

I would recommend smokeping, it won't monitor the quality of the call,
but it will give you a good idea of how the circuit performs.

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Re: [asterisk-users] Asterisk using 100% of CPU

[Anthony Francis]

Plus that originate is going to call the sip device, and upon answer 
connect it to extension 0 in the internal context, is that what you wanted?

Tilghman Lesher wrote:
> On Friday 25 April 2008 15:23:05 Chris Elliott wrote:
>   
>> If I reverse the situation it gets a little better.  Asterisk doesn't
>> use 100% of the CPU, but until SIP/exten-20 answers, the manager
>> interface doesn't respond.  So I can't hangup the line using the manager
>> API if SIP/exten-20 doesn't answer.  SIP/exten-20 is a SPA3102 FXS.
>> Here is that example:
>>
>> Action: Originate
>> Channel: SIP/exten-20
>> Context: internal
>> Extension: 0
>> Priority: 1
>> 
>
> The reason Manager doesn't respond is that it's waiting for a result code to
> give you.  If you don't care, use the "Async: yes" option to the Originate
> action to get AMI to continue past that point.
>
>   

-- 
Thank you and have any kind of day you want,

Anthony Francis
Rockynet VOIP



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[asterisk-users] Graphing Jitter Packet loss and Latency for SIP Calls

[Vikas]

Requirement:  Monitor the QOS for the SIP phones connecting to the voip server.

Ideal solution: Browder based reporting software that I can install on
the asterisk server (I use freepbx) and when I connect to this
reporting engine it gives me the Jitter loss, packet loss and latency
for each of the calls that the extensions connecting to this asterisk
server make and receive.

Network design:
A. The sip endpoints: 6 polycom 650 phones in India connecting to an
VOIP server.
B. Network between the SIP endpoints and VOIP server: The Indian
office has 5 different ISPs giving the internet connection. Each ISP
has a different packet loss latnecy and Jitter and these change over
time. So I want a way to be able to select the best ISP on a given
day.
C. VOIP server: hosted at he.net datacenter and acts as the gateway
between the sip endpoints and the PSTN gateway.
D. PSTN gateway: Broadvoice for outgoing calls and exgn.net for
incoming calls on the 800 number

Things I have looked at:
1. Wireshark -> I did not find a good reporting engine which I can
automate to collect data and then graph it.
2. Endian 2.2
3. IPCop

I would really appreciate any insights on how to monitor the QOS.

Thanks for your time,

Sysadmin
http://www.debtconsolidationcare.com
Internets First get out of debt community

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Re: [asterisk-users] Asterisk using 100% of CPU

[Tilghman Lesher]

On Friday 25 April 2008 15:23:05 Chris Elliott wrote:
> If I reverse the situation it gets a little better.  Asterisk doesn't
> use 100% of the CPU, but until SIP/exten-20 answers, the manager
> interface doesn't respond.  So I can't hangup the line using the manager
> API if SIP/exten-20 doesn't answer.  SIP/exten-20 is a SPA3102 FXS.
> Here is that example:
>
> Action: Originate
> Channel: SIP/exten-20
> Context: internal
> Extension: 0
> Priority: 1

The reason Manager doesn't respond is that it's waiting for a result code to
give you.  If you don't care, use the "Async: yes" option to the Originate
action to get AMI to continue past that point.

-- 
Tilghman

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[asterisk-users] Asterisk using 100% of CPU

[Chris Elliott]

When I initiate a call from the console (Console/dsp) to a local SIP 
extension, asterisk uses up 100% of the CPU until the extension 
answers.  It happens when using .call files or the manager API.  My 
examples are for the manager API, but .call files perform the same way.  
Here is the 100% CPU example:

Action: Originate
Channel: Console/dsp
Context: internal
Extension: 20
Priority: 1

If I reverse the situation it gets a little better.  Asterisk doesn't 
use 100% of the CPU, but until SIP/exten-20 answers, the manager 
interface doesn't respond.  So I can't hangup the line using the manager 
API if SIP/exten-20 doesn't answer.  SIP/exten-20 is a SPA3102 FXS.  
Here is that example:

Action: Originate
Channel: SIP/exten-20
Context: internal
Extension: 0
Priority: 1


If I initiate a call from Console/dsp to the FXO port of a SPA3102, that 
works fine.  Example:

Action: Originate
Channel: Console/dsp
Context: internal
Extension: 98005551212
Priority: 1

I am using asterisk is version 1.2.24.  Basically I would like to be 
able to use the manager API to externally initiate calls from the 
console and hang them up.  Any ideas here?

Here are the relevant parts of extensions.conf:

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]
CONSOLE=Console/dsp
TRUNKMSD=1

[macro-stdexten]
exten => s,1,Dial(${ARG1})
exten => s,n,Playtones(busy)
exten => s,n,Busy

[internal]
include => trunkout
exten => 0,1,Dial(${CONSOLE})
exten => 20,1,Macro(stdexten,SIP/exten-20)
exten => 21,1,Macro(stdexten,SIP/exten-21)

[trunkout]
exten => _9.,1,Dial(SIP/pstn-01/${EXTEN:1})
exten => _9.,n,Dial(SIP/pstn-02/${EXTEN:1})
exten => _9.,n,Playtones(congestion)
exten => _9.,n,Congestion
exten => _9.,n,Hangup


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Re: [asterisk-users] Quality problems with ISDN PRI

[Matt Florell]

On 4/25/08, Jared Smith <[EMAIL PROTECTED]> wrote:
> On Fri, 2008-04-25 at 18:48 +, Arthur wrote:
>  > I still hope someone would enlighten us by his experience in doing
>  > call recordings without  recording to RAM Drive.
>
>
> I can't speak for Steve's solution (as I'm not sure exactly what he's
>  doing) but I could take a stab in the dark and guess that he's capturing
>  the audio at the network layer (on a completely different box than
>  Asterisk is running on) and recording it from there.  But that's just a
>  guess...

To address several points:

OrecX (http://www.orecx.com/) can do call recording outside of the
Asterisk core using several different methods depending on your needs
and channeltypes. In fact even with Sangoma TDM cards you can capture
audio at the kernel level and send the audio as RTP streams very
efficiently(3% CPU load for 92 channels) to an OrecX server on your
network. It must be mentioned that setting up Orecx with retrieval
might be a little complex for some Asterisk users, especially if you
are recording a large amount of calls, or are recording on more than
one Asterisk server, and if you choose this route you would do well to
hire an experienced consultant(or contact Oreca directly) to do the
install for you.

As far as Asterisk-based recording, writing to a RAM drive(or tmpfs)
is about your only option if you are planning on doing more than 50
concurrent recordings, if you are using Asterisk it is a viable and
tested solution. I have several client systems that are recording well
over 50 calls concurrently on a daily basis this way.

If you will be recording directly to hard drives with any frequency or
volume I would strongly recommend NOT using standard IDE or SATA hard
drives, they burn up and fast. Use a caching SCSI drive controller
with some high quality SCSI drives and you can record to those drives
for years even at 40 concurrent channels recording all day every day.

Hope that helps,

MATT---

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Re: [asterisk-users] Quality problems with ISDN PRI

[Arthur]

>
> he's capturing the audio at the network layer

i'd better stay with my 3Gigs RAM drive
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Re: [asterisk-users] ATA FXO / FXS - can forward to sip ?

[Arthur]

i can only think of an asterisk box & the right dialplan.
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Re: [asterisk-users] Quality problems with ISDN PRI

[Jared Smith]

On Fri, 2008-04-25 at 18:48 +, Arthur wrote:
> I still hope someone would enlighten us by his experience in doing
> call recordings without  recording to RAM Drive.

I can't speak for Steve's solution (as I'm not sure exactly what he's
doing) but I could take a stab in the dark and guess that he's capturing
the audio at the network layer (on a completely different box than
Asterisk is running on) and recording it from there.  But that's just a
guess...

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Quality problems with ISDN PRI

[Arthur]

I still hope someone would enlighten us by his experience in doing call
recordings without  recording to RAM Drive.
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[asterisk-users] E-mail date is wrong

[Alejandro Cabrera Obed]

Dear all, I'm using Asterisk 1.4.13 with voicemail feature. When anybody
receive a voice message, he/she receives a mail with the audio attachment.

After that I dial the voicemail number and I hear the envelope message
that is correct (America/Argentina/Buenos_Aires) which is GMT-3, but
when I see the message date header it is wrong because the date
correspond to GMT and not GMT-3.

Where can I set the date/time in order to put Asterisk to send message
with the correct time ???

(The Linux server date is correctly set).

Thanks a lot

Alejandro

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Re: [asterisk-users] Quality problems with ISDN PRI

[Doug Lytle]

Steve Totaro wrote:
> My dual proc, dual core AMD boxen show as four procs.  I guess the AMD
> architecture uses Hypertheading (or whatever the equivalent is for
> AMD, I assume Intel owns the rights to the name Hyperthreading).
>   

Not that I'm aware of.

But I did find this article from back in 2002:

http://www.geek.com/amd-to-do-hyperthreading/


Doug


-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> 
> RTC is available (and used) as of kernel 2.6.15 . The thing that has
> changed in 2.6.13 is that the default of HZ became 250 (but still
> tunable). So unless you build your own kernel, without using RTC you
> would not really get a steady rate of 1000 interrupts per second.

Well, I'm not familiar with the later 2.6 kernels (most of my systems
are at 2.6.12 (FC3) or 2.6.9 (RHEL4 clones)).

However, the USE_RTC code was my creation, so I'm very familiar with the
issues as they were at the time.  See http://bugs.digium.com/view.php?id=4301

The issue was that the original 2.6 version of ztdummy ran off the 1000Hz
kernel jiffy counter. This had a tendency to miss ticks. Having successfully
used the zaprtc module in 2.4, I re-implemented it for 2.6 using the rtc
hooks that the 2.6 kernel provided. It sets the 146818 RTC chip to generate
1024Hz interrupts (it can't do 1000Hz), and then skips 3 every 128, evenly
spaced. This was a huge improvement over the jiffy counter.

Unfortunately, when the patch was applied to CVS, someone screwed up and
missed out ztdummy.h, only doing ztdummy.c. This broke compilation and
caused BKW to throw a fit. The knee-jerk reaction was to slap an #if 0
around the #define USE_RTC, rather than understand the cause of the problem.
Once ztdummy.h was patched correctly, the #if 0 should have been removed,
but it never was, so most people continued to build it with the inferior
jiffy clock.

