Re: [asterisk-users] Fax Server for Asterisk

[Randall]

On 05/30/2012 12:02 AM, Danny Dias wrote:

Hi all,

Does Hylafax and IAXmodem works with analog lines? or only with E1?

I've been checking some commercial solutions (in case Asterisk is not 
on site, and the customer wants a Fax Server as standalone), i saw 
FaxBack and Linkcom e-fax


But again, if Hylafax and iaxmodem works also with analog lines, that 
would be better to use. Could you please confirm? any place to check 
How-To on Hylafax and Iaxmodem?


not sure what your requirements are, but i can +1 the reliability of 
hylafax, have been using it for years problem free although only +- 10 
faxes a day, and its running on an old leftover pc with a cheap modem on 
a analog line.


i use it in combination with avantfax as a web front end and email2fax, 
its a cheap and Free solution, easy to use, low maintenance etc


never used iaxmodem though



Many thanks!!!

2012/5/29 Carlos Alvarez >



On Tue, May 29, 2012 at 8:03 AM, Warren Selby
mailto:wcse...@selbytech.com>> wrote:

On Tue, May 29, 2012 at 3:10 AM, Danny Dias
mailto:ing.diasda...@gmail.com>> wrote:

Hello,

For those customers with only analog lines, who ask for
fax2email and email2fax, whats the most reliable solution
available and tested with Asterisk?

Thanks



I've been real happy with using HylaFax+ and Iaxmodem
implementations.



We have a few Hylafax servers in our network.  Both it and
IAXmodem are a real bear to learn at first (well, so is Asterisk)
but when you get them working, they are rock solid.  I hadn't even
thought about it, but it's been at least a year since I logged
into any of our Hylafax servers and did anything to them.  They
just work.

I would estimate I put in a solid 30 hours into learning and
configuring the first server, and then some more time learning
additional capabilities and best practices.  But again, since
doing that, it's been totally hands-off.

I will add though that we also use Fax for Asterisk simply to
receive and turn faxes into PDF for some customers, and that is
perfectly stable also.


-- 
Carlos Alvarez

TelEvolve
602-889-3003 



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--
www.danntel.net 
_sip:danny4...@thesipschool.com _
sip:dann...@opensips.org 






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Re: [asterisk-users] how to show used "wrong password"

[Randall]

On 03/13/2012 11:06 PM, Dave Platt wrote:

Ouch.  That isn't going to be so easy to spot, then!  You would have to guess
a bunch of likely passwords, fake up a challenge with some known nonce, and
compare the response against those you would expect with each of the various
possible passwords.  (You've already got the Source Code to do all this, of
course.)

You'll have to try the selective unplugging method instead .

There may be a way to do this, even in the face of the nonce-and-hash
security system.

As I understand it:  when a system (re)registers with a good
password, what you'll typically see is:

-  A registration request from the client (with the client's ID
in the SIP parameters)

-  A response from Asterisk, saying something on the order of
"Stale authentication.  Try again.  Here's a new nonce for you."

-  Another registration request from the same client, specifying
the newly-issued nonce, and having a hash based on that nonce and
the shared secret.

-  An "OK" response from Asterisk.

When a system (re)registers, and has the wrong password/secret,
the exchange will be different.

-  A registration request from the client (with the client's ID
in the SIP parameters)

-  A response from Asterisk, saying something on the order of
"Stale authentication.  Try again.  Here's a new nonce for you."

-  Another registration request from the same client, specifying
the newly-issued nonce, and having a hash based on that nonce and
the shared secret.

-  A response from Asterisk, rejecting the second registration request
with something like a "bad digest" error.

So, if you examine all of the SIP protocol exchanges taking place,
you should see a whole bunch of successful four-way handshakes (from
clients that have the correct secrets), and an occasional four-way
handshake failure (from the one client that has the wrong password in
its configuration).