When kernel 2.6.13 came along, the jiffy clock no longer defaulted to 1000Hz,
so USE_RTC was made the default for those versions. I will never understand
why it was never just enabled for *all* 2.6 kernels at that time, like it
should have been in the first place.

The only dependency it has is that the kernel must have been built with
CONFIG_RTC and not CONFIG_GENRTC.

> And then again, on kernels >= 2.6.22 you have hi-resolution timers which
> generally work better.

I have yet to experience these, but it sounds promising.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] Quality problems with ISDN PRI

[Steve Totaro]

On Fri, Apr 25, 2008 at 11:10 AM, Doug Lytle <[EMAIL PROTECTED]> wrote:
> Steve Totaro wrote:
>  > That is interesting.  I have an intel C2D and I can only see two
>  > procs, not four, is that normal?  Are you sure what you are saying is
>  >
>
>  I believe Intel removed HyperThreading after it moved over to dual cores.
>
>  Doug
>
>  --
>
>  Ben Franklin quote:
>
>  "Those who would give up Essential Liberty to purchase a little Temporary 
> Safety, deserve neither Liberty nor Safety."
>

My dual proc, dual core AMD boxen show as four procs.  I guess the AMD
architecture uses Hypertheading (or whatever the equivalent is for
AMD, I assume Intel owns the rights to the name Hyperthreading).

Thanks,
Steve Totaro

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Re: [asterisk-users] Quality problems with ISDN PRI

[Steve Totaro]

On Fri, Apr 25, 2008 at 12:22 PM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> On Thu, Apr 24, 2008 at 4:45 PM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
>  > On Wed, Apr 23, 2008 at 02:14:27PM -0400, Steve Totaro wrote:
>  >  > There are much better solutions than doing a RAM drive.  While it may
>  >  > be stable (not in my experience, I advise using different servers for
>  >  > different tasks (with redundancy obviously).  A phone switch should be
>  >  > just that, a recording server should also be just that (in demanding
>  >  > environments).
>  >
>  >  That would be fine, if Asterisk was capable of buffering recording
>  >  writes, but I'm told it's not; the I/O involved in getting that
>  >  recording data off the box in real time is probably worse than that of
>  >  putting it onto disk -- disks are usually higher bandwidth channels
>  >  than network adapters.
>  >
>  >  For permanent storage, certainly, the recordings should be moved to
>  >  another box, and that's how we do it here.
>  >
>  >  Cheers,
>  >  -- jr '44 byte chunks. Is someone an ATM fan?' a
>  >  --
>  >  Jay R. Ashworth   Baylink  [EMAIL 
> PROTECTED]
>  >  Designer The Things I Think   RFC 
> 2100
>  >  Ashworth & Associates http://baylink.pitas.com 
> '87 e24
>  >  St Petersburg FL USA  http://photo.imageinc.us +1 727 647 
> 1274
>  >
>  >  Those who cast the vote decide nothing.
>  >  Those who count the vote decide everything.
>  >-- (Joseph Stalin)
>  >
>
>  Well in the real world, your hypothesis has been proven wrong.
>
>  Thanks,
>  Steve Totaro
>

To correct my previous statement.  Who said anything about Asterisk
doing the recording function at all?  Certainly not me.  My recording
servers sometimes run Windows but usually Linux just to keep
everything in the "Phone Rack" on the same OS.

Thanks,
Steve Totaro

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Re: [asterisk-users] Quality problems with ISDN PRI

[Steve Totaro]

On Thu, Apr 24, 2008 at 4:45 PM, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
> On Wed, Apr 23, 2008 at 02:14:27PM -0400, Steve Totaro wrote:
>  > There are much better solutions than doing a RAM drive.  While it may
>  > be stable (not in my experience, I advise using different servers for
>  > different tasks (with redundancy obviously).  A phone switch should be
>  > just that, a recording server should also be just that (in demanding
>  > environments).
>
>  That would be fine, if Asterisk was capable of buffering recording
>  writes, but I'm told it's not; the I/O involved in getting that
>  recording data off the box in real time is probably worse than that of
>  putting it onto disk -- disks are usually higher bandwidth channels
>  than network adapters.
>
>  For permanent storage, certainly, the recordings should be moved to
>  another box, and that's how we do it here.
>
>  Cheers,
>  -- jr '44 byte chunks. Is someone an ATM fan?' a
>  --
>  Jay R. Ashworth   Baylink  [EMAIL 
> PROTECTED]
>  Designer The Things I Think   RFC 
> 2100
>  Ashworth & Associates http://baylink.pitas.com '87 
> e24
>  St Petersburg FL USA  http://photo.imageinc.us +1 727 647 
> 1274
>
>  Those who cast the vote decide nothing.
>  Those who count the vote decide everything.
>-- (Joseph Stalin)
>

Well in the real world, your hypothesis has been proven wrong.

Thanks,
Steve Totaro

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Re: [asterisk-users] Forking in Dialplan

[Craig Guy]

On 4/25/08, Tobias Ahlander <[EMAIL PROTECTED]> wrote:
> >Date: Thu, 24 Apr 2008 06:54:27 -0700 (PDT)
> >From: Steve Edwards <[EMAIL PROTECTED]>
> >Subject: Re: [asterisk-users] Forking in Dialplan
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
>  >   
> >Message-ID: <[EMAIL PROTECTED]>
> >Content-Type: text/plain; charset="x-unknown"
>
> >> - "Tobias Ahlander" <[EMAIL PROTECTED]> escreveu:
>
> >>> Is it possible to somehow fork in the dialplan? Say a call comes in.
> >>> Then I want to wait 30 seconds and then write in a database, but at the
> >>> same time while I wait I want to go on with other commands too.
>
> >On Thu, 24 Apr 2008, Vin??cius Fontes wrote:
>
> >> You can call an AGI script that will call another script. That last one
> >> would wait 10 seconds and write in the database. The following example
> >> works for me:
> >>
> >> /var/lib/asterisk/agi-bin/agi-test.agi:
> >>
> >> #!/bin/bash
> >> nohup /root/helloworld.sh 1>/dev/null 2>/dev/null &
> >> exit 0
> >>
> >> /root/helloworld.sh:
> >>
> >> #!/bin/bash
> >> sleep 10
> >> echo "Hello world!" >> /root/helloworld.txt
> >> exit 0
>
> >Why do you need the first AGI? Would:
>
> > exten = _x.,n,system(nohup /root/helloworld.sh 1>/dev/null 2>&1 &)
>
> >suit your needs?
>
> >Thanks in advance,
> >
> >Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
> >Newline Fax: +1-760-731-3000
>
>
> Thank you Steve, this seems to work just as I want it to. Now I just have to
> figure out how to send variables to a system call, but I think I have that
> covered somewhere :)
>
> Best regards,
> Tobias
>

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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Benjamin Jacob]

Benjamin Jacob <[EMAIL PROTECTED]> wrote: 

Tony Mountifield <[EMAIL PROTECTED]> wrote: In article <[EMAIL PROTECTED]>,
Benjamin Jacob  wrote:
> 
> One on my clients' machine had Asterisk 1.4.4. installed. The complained of 
> choppy Playback
> of gsm files.
> So scouring the internet gave me the solution of installing ztdummy and 
> loading it as a module.
> Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and 
> re-installed. Sill
> no effect.
> 
> Do I have to specify any parameter in the Asterisk compilation to look at 
> ztdummy/rtc? As
> far as I remember (am coming back to Asterisk after quite some time now), you 
> don't really
> need to set anything over there for any zaptel specific compilation?
> 
> And yes, all the files are  gsm files and the codec used for the calls is 
> ulaw.
> 
> I even tried converting those gsm files to wav using sox and then playing 
> them, but the
> behaviour is the same.
> 
> Any ideas anyone.. something I am missing ??

Firstly, check whether Asterisk has chan_zap loaded and access to zaptel:

*CLI> zap show channels
   Chan Extension  Context Language   MusicOnHold 
 pseudodefault
*CLI>

If you don't get pseudo shown, then you are not getting the benefit of
ztdummy.

However, the probably main cause of choppy sound is poor timing from the
SIP client (I'm assuming SIP), because Asterisk by default uses the incoming
stream to generate timing for the outbound stream.

There are two main things to try:

1. Make sure that the SIP clients are NOT using silence suppression (may
be referred to as VAD, bandwidth saving, or something similar).

2. If  ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable
the line "internal_timing=yes". That should make it play out based on
internal zaptel timing instead of timing off the incoming stream, I think.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

Thanks Tony for the response.
zap show channels shows that things are fine, as you said :
*CLI> zap show channels
   Chan Extension  Context Language   MOH Interpret   
 pseudodefaultdefault

Tried setting internal_timing to yes as well. Still  no difference.

Also,  I don't think my SIP gateway uses Silence suppression, because the same 
SIP gateway connections work fine with another Asterisk server.

This is getting seriously irritating now!!! Have tried all the tricks and tips 
I've been finding on the net.

Yeah, btw, even Meetme playback is choppy. So, I think its somehow related to 
timing. But I am not the expert. 

- Ben.
 

-
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   http://lists.digium.com/mailman/listinfo/asterisk-usersBtw, I am on CentOS 
5, with uname showing as:
Linux mserver.org 2.6.18-53.1.13.el5 #1 SMP Tue Feb 12 13:01:45 EST 2008 i686 
i686 i386 GNU/Linux

And it is not a multiprocessor machine. Will the SMP option affect the working 
in any way?

- Ben.

   
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Re: [asterisk-users] Digium B410P or Sangoma A502D?

[Lee Jenkins]

Andres wrote:
> We have tested both and they work fine.  The Sangoma is much easier to 
> install as it does not depend on any other driver, you just run 
> 'setup-sangoma' and follow the instructions.  You don't have to fiddle 
> with the linux kernel or  zaptel or chan_misdn.  It just works.  Plus 
> its more modular.  You can chose 2/4/6 ports to buy and if you need more 
> just add remoras up to 24 ports.  The Digium card is fixed to 4 ports, 
> period.
> 
> Having said that, make sure you stick with the version that has hardware 
> echo cancel and not even try the other one.  We made the mistake of 
> buying the first time without echo cancel expecting to test the 
> 'software echo cancel'.  But there is no such thing as 'software echo 
> cancel' on this card.  I do not even understand why Sangoma would make a 
> version without the hardware echo cancel.  You get some degree of echo 
> on practically every call.
> 
> Andres.

I have a couple of installs using the A200 analog card with FXO modules and the 
Octware echo cancel software works like a charm.  These are 2-4 POTS line 
installs.


-- 

Warm Regards,

Lee


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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Tzafrir Cohen]

On Fri, Apr 25, 2008 at 03:02:14PM +, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
> Benjamin Jacob <[EMAIL PROTECTED]> wrote:
> > 
> > Also,  I don't think my SIP gateway uses Silence suppression, because the 
> > same SIP gateway
> > connections work fine with another Asterisk server.
> 
> OK, I think you need to home in on the differences between the server(s)
> that work fine and the one that doesn't.
> 
> What version of kernel is it running? If it less than 2.6.13, make sure
> you change "#if 0" to "#if 1" in ztdummy.c so that USE_RTC still gets
> enabled.

RTC is available (and used) as of kernel 2.6.15 . The thing that has
changed in 2.6.13 is that the default of HZ became 250 (but still
tunable). So unless you build your own kernel, without using RTC you
would not really get a steady rate of 1000 interrupts per second.

And then again, on kernels >= 2.6.22 you have hi-resolution timers which
generally work better.

> 
> Try "watch -dn 1 cat /proc/interrupts" and check that the RTC interrupts
> are going up by 1024 per second. This is with ztdummy running.

And if using something other than RTC: 1000 interrupts per second.
Anyway, "close to 1000" is easy to spot there.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Quality problems with ISDN PRI

[Doug Lytle]

Steve Totaro wrote:
> That is interesting.  I have an intel C2D and I can only see two
> procs, not four, is that normal?  Are you sure what you are saying is
>   

I believe Intel removed HyperThreading after it moved over to dual cores.

Doug

-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Quality problems with ISDN PRI

[Rob Hillis]

To the best of my knowledge, multi-core processors are not hyperthreaded 
- certainly my Core 2 Quad processor isn't.  I would expect a Core 2 Duo 
to be the same.



Steve Totaro wrote:

On Thu, Apr 24, 2008 at 11:18 AM, Rob Hillis <[EMAIL PROTECTED]> wrote:
  

 Every CPU core shows up as a separate CPU under Linux.  For those that have
hyperthreaded processors, a single core processor will show up as two
processors - assuming you have hyperthreading enabled.




That is interesting.  I have an intel C2D and I can only see two
procs, not four, is that normal?  Are you sure what you are saying is
correct?  I am obviously running SMP.

Thanks
Steve totaro

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!DSPAM:4811f27e213018250212299!


  
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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Benjamin Jacob <[EMAIL PROTECTED]> wrote:
> 
> Also,  I don't think my SIP gateway uses Silence suppression, because the 
> same SIP gateway
> connections work fine with another Asterisk server.

OK, I think you need to home in on the differences between the server(s)
that work fine and the one that doesn't.

What version of kernel is it running? If it less than 2.6.13, make sure
you change "#if 0" to "#if 1" in ztdummy.c so that USE_RTC still gets
enabled.

Try "watch -dn 1 cat /proc/interrupts" and check that the RTC interrupts
are going up by 1024 per second. This is with ztdummy running.

What else is going on on this server? Does it have any virtual machines
on it? Does it have X Windows running? What does "top" show?

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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[asterisk-users] Upgrading to 1.4

[lotusscript]

A good while back when installing 1.2 there were major issues with UK
callerid.  Asterisk 1.2 didn't recognise the callerid correctly because
of the way BT sent the information.  Sometimes before the first ring or
just after.  After applying a third party patch we got it to work.  We
were afraid to touch it after that  :-)  Has this problem now gone away
with 1.4?


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[asterisk-users] Cisco 7960 odd behaviour ...

[lotusscript]

Been using the Snom 360 and 190 for a while and decided to try the Cisco
7960.  The problem I'm seeing is the call terminates between 2:34 and
3:00 minutes.  This only happens when using Zap channels.  Internal
calls work fine.  No probs with the Snoms.  No errors show on the * box
when the line drops.

Anyone seen this?

Asterisk 1.2.14
Cisco Firmware: P0S3-08-8-00


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Re: [asterisk-users] Quality problems with ISDN PRI

[Steve Totaro]

On Thu, Apr 24, 2008 at 11:18 AM, Rob Hillis <[EMAIL PROTECTED]> wrote:
>
>  Every CPU core shows up as a separate CPU under Linux.  For those that have
> hyperthreaded processors, a single core processor will show up as two
> processors - assuming you have hyperthreading enabled.
>

That is interesting.  I have an intel C2D and I can only see two
procs, not four, is that normal?  Are you sure what you are saying is
correct?  I am obviously running SMP.

Thanks
Steve totaro

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Re: [asterisk-users] Digium B410P or Sangoma A502D?

[Tzafrir Cohen]

On Fri, Apr 25, 2008 at 09:53:16AM -0400, Andres wrote:

> All Sangoma Asterisk cards **except** the BRI cards will indeed use 
> Zaptel.  But the BRI cards use a totally independent driver that 
> communicates with the WOOMERA channel.  It does not use Zaptel at all 
> and therefore no software echo cancel or even HPEC will help.  Believe 
> me, we talked to Sangoma in detail about this.

Sangoma BRI cards can use Zaptel as well. They just need support for it
in Asterisk. This is either from bristuff or from Asterisk 1.6 .

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] Friday Apr 25 @ 12 Noon EDT VoIP Users Conference

[randulo]

exten => 1234,1,Dial(SIP/[EMAIL PROTECTED],60,D(22622#${PIN}#))

Digium specifically has asked me not to use the name Asterisk Users
Conference, but that is mostly what we talk about. We expect that to
change soon, however as we expand into more genera VoIP talk.

Today, FRIDAY April 25 2008 at 12 Noon Eastern Daylight Time ( 4PM UTC )
Call (724) 444-7444 and enter 22622# 1#

IRC  #voip-users-conference  on  Freenode.net

See http://VoipUsersConference.org for more info.

People from Voiceroute on the call:
Vikram Rangnekar, COO Voiceroute and Druid Community Honcho
Navin Kumar, CTO Voiceroute
Ming Yong, CEO Voiceroute

Things they will talk about this friday: Druid OSE v1.0.2

New Features:
1. Backup Manager (Snapshots)
2. Improved Queue & Agent Management
3. Static & Dynamic Agents
4. Visual Real-time Status for Trunks, Stations and Extensions
5. Extended call recoding support for extensions, queues and conferences

More: http://forums.voiceroute.org/showpost.php?p=375&postcount=1

Screenshots for Druid OSE is available from
http://www.voiceroute.org/druidose/screenshots

/r

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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Benjamin Jacob]

Tony Mountifield <[EMAIL PROTECTED]> wrote: In article <[EMAIL PROTECTED]>,
Benjamin Jacob  wrote:
> 
> One on my clients' machine had Asterisk 1.4.4. installed. The complained of 
> choppy Playback
> of gsm files.
> So scouring the internet gave me the solution of installing ztdummy and 
> loading it as a module.
> Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and 
> re-installed. Sill
> no effect.
> 
> Do I have to specify any parameter in the Asterisk compilation to look at 
> ztdummy/rtc? As
> far as I remember (am coming back to Asterisk after quite some time now), you 
> don't really
> need to set anything over there for any zaptel specific compilation?
> 
> And yes, all the files are gsm files and the codec used for the calls is ulaw.
> 
> I even tried converting those gsm files to wav using sox and then playing 
> them, but the
> behaviour is the same.
> 
> Any ideas anyone.. something I am missing ??

Firstly, check whether Asterisk has chan_zap loaded and access to zaptel:

*CLI> zap show channels
   Chan Extension  Context Language   MusicOnHold 
 pseudodefault
*CLI>

If you don't get pseudo shown, then you are not getting the benefit of
ztdummy.

However, the probably main cause of choppy sound is poor timing from the
SIP client (I'm assuming SIP), because Asterisk by default uses the incoming
stream to generate timing for the outbound stream.

There are two main things to try:

1. Make sure that the SIP clients are NOT using silence suppression (may
be referred to as VAD, bandwidth saving, or something similar).

2. If ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable
the line "internal_timing=yes". That should make it play out based on
internal zaptel timing instead of timing off the incoming stream, I think.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

Thanks Tony for the response.
zap show channels shows that things are fine, as you said :
*CLI> zap show channels
   Chan Extension  Context Language   MOH Interpret   
 pseudodefaultdefault

Tried setting internal_timing to yes as well. Still no difference.

Also,  I don't think my SIP gateway uses Silence suppression, because the same 
SIP gateway connections work fine with another Asterisk server.

This is getting seriously irritating now!!! Have tried all the tricks and tips 
I've been finding on the net.

Yeah, btw, even Meetme playback is choppy. So, I think its somehow related to 
timing. But I am not the expert. 

- Ben.

   
-
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Re: [asterisk-users] followme scenarios

[Jared Smith]

On Thu, 2008-04-24 at 18:57 -0700, ronald ramos wrote:
> ami using GotoIf correctly?

No, you're not using GotoIf() correctly.  In fact, there are several
problems with your syntax.  The first problem I spot is that the
GotoIf() application requires an expression, and you haven't supplied an
expression.  Expressions in Asterisk are easy to spot, as they're
enclosed in $[ ].  You also need to put ${ } around your variables when
you're trying to retrieve their value, and you've got an errant pipe in
the line as well.

Replace all the lines that look like:

GotoIf(FM = "NEVER"|?vm)

with this syntax:

GotoIf($["${FM}" = "NEVER"]?vm)

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Digium B410P or Sangoma A502D?

[Andres]

Olivier wrote:

>
>
> 2008/4/24 Andres <[EMAIL PROTECTED] >:
>
>  
>
>   You can chose 2/4/6 ports to buy and if you need more
> just add remoras up to 24 ports. 
>
>
> Is this still usable  within  1U server, when you cannot "stack" PCI 
> cards like this
>
> xxx
>
> xxx
>
>   but you must align them like this
>
> xxx  xxx

no, you must align the Remoras vertically.