You won't be able to tell what password the client is actually trying
to use - that's the whole point of the nonce-and-hash approach -
but you'll be able to identify its client name, and (unless the
far end is using a NAT or proxy) its IP address.

To pin down the actual location of the client, you'll either have
to go there, or have someone at the remote site do some investigation
and (possibly) packet tracing on the LAN.


this will be of little use in this situation, the location is a shared 
office space/building in Vietnam and the local hands i have already 
checked our computers for soft phones, but quit possible some machines 
got swapped there or some local admin installed it somewhere  for 
testing purposes... and the local hands i have, not really usefull 
explaining them to look up the meaning of "packet tracing"




Or, I suppose one could simply use Asterisk to try to phone the
device or softphone in question, at whatever address it called in
from, and ask whoever answers the phone to disable it!


this was my original idea yes, but how can i call it without it being 
registered?






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Re: [asterisk-users] how to show used "wrong password"

[Randall]

On 03/13/2012 03:53 PM, Kevin P. Fleming wrote:

On 03/13/2012 08:11 AM, A J Stiles wrote:

On Tuesday 13 March 2012, Randall wrote:

hi all,

have asterisk set up in combination with fail2ban.
all works as expected only there is 1 extension that is trying to
register with a wrong password causing fail2ban to block the IP 
address,

normally that is ok behaviour but i have several extensions on that IP
address.
. snip .
anyway to see which "wrong password" is being used?


tcpflow.

(And don't underestimate the power of simply disconnecting things 
until it

works .  last thing you disconnected was the faulty one.)


This will not help. Assuming we are talking about a SIP REGISTER here, 
the password is *not* sent in the request. Asterisk issues a challenge 
including a randomly generated value (called a 'nonce'), then the UA 
attempting to register responds to that challenge with an MD5 digest 
of a string composed of various elements, including both the nonce and 
the shared secret ('password'). Asterisk computes the same digest 
internally, and if they match, then the assumption is that both ends 
know the shared secret.


By their very nature, digest functions are not reversible; given the 
MD5 digest present in an SIP request containing an Authorization 
header, there is no way to figure out what shared secret was used in 
the computation of that digest. Since you know the nonce and the other 
portions of the calculation, you could attempt to try various 'likely' 
passwords to see if any of them result in the same digest value... 
this is called the brute-force method, and it could take a *very* long 
time to arrive at a shared secret that would allow the endpoint to 
register.



confirmed,

doesn't work

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Re: [asterisk-users] how to show used "wrong password"

[Randall]

On 03/13/2012 02:11 PM, A J Stiles wrote:

On Tuesday 13 March 2012, Randall wrote:

hi all,

have asterisk set up in combination with fail2ban.
all works as expected only there is 1 extension that is trying to
register with a wrong password causing fail2ban to block the IP address,
normally that is ok behaviour but i have several extensions on that IP
address.
. snip .
anyway to see which "wrong password" is being used?

tcpflow.

(And don't underestimate the power of simply disconnecting things until it
works .  last thing you disconnected was the faulty one.)



Thanks will give that a try.

p.s.
 i know the method, only problem that its a time consuming process (in 
this case it includes a 9000 km travel and not all equipment on that 
side is mine)




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[asterisk-users] how to show used "wrong password"

[Randall]

hi all,

have asterisk set up in combination with fail2ban.
all works as expected only there is 1 extension that is trying to 
register with a wrong password causing fail2ban to block the IP address, 
normally that is ok behaviour but i have several extensions on that IP 
address.


someone has once setup this extension as a test, but it seems to be 
impossible to find where it was installed (probably a softphone) and its 
a bit far away for me to go check it out.


since there is no way to stop the troubling extension i figured i might 
as well let it connect by setting the wrongly used password as correct, 
at least in that case the other extensions won't be blocked and i can 
try to call that extension myself.



anyway to see which "wrong password" is being used?


much obliged,

Randall

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[asterisk-users] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server

[Randall Degges]

Hi Everyone,

I've been trying to get asynchronous AGIs working in some Asterisk code I
have. I'm using Asterisk 1.8.7.0, and I'm very familiar with dialplan and
AGI scripting overall. Here's my problem: I can't get Asterisk to execute
*any* AGIs asynchronously.

Firstly, I discovered asynchronous AGIs via "Asterisk: The Definitive
Guide". The asynchronous AGI information I read can be found online, here:
http://ofps.oreilly.com/titles/9780596517342/AGI.html (scroll down to the
section titled "Async AGI--AMI Controlled AGI").

According to the book, since Asterisk 1.6.0 the AGI dialplan application has
been able to execute AGI scripts asynchronously, via the syntax:

exten => s,1,AGI(async:script)

According to the book, using the "async:" prefix should have Asterisk run
the AGI script in the background and instantly continue executing dialplan
code.

So here's my Asterisk dialplan code that's being run:

[hangup]
exten => s,1,AGI(async:/etc/asterisk/scripts/hangup.py)
exten => s,n,Return()

Pretty simple context--essentially my AGI script just does some call clean
up logic before a caller hangs up, talking to a few web servers and
generating statistics for later usage. What happens when Asterisk executes
this context, is:

WARNING[7911]: res_agi.c:1622 launch_script: Failed to execute
'/var/lib/asterisk/agi-bin/async:/etc/asterisk/scripts/hangup.py': File does
not exist.

As you can see, Asterisk is ignoring the async: directive, and treating it
as part of the AGI script path.

Is there anyway for me to make asynchronous AGIs work? I've tried searching
online to no avail.

I'd greatly appreciate any responses, thanks for your time.

-Randall

-- 
Randall Degges
*http://rdegges.com/*
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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

On 06/01/2011 01:12 PM, Karsten Wemheuer wrote:
> Hi randall,
> 
> Am Mittwoch, den 01.06.2011, 10:00 +0200 schrieb randall:
>>> > i get the following errors:
>>> > pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel
>>> > of span 2
>>> >
>>> >
>>> > Your telco provider has crc on or off , that is not matching with
>>> > your server cross check with them.
>>> > and this problem solve 4 problems
>>> >
>>> >
>>> thanks for the reply,
>>>
>>> what is crc (same as crc-4?)
>>> and where can i set this?
>>>
>>> same crc or crc4
>>>
>>> --
>> adding crc as follows, span=1,1,0,ccs,ami,crc, causes DAHDI to not load
>> at all
> 
> As I can see from Your first post, You are using BRI in
> point-to-multipoint mode. On BRI lines there is nothing like CRC/CRC4
> and that is the reason, why the config is not loading any more.
> 
> On a PTMP line there may be some CRC-errors from time to time, when the
> provider shuts down the line, which is normal in some countries. But
> this has nothing to do with Your initial problem.
> 
> Unfortunately I don't know a solution for Your problem. It may be a
> hardware issue.

is there a way to trace down the hardware causing this?


> 
> HTH,
> 
> Karsten
> 
> 
> 
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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