>
>
>
>
> Andres.
>
>
>


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Re: [asterisk-users] MFC/R2 in chan_zap , Testers Wanted

[Lex Lethol]

Congrats on going forward with the project Moises.  MFC/R2 support on
chan_zap sounds great, looking forward on trying it out.

Regards,
Lex

On Thu, Apr 24, 2008 at 3:03 PM, Moises Silva <[EMAIL PROTECTED]> wrote:
> >  Unicall MFC/R2 is activelly maintained. by Moy. Actually it's a backport 
> > of the Steve driver (now coded for Callweaver derivative) to Asterisk (1.2, 
> > 1.4, and 1.6 soon). It works pretty well. In fact, it works more stable in 
> > 1.4 than the original Steve driver in 1.2, and with better sound under 
> > heavy loads.
>  >  The Asutunicall page can be found here:
>  >  http://www.moythreads.com/astunicall/
>
>  Hum, wonder who this moy is  hey wait, that's me! . Even when is
>  in my plans to keep giving general maintenance to chan_unicall, my
>  long term plan is to leave R2 support into chan_zap, so I would
>  recommend to all users to try chan_zap R2 support, the more users we
>  get the faster the driver will be stable enough to replace
>  chan_unicall, the less headaches you will have (I hope).
>
>  - Moy or Moisés Silva, same shit :-)
>
>
>
>  --
>  "I do not agree with what you have to say, but I'll defend to the
>  death your right to say it." Voltaire
>
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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Doug Lytle]

Tony Mountifield wrote:
> 2. If ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable
> the line "internal_timing=yes". That should make it play out based on
>   

One other thing comes to mind, make sure you compile with 'Don't 
optimize' if you're using gcc 4.2.2

Doug

-- 
 
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Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] Digium B410P or Sangoma A502D?

[Andres]

Matt Watson wrote:

>I haven;t used any BRI cards but... call me crazy but wouldn;t they still be 
>using Zaptel (even your sangoma... the script might just be configuring it for 
>you)...
>
>  
>
All Sangoma Asterisk cards **except** the BRI cards will indeed use 
Zaptel.  But the BRI cards use a totally independent driver that 
communicates with the WOOMERA channel.  It does not use Zaptel at all 
and therefore no software echo cancel or even HPEC will help.  Believe 
me, we talked to Sangoma in detail about this.

Andres.

>and btw, software echo cancel happens in the zaptel kernel driver... it has 
>nothing to do with the hardware (hence why its a software echo cancel)
>
>You also would of had the option of buying HPEC licenses for software echo 
>cancel from digium for a rather cheap price.
>
>--
>Matt
>
>From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Andres [EMAIL PROTECTED]
>Sent: Thursday, April 24, 2008 5:04 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [asterisk-users] Digium B410P or Sangoma A502D?
>
>We have tested both and they work fine.  The Sangoma is much easier to
>install as it does not depend on any other driver, you just run
>'setup-sangoma' and follow the instructions.  You don't have to fiddle
>with the linux kernel or  zaptel or chan_misdn.  It just works.  Plus
>its more modular.  You can chose 2/4/6 ports to buy and if you need more
>just add remoras up to 24 ports.  The Digium card is fixed to 4 ports,
>period.
>
>Having said that, make sure you stick with the version that has hardware
>echo cancel and not even try the other one.  We made the mistake of
>buying the first time without echo cancel expecting to test the
>'software echo cancel'.  But there is no such thing as 'software echo
>cancel' on this card.  I do not even understand why Sangoma would make a
>version without the hardware echo cancel.  You get some degree of echo
>on practically every call.
>
>Andres.
>
>
>
>Patrick wrote:
>
>  
>
>>Hi,
>>
>>I need to setup an Asterisk box with 4x ISDN BRI links. Looking at the
>>specs of various cards I favor the Digium B410P and Sangoma A502D
>>because of hardware echo cancellation. Does anyone have any experience
>>with either card, good or bad? Which one would you choose and why?
>>
>>Thanks for your insight.
>>
>>Regards,
>>Patrick
>>
>>
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>>
>>
>
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Re: [asterisk-users] No CallerID Transfer Problem

[Ken Williams]

Actually, the code below works perfectly to fix the transfer disconnect
problem.  I was asking of other, better ways, aside from manually
defining on all incoming calls a dummy CID.

To answer Steve's question, using a single TDM400 card for the incoming
PSTN (it's one line, a remote office that most of their communication is
done over IAX back to our main location).  The three handsets are
Grandstream GXP-2000 (let the flaming begin, we currently have about 40
GXP-2000's in production and yes, we've had strange issues, but they're
working quite well now).

Anyway, it's really not a huge deal, but I had work arounds.  I'd prefer
the 'usecallerid=no' type route instead of making a fix in the dialplan,
that's all I was looking for.

Ken 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: Friday, April 25, 2008 7:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No CallerID Transfer Problem

Try removing the quotes from the Caller*ID info.

Steve Davies wrote:
> 2008/4/24 Ken Williams <[EMAIL PROTECTED]>:
>> Came upon a problem today that I thought I'd see if it's by design, 
>> if I'm missing an option somewhere, or if my fix is the way to fix
it.
>>
>> We setup a remote location with a server, same as we've done with 
>> others, but for some reason when they would transfer an outside call 
>> anywhere it would pause for a few seconds and hang up the line.
>>
>> Well, after spending most of the day on it, it turns out it's because

>> they don't have callerID on the PSTN lines coming in through zaptel.

>> My first thought was, set "usecallerid=no" and all would be well, but

>> this didn't do any good.  After playing a bit longer I just set the
following:
>>
>> exten => 900,2,set(CALLERID(num)="606-555-1212")
>> exten => 900,3,set(CALLERID(name)="Outside Call")
>> exten => 
>> 900,4,Dial(${DIALEXTENSIONS},${RINGTIMER},${DIAL_OPTIONS})
>>
>> Now all works well.
>>
>> So is there another option somewhere to keep asterisk from killing a 
>> transfer without callerid?  This happened on both 1.4.17 & 1.4.18.1.
>>
>> Thanks,
>> Ken
> 
> Can I guess that they are using snom phones with firmware 7.1.30? I 
> encountered exactly that bug here, but only if I enabled "sendrpid" in

> the sip.conf of the asterisk system. Downgrading to a more-stable 
> 6.5.x snom firmware, or disabling "sendrpid" for all of the snom 
> devices fixed this in our case. (Roll on the next snom firmware
> release!)
> 
> If not, then can I suggest that you provide more detail of equipment 
> involved - PCI cards, handsets etc etc?
> 
> Hope that helps,
> Steve
> 
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--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design.  Based near
Birmingham, AL.  Now accepting clients worldwide.

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Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Benjamin Jacob <[EMAIL PROTECTED]> wrote:
> 
> One on my clients' machine had Asterisk 1.4.4. installed. The complained of 
> choppy Playback
> of gsm files.
> So scouring the internet gave me the solution of installing ztdummy and 
> loading it as a module.
> Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and 
> re-installed. Sill
> no effect.
> 
> Do I have to specify any parameter in the Asterisk compilation to look at 
> ztdummy/rtc? As
> far as I remember (am coming back to Asterisk after quite some time now), you 
> don't really
> need to set anything over there for any zaptel specific compilation?
> 
> And yes, all the files are gsm files and the codec used for the calls is ulaw.
> 
> I even tried converting those gsm files to wav using sox and then playing 
> them, but the
> behaviour is the same.
> 
> Any ideas anyone.. something I am missing ??

Firstly, check whether Asterisk has chan_zap loaded and access to zaptel:

*CLI> zap show channels
   Chan Extension  Context Language   MusicOnHold 
 pseudodefault
*CLI>

If you don't get pseudo shown, then you are not getting the benefit of
ztdummy.

However, the probably main cause of choppy sound is poor timing from the
SIP client (I'm assuming SIP), because Asterisk by default uses the incoming
stream to generate timing for the outbound stream.

There are two main things to try:

1. Make sure that the SIP clients are NOT using silence suppression (may
be referred to as VAD, bandwidth saving, or something similar).

2. If ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable
the line "internal_timing=yes". That should make it play out based on
internal zaptel timing instead of timing off the incoming stream, I think.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] No CallerID Transfer Problem

[Eric Wieling]

Try removing the quotes from the Caller*ID info.

Steve Davies wrote:
> 2008/4/24 Ken Williams <[EMAIL PROTECTED]>:
>> Came upon a problem today that I thought I'd see if it's by design, if I'm
>> missing an option somewhere, or if my fix is the way to fix it.
>>
>> We setup a remote location with a server, same as we've done with others,
>> but for some reason when they would transfer an outside call anywhere it
>> would pause for a few seconds and hang up the line.
>>
>> Well, after spending most of the day on it, it turns out it's because they
>> don't have callerID on the PSTN lines coming in through zaptel.  My first
>> thought was, set "usecallerid=no" and all would be well, but this didn't do
>> any good.  After playing a bit longer I just set the following:
>>
>> exten => 900,2,set(CALLERID(num)="606-555-1212")
>> exten => 900,3,set(CALLERID(name)="Outside Call")
>> exten => 900,4,Dial(${DIALEXTENSIONS},${RINGTIMER},${DIAL_OPTIONS})
>>
>> Now all works well.
>>
>> So is there another option somewhere to keep asterisk from killing a
>> transfer without callerid?  This happened on both 1.4.17 & 1.4.18.1.
>>
>> Thanks,
>> Ken
> 
> Can I guess that they are using snom phones with firmware 7.1.30? I
> encountered exactly that bug here, but only if I enabled "sendrpid" in
> the sip.conf of the asterisk system. Downgrading to a more-stable
> 6.5.x snom firmware, or disabling "sendrpid" for all of the snom
> devices fixed this in our case. (Roll on the next snom firmware
> release!)
> 
> If not, then can I suggest that you provide more detail of equipment
> involved - PCI cards, handsets etc etc?
> 
> Hope that helps,
> Steve
> 
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T-1, PRI, Frame Relay, Linux, and network design.  Based near 
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Re: [asterisk-users] noisy analog lines

[Tzafrir Cohen]

One other thing,

On Fri, Apr 25, 2008 at 10:06:08AM +0200, Ian wrote:
> Hi all
> 
> I have a small problem here.
> 
> We are running Asterisk 1.4.18 and libpri 1.4.3 (how can I check the 
> zaptel version again?) on an Intel core 2 duo 1.6Ghz, 2GB ram under 
> Ubuntu server.
> 
> We have 4 analog line coming into the box via a TDM 800 wildcard with 
> echo cancel module and quad fxo modules.
> 
> The server has been running smoothly with almost no problems for awhile now.
> 
> Recently I started picking up problem with the voice clarity on our end, 
> it sounds like a mobile going through a low signal patch. I asked the 
> person on the other end and they can hear me loud and clear.
> 
> I bumped the txgain up a notch a while back, can it be because of this?