On 06/01/2011 06:28 PM, Steve Davies wrote:
> On 1 June 2011 15:10, randall  wrote:
>> On 06/01/2011 03:55 PM, randall wrote:
>>> On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
>>>> On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
>>>>> Hi all,
>>>>>
>>>>> After running fine for a few months now asterisk seems to hang
>>>>> frequently , still functioning but the DAHDI channels seem busy  (users
>>>>> report a busy signal when calling or being called)
>>>>>
>>>>> A reboot will allow it to run for another day or maybe 2  or 3 till the
>>>>> problem occurs again.
>>>>>
>>>>>
>>>>> running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
>>>>> 2.6.32-5-686
>>>>>
>>>>> i get the following errors:
>>>>> pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of 
>>>>> span 2
>>>>>
>>>>> (happens on all 4 spans)
>>>>>
>>>>> and the following in dmesg:
>>>>> [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252
>>>>> [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:current
>>>>> packet[0..2]: 55 55 FC
>>>>> [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71
>>>>>
>>>>>
>>>>> Channel 0/1, span 1 got hangup, cause 18
>>>>
>>>> Is this happening in the middle of a call? Or only a while after the
>>>> call ended?
>>>>
>>>
>>> the "bad fcs" messages seem to happen random
>> there seems to be a relation indeed, have seen them happen randomly
>> quite spurious, but they indeed tend to happen a while after the call is
>> made.
>>>
>>> the hangup happens when a call through DAHDI is attempted,
>>> (usually after it has been working fine for a while a day or 2)
> 
> In my experience, FCS errors are caused by line quality issues, and
> usually (not always) are in the telco's equipment. If they are only
> happening occasionally, it may be a marginal, but mostly-OK signal on
> the wire.
> 
> Do you also get occasional poor-quality audio on calls? The issue will
> happen more when a call is being setup, or is progressing because
> there are more frames being exchanged when a call is in progress.

audio quality seems to be fine ( e.g. had no complaints about that)

> 
> I have also seen a bad component or dry solder on a voice card cause
> this, and even a badly made ISDN cable can be part of the problem. If
> none of that helps, I would ask the telco to put a trace on the line.

would these FCS errors cause the system to become unresponsive leading
to "Channel 0/1, span 1 got hangup, cause 18" after a period of working
fine, or is this an unrelated issue?


> 
> Hope that helps,
> Steve
> 
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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

On 06/01/2011 05:42 PM, Tzafrir Cohen wrote:
> On Wed, Jun 01, 2011 at 04:10:34PM +0200, randall wrote:
>> On 06/01/2011 03:55 PM, randall wrote:
>>> On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
>>>> On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
>>>>> Hi all,
>>>>>
>>>>> After running fine for a few months now asterisk seems to hang
>>>>> frequently , still functioning but the DAHDI channels seem busy  (users
>>>>> report a busy signal when calling or being called)
>>>>>
>>>>> A reboot will allow it to run for another day or maybe 2  or 3 till the
>>>>> problem occurs again.
>>>>>
>>>>>
>>>>> running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
>>>>> 2.6.32-5-686
>>>>>
>>>>> i get the following errors:
>>>>> pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of 
>>>>> span 2
>>>>>
>>>>> (happens on all 4 spans)
>>>>>
>>>>> and the following in dmesg:
>>>>> [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252
>>>>> [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:current
>>>>> packet[0..2]: 55 55 FC
>>>>> [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71
>>>>>
>>>>>
>>>>> Channel 0/1, span 1 got hangup, cause 18
>>>>
>>>> Is this happening in the middle of a call? Or only a while after the
>>>> call ended?
>>>>
>>>
>>> the "bad fcs" messages seem to happen random
>> there seems to be a relation indeed, have seen them happen randomly
>> quite spurious, but they indeed tend to happen a while after the call is
>> made.
> 
> A while after a call is made? A while after a call is ended?

kept an eye on this and it seems to happen after a call is ended (+- 25
- 30 seconds) and only when dialed out, but not when another call is in
progress.


> 
> Maybe the provider intentionally sets layer 1 down ("to save power")?
sounds logical with the behaviour mentioned above

> 
> That makes sense on PtMP, though I was not aware of this being used on
> PtP.
> 
i'm clueless on this, telco is China Unicom btw

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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