You bumped both txgain and rxgain. txgain is for audio Asterisk
transmits to the device (the telephone line, in this case). rxgain is
for audio recieves from the device.

I suspect you don't really need both . And a value of 10 might actually
influence the echo canceller. Try removing the two and see how things
work. Generally this value (like most values) is applied on a reload
(with asterisk 1.6 you can even use 'zap set {hw|sw}gain') . Generally
remove all of those txgain and rxgain limes from your zapata.conf, and
leave potentially just one near the top of the [channels] section - it
will effect all of them. 

Use 'reload' or 'module reload chan_zap.so' to apply your changes and
ignore the scary messages about the "signalling" being ignored.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
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Re: [asterisk-users] No CallerID Transfer Problem

[Steve Davies]

2008/4/24 Ken Williams <[EMAIL PROTECTED]>:
>
> Came upon a problem today that I thought I'd see if it's by design, if I'm
> missing an option somewhere, or if my fix is the way to fix it.
>
> We setup a remote location with a server, same as we've done with others,
> but for some reason when they would transfer an outside call anywhere it
> would pause for a few seconds and hang up the line.
>
> Well, after spending most of the day on it, it turns out it's because they
> don't have callerID on the PSTN lines coming in through zaptel.  My first
> thought was, set "usecallerid=no" and all would be well, but this didn't do
> any good.  After playing a bit longer I just set the following:
>
> exten => 900,2,set(CALLERID(num)="606-555-1212")
> exten => 900,3,set(CALLERID(name)="Outside Call")
> exten => 900,4,Dial(${DIALEXTENSIONS},${RINGTIMER},${DIAL_OPTIONS})
>
> Now all works well.
>
> So is there another option somewhere to keep asterisk from killing a
> transfer without callerid?  This happened on both 1.4.17 & 1.4.18.1.
>
> Thanks,
> Ken

Can I guess that they are using snom phones with firmware 7.1.30? I
encountered exactly that bug here, but only if I enabled "sendrpid" in
the sip.conf of the asterisk system. Downgrading to a more-stable
6.5.x snom firmware, or disabling "sendrpid" for all of the snom
devices fixed this in our case. (Roll on the next snom firmware
release!)

If not, then can I suggest that you provide more detail of equipment
involved - PCI cards, handsets etc etc?

Hope that helps,
Steve

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Re: [asterisk-users] noisy analog lines

[Tzafrir Cohen]

(Not a real answer to your qustion, but still)

On Fri, Apr 25, 2008 at 10:06:08AM +0200, Ian wrote:
> Hi all
> 
> I have a small problem here.
> 
> We are running Asterisk 1.4.18 and libpri 1.4.3 (how can I check the 
> zaptel version again?) on an Intel core 2 duo 1.6Ghz, 2GB ram under 
> Ubuntu server.

The current version of the loaded Zaptel modules (works as of kernel
2.6.12 or so) -

  cat /sys/module/zaptel/version

The version of the installed module:

  modinfo zaptel | grep ^version

> 
> I ran a top and saw that the server only have about 16Mb free ram, can 
> this be a possible cause?

Generally, no. But where exactly do you see that?

[EMAIL PROTECTED]:~$ top -b -n 1 | head -n 5; free
top - 15:59:54 up 51 days,  7:12, 10 users,  load average: 0.00, 0.00, 0.00
Tasks: 132 total,   1 running, 124 sleeping,   7 stopped,   0 zombie
Cpu(s):  0.7%us,  0.6%sy,  0.2%ni, 92.2%id,  6.1%wa,  0.0%hi,  0.0%si,  0.0%st
Mem:505468k total,   497452k used, 8016k free,   115608k buffers
Swap:   979956k total,17980k used,   961976k free,58608k cached
 total   used   free sharedbuffers cached
Mem:505468 497452   8016  0 115608  58608
-/+ buffers/cache: 323236 182232
Swap:   979956  17980 961976


My system has 182232 kB free, not just 8016 kB. The extra free memory is
used by the system for improving access to the hardware rather than
being wasted. 8016 kB are being wasted right now.

In short: look at the second line at the output of 'free', or do the
math yourself.

-- 
   Tzafrir Cohen
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+972-50-7952406   mailto:[EMAIL PROTECTED]
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Re: [asterisk-users] Full queue issues

[Vinícius Fontes]

Oops, seems like I didn't realized something: the queue size can't be zero. I 
solved the problem by setting maxlen=1 and defining a timeout on the Queue() 
app. That way when all the agents are busy, the call gets diverted after 
[TIMEOUT] seconds, which is ok to me.


Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.

- "Vinícius Fontes" <[EMAIL PROTECTED]> escreveu:

> Hello everyone.
> 
> I got a little problem in here: I want to set up a queue so that if
> anything of these happens:
> 
> a) No agents logged in
> b) All agents busy
> 
> Then the user gets diverted somewhere. I used this (for testing
> purposes only, of course):
> 
> exten => 7080,1,Answer()
> exten => 7080,n,Queue(teste)
> exten => 7080,n,Goto(${QUEUESTATUS})
> exten => 7080,n(ERROR),NoOp(${QUEUESTATUS})
> exten => 7080,n,Hangup()
> exten => 7080,n(LEAVEEMPTY),Goto(ERROR)
> exten => 7080,n(TIMEOUT),Goto(ERROR)
> exten => 7080,n(JOINUNAVAIL),Goto(ERROR)
> exten => 7080,n(LEAVEUNAVAIL),Goto(ERROR)
> exten => 7080,n(JOINEMPTY),Goto(ERROR)
> exten => 7080,n(TIMEOUT),Goto(ERROR)
> 
> exten => *210,1,AddQueueMember(teste,SIP/${CALLERID(num)})
> exten => *210,n,UserEvent(RefreshQueue)
> exten => *210,n,Playback(agent-loginok)
> 
> exten => *220,1,RemoveQueueMember(teste,SIP/${CALLERID(num)})
> exten => *220,n,UserEvent(RefreshQueue)
> exten => *220,n,Playback(agent-loggedoff)
> 
> 
> 
> In queues.conf:
> 
> [teste]
> strategy=roundrobin
> music=default
> timeout=10
> retry=0
> maxlen=1
> ringinuse=no
> leavewhenempty=strict
> joinempty=strict
> 
> 
> Then I have those scenarios:
> 
> a) There is no agents logged in, a call tries to enter the queue, the
> ${QUEUESTATUS} variable is set to LEAVEEMPTY and the call is
> disconnected. Everything fine in here.
> 
> b) There is only one agent logged in, he's in a call (InUse), the call
> enters the queue and stays there. I would like the call NOT to enter
> the queue and the ${QUEUESTATUS} variable to be set to something
> different.
> 
> Am I missing something or it's just not possible? I'm using SIP phones
> for the agents and Asterisk 1.4.15.
> 
> 
> Att
> Vinícius Fontes
> Desenvolvimento
> Canall Tecnologia em Comunicações Ltda.
> 
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Re: [asterisk-users] Digium B410P or Sangoma A502D?

[Tzafrir Cohen]

On Thu, Apr 24, 2008 at 02:04:43PM -0300, Vinícius Fontes wrote:
>
> I have a box running a TE410P with echo cancelling and it works like 
> a charm. Set up once, forget about it.
> 

That card is E1/J1/T1 (like the Sangoma A10x cards), and not BRI.

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Re: [asterisk-users] Digium B410P or Sangoma A502D?

[Tzafrir Cohen]

On Fri, Apr 25, 2008 at 01:37:15PM +0200, Patrick wrote:
> 
> On Thu, 2008-04-24 at 20:17 -0400, Steve Totaro wrote:
> > The Sangoma kernel drivers are different than Zaptel, while running
> > the install script you are asked if you would like to generate the
> > Zaptel configs but it is not required, you must also run wancfg to
> > configure the cards beyond the Zaptel configs.  The Sangoma drivers
> > kind of run on top of the Zaptel.
> 
> Thanks for clearing that up. That's what I thought too.
> 
> > It seems that the newest wanpipe drivers and Zaptel 1.4 work without
> > the D chan patch which is very nice IMO, I hate patches.
> 
> That's good to hear. The less patches the better.
> 
> >   I have run
> > the BRIStuff install and it has tons of patches!  Kind of scary but it
> > works for it's purpose.
> 
> Yesterday I had to update a set of 1.2 Asterisk RPMs with the latest
> Junghanns patch. So I downloaded the latest Junghanns patch and had a
> look. Wow, that's a serious set of patches to zaptel, libpri and
> asterisk. Kind of scary indeed.
> 
> > I have only done BRI once but there was absolutely no echo by simply
> > setting echocancel=yes, echocancelwhenbridged=no.
> 
> Personally I use an Eicon Diva Server card with onboard echo can in my *
> box and calls to cellphones can still generate a ton of echo. Same
> applies to calls to POTS phones that are hooked up to copper that's in a
> rather bad state or use a €5 phone.
> 
> > I hear "might as well get the hardware EC board" quite a bit, but on
> > all the many dozens of PRIs I have installed, software EC has been
> > adequate (if needed at all).  It would have meant quite a bit of
> > wasted money that was better spent on a nice 48 port gigabit switch.
> 
> Lucky you :) Or your upstream telco has deployed some serious echo can
> boxes throughout its network doing echo cancellation already for you.

With the channels count of BRI, there's typically much less of a problem
with software echo cancellation. 

BTW: OSLEC is said to be usable with mISDN as well. Never really tried
it.

> 
> > I have tested both ways (hardware vs. software), no difference really
> > (Sangoma).  Sangoma actually sent me one of each before purchasing
> > seven quad cards to test if hardware EC was going to be required for
> > one deployment.  I returned the hardware EC card and ordered seven
> > quad PRI cards.
> > 
> > Maybe I am just lucky or have not had enough exposure to BRI but ISDN
> > is ISDN, right (it really is a question, I don't know)?
> 
> EuroISDN is EuroISDN. Doesn't matter if it's BRI or PRI.

It slightly does. PRI does not require supporting PtMP. Thus BRI requires
better EuroISDN support.

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[asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy

[Benjamin Jacob]

Are my messages getting through?

This is urgent!! Any pointers?


Benjamin Jacob <[EMAIL PROTECTED]> wrote: Date: Thu, 24 Apr 2008 23:23:08 -0700 
(PDT)
From: Benjamin Jacob <[EMAIL PROTECTED]>
Subject: Playback / Background / Read choppy, but musiconhold fine, even with 
ztdummy
To: asterisk-users@lists.digium.com

 
Hello ppl,

One on my clients' machine had Asterisk 1.4.4. installed. The complained of 
choppy Playback of gsm files.
So scouring the internet gave me the solution of installing ztdummy and loading 
it as a module.
Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and 
re-installed. Sill no effect.

Do I have to specify any parameter in the Asterisk compilation to look at 
ztdummy/rtc? As far as I remember (am coming back to Asterisk after quite some 
time now), you don't really need to set anything over there for any zaptel 
specific compilation?

And yes, all the files are gsm files and the codec used for the calls is ulaw.

I even tried converting those gsm files to wav using sox and then playing them, 
but the behaviour is the same.

Any ideas anyone.. something I am missing ??

TiA,

- Ben.



   

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Re: [asterisk-users] Asterisk for larg

[Steve Totaro]

On Fri, Apr 25, 2008 at 7:03 AM, gmail <[EMAIL PROTECTED]> wrote:
>
>
> Does anybody know how to off-load an Asterisk Box so that to distribute its
> functions like IVR and VoiceMail or its PTSN gateway function into different
> servers? in this case , will the installation of Asterisk on each server
> differe and how these different  servers will interact as a single logical
> -vs physical- server? thx alot
> ___


This is how I design any large system.  I try to break up
functionality as much as possible on separate boxes.  Basically, you
just want to load just the modules and applications you need on that
particular box in modules.conf.

Thanks,
Steve Totaro

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[asterisk-users] Upgrading to 1.4

[lotusscript]

A good while back when installing 1.2 there were major issues with UK 
callerid.  Asterisk 1.2 didn't recognise the callerid correctly because 
of the way BT sent the information.  Sometimes before the first ring or 
just after.  After applying a third party patch we got it to work.  We 
were afraid to touch it after that  :-)  Has this problem now gone away 
with 1.4?

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Re: [asterisk-users] Asterisk for larg

[Mike Trest - On Travel]

Hmmm,

IMHO this is a fundamental SIP architecture issue.

To meet my understanding of distribution, this would required a proxy 
function before the call "answer()" on the Asterisk.   If , in an 
ideal world, this proxy function were to be in the path before 
answer(), the proxy would need added intelligence to examine the 
INVITE and deduce from it's content the need for IVR or VoiceMail. 
This is an Active Call Director functionality.  IMHO, this well 
within the capabilities of several SIP proxy packages available today.


Once answered, the Asterisk will remain in the call path.   If that 
is ok in your need for distribution, you can push the call onward via 
any Asterisk dialplan extension to be serviced by a whole farm of 
other Asterisks for specific chores.


..mike..




At 07:03 AM 4/25/2008, you wrote:
Does anybody know how to off-load an Asterisk Box so that to 
distribute its functions like IVR and VoiceMail or its PTSN gateway 
function into different servers? in this case , will the 
installation of Asterisk on each server differe and how these 
different  servers will interact as a single logical -vs physical- 
server? thx alot

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Re: [asterisk-users] Digium B410P or Sangoma A502D?

[Patrick]

On Fri, 2008-04-25 at 08:21 +0200, Olivier wrote:
> 
> 
> 2008/4/24 Patrick <[EMAIL PROTECTED]>:
> Hi,
> 
> I need to setup an Asterisk box with 4x ISDN BRI links.
> Looking at the
> specs of various cards I favor the Digium B410P and Sangoma
> A502D
> because of hardware echo cancellation. Does anyone have any
> experience
> with either card, good or bad? Which one would you choose and
> why?
> 
> Maybe you should also care about PCI or PCI-E interface. 

Good point. The client already bought the server. Iirc it's an HP DL360
G5 so I will investigate which kind of slots it has.

Thanks for pointing that out.

Regards,
Patrick


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Re: [asterisk-users] Digium B410P or Sangoma A502D?

[Patrick]

On Fri, 2008-04-25 at 08:13 +0200, Olivier wrote:
> 
> 2008/4/25 Matt Watson <[EMAIL PROTECTED]>:
> I haven;t used any BRI cards but... call me crazy but wouldn;t
> they still be using Zaptel (even your sangoma... the script
> might just be configuring it for you)...
> 
> and btw, software echo cancel happens in the zaptel kernel
> driver...
>  
> I think (but I'm not certain) that it's correct :
> Digium's B410P are used through chan_misdn.
> 
> (Please, do not hesitate to correct this)

Afaik that is correct.

> it has nothing to do with the hardware (hence why its a
> software echo cancel)
> 
> You also would of had the option of buying HPEC licenses for
> software echo cancel from digium for a rather cheap price.
> This also doesn't apply to chan_misdn hardware ... 

Afaik you are right. I don't think you can use HPEC with the Digium BRI
card. Please correct me if I'm wrong. If you don't mind using some
experimental stuff give OSLEC a try since mISDN/chan_misdn does support
that and apparently OSLEC does a great job of canceling echo.

Regards,
Patrick


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Re: [asterisk-users] Digium B410P or Sangoma A502D?

[Patrick]

On Thu, 2008-04-24 at 20:17 -0400, Steve Totaro wrote:
> The Sangoma kernel drivers are different than Zaptel, while running
> the install script you are asked if you would like to generate the
> Zaptel configs but it is not required, you must also run wancfg to
> configure the cards beyond the Zaptel configs.  The Sangoma drivers
> kind of run on top of the Zaptel.

Thanks for clearing that up. That's what I thought too.

> It seems that the newest wanpipe drivers and Zaptel 1.4 work without
> the D chan patch which is very nice IMO, I hate patches.

That's good to hear. The less patches the better.

>   I have run
> the BRIStuff install and it has tons of patches!  Kind of scary but it
> works for it's purpose.

Yesterday I had to update a set of 1.2 Asterisk RPMs with the latest
Junghanns patch. So I downloaded the latest Junghanns patch and had a
look. Wow, that's a serious set of patches to zaptel, libpri and
asterisk. Kind of scary indeed.

> I have only done BRI once but there was absolutely no echo by simply
> setting echocancel=yes, echocancelwhenbridged=no.

Personally I use an Eicon Diva Server card with onboard echo can in my *
box and calls to cellphones can still generate a ton of echo. Same
applies to calls to POTS phones that are hooked up to copper that's in a
rather bad state or use a €5 phone.

> I hear "might as well get the hardware EC board" quite a bit, but on
> all the many dozens of PRIs I have installed, software EC has been
> adequate (if needed at all).  It would have meant quite a bit of
> wasted money that was better spent on a nice 48 port gigabit switch.

Lucky you :) Or your upstream telco has deployed some serious echo can
boxes throughout its network doing echo cancellation already for you.

> I have tested both ways (hardware vs. software), no difference really
> (Sangoma).  Sangoma actually sent me one of each before purchasing
> seven quad cards to test if hardware EC was going to be required for
> one deployment.  I returned the hardware EC card and ordered seven
> quad PRI cards.
> 
> Maybe I am just lucky or have not had enough exposure to BRI but ISDN
> is ISDN, right (it really is a question, I don't know)?

EuroISDN is EuroISDN. Doesn't matter if it's BRI or PRI.

>   Now on
> analog, that is a horse of a different color, also the  phone on
> either side, but especially your side can be the culprit (older
> Grandstream for one) Polycom seems to eliminate much of this.

Point well taken. You get what you pay for.

Regards,
Patrick

> Thanks,
> Steve Totaro
> 
> On Thu, Apr 24, 2008 at 7:50 PM, Matt Watson <[EMAIL PROTECTED]>
> wrote:
> > I haven;t used any BRI cards but... call me crazy but wouldn;t they
> still be using Zaptel (even your sangoma... the script might just be
> configuring it for you)...
> >
> >  and btw, software echo cancel happens in the zaptel kernel
> driver... it has nothing to do with the hardware (hence why its a
> software echo cancel)
> >
> >  You also would of had the option of buying HPEC licenses for
> software echo cancel from digium for a rather cheap price.
> >
> >  --
> >  Matt
> >  
> >  From: [EMAIL PROTECTED]
> [EMAIL PROTECTED] On Behalf Of Andres
> [EMAIL PROTECTED]
> >  Sent: Thursday, April 24, 2008 5:04 PM
> >  To: Asterisk Users Mailing List - Non-Commercial Discussion
> >  Subject: Re: [asterisk-users] Digium B410P or Sangoma A502D?
> >
> >
> >
> >  We have tested both and they work fine.  The Sangoma is much easier
> to
> >  install as it does not depend on any other driver, you just run
> >  'setup-sangoma' and follow the instructions.  You don't have to
> fiddle
> >  with the linux kernel or  zaptel or chan_misdn.  It just works.
> Plus
> >  its more modular.  You can chose 2/4/6 ports to buy and if you need
> more
> >  just add remoras up to 24 ports.  The Digium card is fixed to 4
> ports,
> >  period.
> >
> >  Having said that, make sure you stick with the version that has
> hardware
> >  echo cancel and not even try the other one.  We made the mistake of
> >  buying the first time without echo cancel expecting to test the
> >  'software echo cancel'.  But there is no such thing as 'software
> echo
> >  cancel' on this card.  I do not even understand why Sangoma would
> make a
> >  version without the hardware echo cancel.  You get some degree of
> echo
> >  on practically every call.
> >
> >  Andres.
> >
> >
> >
> >  Patrick wrote:
> >
> >  >Hi,
> >  >
> >  >I need to setup an Asterisk box with 4x ISDN BRI links. Looking at
> the
> >  >specs of various cards I favor the Digium B410P and Sangoma A502D
> >  >because of hardware echo cancellation. Does anyone have any
> experience
> >  >with either card, good or bad? Which one would you choose and why?
> >  >
> >  >Thanks for your insight.
> >  >
> >  >Regards,
> >  >Patrick



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[asterisk-users] Cisco 7960 odd behaviour ...

[lotusscript]

Been using the Snom 360 and 190 for a while and decided to try the Cisco 
7960.  The problem I'm seeing is the call terminates between 2:34 and 
3:00 minutes.  This only happens when using Zap channels.  Internal 
calls work fine.  No probs with the Snoms.  No errors show on the * box 
when the line drops.

Anyone seen this?

Asterisk 1.2.14
Cisco Firmware: P0S3-08-8-00

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[asterisk-users] Asterisk for larg

[gmail]

Does anybody know how to off-load an Asterisk Box so that to distribute its 
functions like IVR and VoiceMail or its PTSN gateway function into different 
servers? in this case , will the installation of Asterisk on each server 
differe and how these different  servers will interact as a single logical -vs 
physical- server? thx alot ___
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Re: [asterisk-users] help...i cant do more...

[Bruno Pereira]

Thanks for the answers.
I need to say that this command is executed from another machine, with the
command ssh
because in ocalhost is all ok, with sudo or with root.

I will try that trace to see if it helps me, but the bg probem is start the
service from another machine with ssh .

Best regards,
Bruno Pereira


On 4/24/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> On Thu, Apr 24, 2008 at 11:23:21AM -0700, Steve Edwards wrote:
> > On Thu, 24 Apr 2008, Bruno Pereira wrote:
> >
> > > ssh etx9  'sudo /etc/init.d/asterisk start'
> > > [EMAIL PROTECTED]:~$ ssh etx9  'sudo /etc/init.d/asterisk start'
> > > start ini
> > > Starting asterisk: [  OK  ]
> > > decrease the verbosity level to zero: OK
> > > start fim
> > >
> > > and just stays there, like waiting for something.
> >
> > Sudo recently (?) added a new parameter in /etc/sudoers that caused me a
> > lot of grief. Comment out "Defaults requiretty" and see if it helps.
> >
> > Also, your custom /etc/init.d/asterisk script ("start fim?") may not be
> > redirecting stdin/stdout/stderr correctly.
>
> Or Asterisk may not be closing file descriptors properly?
>
> I have such an issue with some scripts getting hanged on installation of
> the package asterisk on Debian until I restart Asterisk.
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] Cisco to Asterisk migration

[Duncan Turnbull]

Hi Femi

We have about 50 Cisco 7960s on one site off Asterisk 1.4.18

Its all SIP and it doesn't stress a P3 system much at all.

I am not sure what phones you are using - the 7960s are not hard to configure, 
a bit of process to convert from the Cisco Skinny to
SIP (using SIP v8.6) but everything seems to work well. The 7961s or 7971s use 
an XML config which is probably 

Everything loads off the TFTP server. We are using the Linksys POE Switches 
SFE2000P which seem okay but don't always like to be
fully loaded 

Things I would work on are automating or simplifying the provisioning (doesn't 
change that much once its done), firmware upgrades,
and getting to know the config files well. 

Cheers Duncan

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Femi
Sent: Friday, 25 April 2008 21:34
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Cisco to Asterisk migration

Hi Guys,
I have client with a Cisco 2690 call manager solution that wants to upgrade
but cannot stomach the costs of continuing with Cisco

The installation will go up to 100 users
The client currently has about 40 Cisco phones and would like to continue
with these phones with the odd Polycom

I'm looking at plugging in an Asterisk box and using the existing Cisco box
as a PSTN gateway only

Has anyone on the list done this?
Any pitfalls or tips you would like to share?


Thanks

Femi


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Re: [asterisk-users] Digium B410P or Sangoma A502D?

[Patrick]

On Thu, 2008-04-24 at 19:50 -0400, Matt Watson wrote:
> I haven;t used any BRI cards but... call me crazy but wouldn;t they
> still be using Zaptel (even your sangoma... the script might just be
> configuring it for you)...

Last time I installed them the Sangoma drivers sorta run on top of
zaptel.

> and btw, software echo cancel happens in the zaptel kernel driver...
> it has nothing to do with the hardware (hence why its a software echo
> cancel)
> 
> You also would of had the option of buying HPEC licenses for software
> echo cancel from digium for a rather cheap price.

That's a solution too but a bit of a risk since afaik the hardware echo
cancellation can perform better than the software one. The CEO at the
client has echo to certain POTS destinations and I want to make sure
everything on his side is top notch.

Regards,
Patrick

___
> From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Andres [EMAIL 
> PROTECTED]
> Sent: Thursday, April 24, 2008 5:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Digium B410P or Sangoma A502D?
> 
> We have tested both and they work fine.  The Sangoma is much easier to
> install as it does not depend on any other driver, you just run
> 'setup-sangoma' and follow the instructions.  You don't have to fiddle
> with the linux kernel or  zaptel or chan_misdn.  It just works.  Plus
> its more modular.  You can chose 2/4/6 ports to buy and if you need more
> just add remoras up to 24 ports.  The Digium card is fixed to 4 ports,
> period.
> 
> Having said that, make sure you stick with the version that has hardware
> echo cancel and not even try the other one.  We made the mistake of
> buying the first time without echo cancel expecting to test the
> 'software echo cancel'.  But there is no such thing as 'software echo
> cancel' on this card.  I do not even understand why Sangoma would make a
> version without the hardware echo cancel.  You get some degree of echo
> on practically every call.
> 
> Andres.
> 
> 
> 
> Patrick wrote:
> 
> >Hi,
> >
> >I need to setup an Asterisk box with 4x ISDN BRI links. Looking at the
> >specs of various cards I favor the Digium B410P and Sangoma A502D
> >because of hardware echo cancellation. Does anyone have any experience
> >with either card, good or bad? Which one would you choose and why?
> >
> >Thanks for your insight.
> >
> >Regards,
> >Patrick
> >
> >
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> >
> >
> >
> 
> 
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Re: [asterisk-users] Digium B410P or Sangoma A502D?

[Patrick]

On Thu, 2008-04-24 at 17:04 -0400, Andres wrote:
> We have tested both and they work fine.  The Sangoma is much easier to 
> install as it does not depend on any other driver, you just run 
> 'setup-sangoma' and follow the instructions.  You don't have to fiddle 
> with the linux kernel or  zaptel or chan_misdn.  It just works.  Plus 
> its more modular.  You can chose 2/4/6 ports to buy and if you need more 
> just add remoras up to 24 ports.  The Digium card is fixed to 4 ports, 
> period.
> 
> Having said that, make sure you stick with the version that has hardware 
> echo cancel and not even try the other one.  We made the mistake of 
> buying the first time without echo cancel expecting to test the 
> 'software echo cancel'.  But there is no such thing as 'software echo 
> cancel' on this card.  I do not even understand why Sangoma would make a 
> version without the hardware echo cancel.  You get some degree of echo 
> on practically every call.

Thanks Andres. Your feedback is most helpful.

Regards,
Patrick



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Re: [asterisk-users] Forking in Dialplan

[Tobias Ahlander]

>Date: Thu, 24 Apr 2008 06:54:27 -0700 (PDT)
>From: Steve Edwards <[EMAIL PROTECTED]>
>Subject: Re: [asterisk-users] Forking in Dialplan
>To: Asterisk Users Mailing List - Non-Commercial Discussion
 >   
>Message-ID: <[EMAIL PROTECTED]>
>Content-Type: text/plain; charset="x-unknown"

>> - "Tobias Ahlander" <[EMAIL PROTECTED]> escreveu:

>>> Is it possible to somehow fork in the dialplan? Say a call comes in.
>>> Then I want to wait 30 seconds and then write in a database, but at the
>>> same time while I wait I want to go on with other commands too.

>On Thu, 24 Apr 2008, Vin??cius Fontes wrote:

>> You can call an AGI script that will call another script. That last one
>> would wait 10 seconds and write in the database. The following example
>> works for me:
>>
>> /var/lib/asterisk/agi-bin/agi-test.agi:
>>
>> #!/bin/bash
>> nohup /root/helloworld.sh 1>/dev/null 2>/dev/null &
>> exit 0
>>
>> /root/helloworld.sh:
>>
>> #!/bin/bash
>> sleep 10
>> echo "Hello world!" >> /root/helloworld.txt
>> exit 0

>Why do you need the first AGI? Would:

> exten = _x.,n,system(nohup /root/helloworld.sh 1>/dev/null 2>&1 &)

>suit your needs?

>Thanks in advance,
>
>Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
>Newline Fax: +1-760-731-3000


Thank you Steve, this seems to work just as I want it to. Now I just have to
figure out how to send variables to a system call, but I think I have that
covered somewhere :)

Best regards,
Tobias
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[asterisk-users] Cisco to Asterisk migration

[Femi]

Hi Guys,
I have client with a Cisco 2690 call manager solution that wants to upgrade
but cannot stomach the costs of continuing with Cisco

The installation will go up to 100 users
The client currently has about 40 Cisco phones and would like to continue
with these phones with the odd Polycom

I'm looking at plugging in an Asterisk box and using the existing Cisco box
as a PSTN gateway only

Has anyone on the list done this?
Any pitfalls or tips you would like to share?


Thanks

Femi


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[asterisk-users] SIP response 400 on attended transfer

[Mathieu]

Hi,

I've a probleme since few weeks that I don't be able to solve.

I use Thomson ST2030 phone and I've an error when I want to do an 
attended transfer with the soft key.
The receiver of the transfer return an : Got SIP response 400 "Bad 
Request" back from 192.168.2.13

The direct transfer with soft key works fine and attended transfer with 
*2 (features.conf) works too.

Can you help me ?

This is what I've during a sip debug of the receiver of the transfer.

localhost*CLI> sip debug ip 192.168.2.13
SIP Debugging Enabled for IP: 192.168.2.13
-- Executing Macro("SIP/9714-08ade6a8", "externe|[TEL NUMBER]|[NAME] 
<[TEL NUMBER]>") in new stack
-- Executing Set("SIP/9714-08ade6a8", "CALLERID(all)=[NAME] <[TEL 
NUMBER]>") in new stack
-- Executing Dial("SIP/9714-08ade6a8", "Zap/g1/[TEL NUMBER]||tT") in 
new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/[TEL NUMBER]
-- Zap/2-1 is proceeding passing it to SIP/9714-08ade6a8
-- Zap/2-1 is ringing
-- Zap/2-1 answered SIP/9714-08ade6a8
-- Started music on hold, class 'default', on channel 'Zap/2-1'
-- Stopped music on hold on Zap/2-1
-- Executing Macro("SIP/9714-08affc58", "local|9710") in new stack
-- Executing Answer("SIP/9714-08affc58", "") in new stack
-- Executing Dial("SIP/9714-08affc58", "SIP/9710|20|tT") in new stack
Apr 11 17:43:59 NOTICE[22239]: chan_sip.c:2142 sip_call: called party 
number = 9710
We're at 192.168.2.254 port 10330
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 192.168.2.13:5060:
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK655cd59a;rport
From: "[USER NAME]" ;tag=as548073e8
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Apr 2008 15:43:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 2453 2453 IN IP4 192.168.2.254
s=session
c=IN IP4 192.168.2.254
t=0 0
m=audio 10330 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 9710
localhost*CLI>
<-- SIP read from 192.168.2.13:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK655cd59a;rport
From: "[USER NAME]";tag=as548073e8
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines) ---
localhost*CLI>
<-- SIP read from 192.168.2.13:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK655cd59a;rport
From: "[USER NAME]";tag=as548073e8
To: ;tag=c0a80101-6628fd
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Allow: 
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: 
Content-Length: 0


--- (9 headers 0 lines) ---
-- SIP/9710-08aec0b8 is ringing
localhost*CLI>
<-- SIP read from 192.168.2.13:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK655cd59a;rport
From: "[USER NAME]";tag=as548073e8
To: ;tag=c0a80101-6628fd
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Allow: 
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: 
Content-Type: application/sdp
Content-Length: 199

v=0
o=9710 6696599 6696599 IN IP4 192.168.2.13
s=-
c=IN IP4 192.168.2.13
t=0 0
m=audio 41000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (10 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.13:41000
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 
(nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: 
set_destination: Parsing  for 
address/port to send to
set_destination: set destination to 192.168.2.13, port 5060
Transmitting (no NAT) to 192.168.2.13:5060:
ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK7a52f4a4;rport
From: "[USER NAME]" ;tag=as548073e8
To: ;tag=c0a80101-6628fd
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/9710-08aec0b8 answered SIP/9714-08affc58
-- Started music on hold, class 'default', on channel 
'SIP/9710-08aec0b8'
-- Stopped music on hold on SIP/9710-08aec0b8
set_destination: Parsing  for 
address/port to send to
set_destination: set destination to 192.168.2.13, port 5060
Transmitting (no NAT) to 192.168.2.13:5060:
INFO sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK7a6adfe4;rport
From: "[USER NAME]" ;tag=as548073e8
To: ;tag=c0a80101-6628fd
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INFO
User-Agent:

[asterisk-users] DNS Problems during zaptel upgrade

[Hanna Wallin]

Hi List!

 

I got this error while upgrading zaptel:

 

make -C firmware hotplug-install DESTDIR=

make[1]: Entering directory `/usr/src/zaptel-1.4.7.1/firmware'

Attempting to download zaptel-fw-oct6114-064-1.05.01.tar.gz

--10:53:09--  
http://downloads.digium.com/pub/telephony/firmware/releases/zaptel-fw-oct6114-064-1.05.01.tar.gz

Resolving downloads.digium.com... failed: Temporary failure in name resolution.

make[1]: *** [zaptel-fw-oct6114-064-1.05.01.tar.gz] Error 1

make[1]: Leaving directory `/usr/src/zaptel-1.4.7.1/firmware'

make: *** [install-firmware] Error 2

 

Is there a workaround? I've tried downloading the 
zaptel-fw-oct6114-064-1.05.01.tar.gz manually and placed it in the correct 
folder, but no luck.

 

 

/ hanna

 

 

Hanna Wallin
System Development

Direct: +46 (0)8 736 77 29
Mobile: +46 (0)73 414 13 38
Fax: +46 (0)8 736 77 91
E-mail: [EMAIL PROTECTED]  

 

PocketMobile Communications AB 
Wenner-Gren Center
Sveavägen 168, 3 tr
113 46 Stockholm

Nordic web page: www.pocketmobile.se http://www.pocketmobile.se> 
International web page: www.pocketmobileworld.com 
http://www.pocketmobileworld.com/> 

 

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[asterisk-users] noisy analog lines

[Ian]

Hi all

I have a small problem here.

We are running Asterisk 1.4.18 and libpri 1.4.3 (how can I check the 
zaptel version again?) on an Intel core 2 duo 1.6Ghz, 2GB ram under 
Ubuntu server.

We have 4 analog line coming into the box via a TDM 800 wildcard with 
echo cancel module and quad fxo modules.

The server has been running smoothly with almost no problems for awhile now.

Recently I started picking up problem with the voice clarity on our end, 
it sounds like a mobile going through a low signal patch. I asked the 
person on the other end and they can hear me loud and clear.

I bumped the txgain up a notch a while back, can it be because of this?

I ran a top and saw that the server only have about 16Mb free ram, can 
this be a possible cause?

My zapata.conf and zaptel.conf are below.

Thanks in advance
Ian
> # less /etc/zaptel.conf
> # Autogenerated by /usr/sbin/zapconf on Fri Feb 29 16:12:07 2008 -- do 
> not hand edit
> # Zaptel Configuration File
> #
> # This file is parsed by the Zaptel Configurator, ztcfg
> #
> # Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
> fxsks=1
> fxsks=2
> fxsks=3
> fxsks=4
> # channel 5, WCTDM/0/4, no module.
> # channel 6, WCTDM/0/5, no module.
> # channel 7, WCTDM/0/6, no module.
> # channel 8, WCTDM/0/7, no module.
>
> # Global data
>
> loadzone= za
> defaultzone = za
> # less /etc/asterisk/zapata.conf
> [trunkgroups]
> ; define any trunk groups
>
> [channels]
> ;hardware channels
>
> ;default
> ;groep nommers en rede
> ; 1 => Landlyn
> ; 2 => Selfoon
>
> ; Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
> ;;; line="1 WCTDM/0/0"
> signalling=fxs_ks
> callerid=asreceived
> context=incoming_calls
> group=2
> busydetect=yes
> usecallerid=yes
> hidecallerid=no
> callwaiting=no
> threewaycalling=yes
> transfer=yes
> echocancel=yes
> pulsedial=no
> callprogress=yes
> busycount=5
> subscribecontext=GXP_BLF
> overlapdial=no
> toneduration=200
> txgain=10.0
> rxgain=10.0
> channel => 1
>
> ;;; line="2 WCTDM/0/1 FXSLS"
> signalling=fxs_ks
> callerid=asreceived
> context=incoming_calls
> group=1,2
> busydetect=yes
> usecallerid=yes
> hidecallerid=no
> callwaiting=no
> threewaycalling=yes
> transfer=yes
> echocancel=yes
> pulsedial=no
> callprogress=yes
> busycount=5
> subscribecontext=GXP_BLF
> txgain=10.0
> rxgain=10.0
> overlapdial=yes
> channel => 2
>
> ;;; line="3 WCTDM/0/2"
> signalling=fxs_ks
> callerid=asreceived
> context=incoming_calls
> group=1
> busydetect=yes
> usecallerid=yes
> hidecallerid=no
> callwaiting=no
> threewaycalling=yes
> transfer=yes
> echocancel=yes
> pulsedial=no
> callprogress=yes
> busycount=5
> subscribecontext=GXP_BLF
> txgain=10.0
> rxgain=10.0
> overlapdial=yes
> channel => 3
>
> ;;; line="4 WCTDM/0/3"
> signalling=fxs_ks
> callerid=asreceived
> context=incoming_calls
> group=1
> busydetect=yes
> usecallerid=yes
> hidecallerid=no
> callwaiting=no
> threewaycalling=yes
> transfer=yes
> echocancel=yes
> pulsedial=no
> callprogress=yes
> busycount=5
> subscribecontext=GXP_BLF
> txgain=20.0
> rxgain=10.0
> overlapdial=yes
> channel => 4


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[asterisk-users] using "m" switch in dialplan

[Ian]

Hi all

Sorry for reposting this, but there haven't been an answer to my 
previous message.

I have a minor inconvenience here.

I want to use the "m" switch in the "dial" command on our outgoing lines
to play music to the caller whilst asterisk and our telecoms provider
connects the call. It works, as long as the called person is available.
When the called  phone (my mobile in this case) is off and it redirects
to voicemail, asterisk does not detect that the call is answered and
continues to play the music.

The reason I want to use the switch is that sometimes there is a silence
period that has been known to last up to 1 minute, before the called
party's phone begins to ring, and I want to fill that gap with something
else, I tried the "r" switch as well but for some reason the Grandstream
I tested on did not like the "r" switch, my x-lite did however indicate
ringing.

If any of you can direct me to the right site I would be realy greatfull.

If you need any more info, I will be only to happy to provide it, the
piece of my dialplan is below.

Thanks in advance

Regards
Ian

>
> exten => 9876,1,Progress()
> exten => 9876,n,dial(ZAP/1/0720311294,,m(default))
> exten => 9876,n,Hangup(
>



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[asterisk-users] Play sounds to both caller and callee at the same time

[Tobias Ahlander]

Hello,

I'm having problems with LIMIT_PLAYAUDIO_CALLEE in the Dial application. I
want to play the limit file to both caller and callee at the same time, but
it plays the limit file first to the caller and then to the callee. I
searched the list and found someone with the same problem back in '06, but
couldn't find any solution for the problem :(

Anyone knows?

Thanks,
Best regards,
Tobias
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Re: [asterisk-users] Disable transfer on all calls

[Grey Man]

> > > Thanks to your answers, but i found more beautiful way to do this -
> > > there is some system variable __TRANSFER_CONTEXT, which defines context
> > > to handle the transfered number, so you can create a new context and
> > > there you can do anything with transfered call - i just hang it up.
> > >

It's only relevant for blind transfers. For attended transfers that
mechanism won't work.

Regards,

Greyman.

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