On 06/01/2011 03:55 PM, randall wrote:
> On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
>> On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
>>> Hi all,
>>>
>>> After running fine for a few months now asterisk seems to hang
>>> frequently , still functioning but the DAHDI channels seem busy  (users
>>> report a busy signal when calling or being called)
>>>
>>> A reboot will allow it to run for another day or maybe 2  or 3 till the
>>> problem occurs again.
>>>
>>>
>>> running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
>>> 2.6.32-5-686
>>>
>>> i get the following errors:
>>> pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2
>>>
>>> (happens on all 4 spans)
>>>
>>> and the following in dmesg:
>>> [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252
>>> [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:current
>>> packet[0..2]: 55 55 FC
>>> [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71
>>>
>>>
>>> Channel 0/1, span 1 got hangup, cause 18
>>
>> Is this happening in the middle of a call? Or only a while after the
>> call ended?
>>
> 
> the "bad fcs" messages seem to happen random
there seems to be a relation indeed, have seen them happen randomly
quite spurious, but they indeed tend to happen a while after the call is
made.
> 
> the hangup happens when a call through DAHDI is attempted,
> (usually after it has been working fine for a while a day or 2)


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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
> On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
>> Hi all,
>>
>> After running fine for a few months now asterisk seems to hang
>> frequently , still functioning but the DAHDI channels seem busy  (users
>> report a busy signal when calling or being called)
>>
>> A reboot will allow it to run for another day or maybe 2  or 3 till the
>> problem occurs again.
>>
>>
>> running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
>> 2.6.32-5-686
>>
>> i get the following errors:
>> pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2
>>
>> (happens on all 4 spans)
>>
>> and the following in dmesg:
>> [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252
>> [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:current
>> packet[0..2]: 55 55 FC
>> [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71
>>
>>
>> Channel 0/1, span 1 got hangup, cause 18
> 
> Is this happening in the middle of a call? Or only a while after the
> call ended?
> 

the "bad fcs" messages seem to happen random

the hangup happens when a call through DAHDI is attempted,
(usually after it has been working fine for a while a day or 2)

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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

On 06/01/2011 10:07 AM, mahesh katta wrote:
> 
> 
> On Wed, Jun 1, 2011 at 1:30 PM, randall  <mailto:rand...@songshu.org>> wrote:
> 
> 
> > > i get the following errors:
> > > pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
> D-channel
> > > of span 2
> > >
> > >
> > > Your telco provider has crc on or off , that is not
> matching with
> > > your server cross check with them.
> > > and this problem solve 4 problems
> > >
> > >
> > thanks for the reply,
> >
> > what is crc (same as crc-4?)
> > and where can i set this?
> >
> > same crc or crc4
> >
> > --
> adding crc as follows, span=1,1,0,ccs,ami,crc, causes DAHDI to not load
> at all
> 
>span=1,1,0,ccs,ami,crc4
crc4 has the same effect

> 
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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

> > i get the following errors:
> > pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel
> > of span 2
> >
> >
> > Your telco provider has crc on or off , that is not matching with
> > your server cross check with them.
> > and this problem solve 4 problems
> >
> >
> thanks for the reply,
> 
> what is crc (same as crc-4?)
> and where can i set this?
> 
> same crc or crc4
> 
> --
adding crc as follows, span=1,1,0,ccs,ami,crc, causes DAHDI to not load
at all

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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

On 06/01/2011 09:04 AM, mahesh katta wrote:
> 
> 
> On Wed, Jun 1, 2011 at 11:36 AM, randall  <mailto:rand...@songshu.org>> wrote:
> 
> Hi all,
> 
> After running fine for a few months now asterisk seems to hang
> frequently , still functioning but the DAHDI channels seem busy  (users
> report a busy signal when calling or being called)
> 
> A reboot will allow it to run for another day or maybe 2  or 3 till the
> problem occurs again.
> 
> 
> running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
> 2.6.32-5-686
> 
> i get the following errors:
> pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel
> of span 2
>  
> 
> Your telco provider has crc on or off , that is not matching with
> your server cross check with them.
> and this problem solve 4 problems
>  
> 
thanks for the reply,

what is crc (same as crc-4?)
and where can i set this?

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[asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

Hi all,

After running fine for a few months now asterisk seems to hang
frequently , still functioning but the DAHDI channels seem busy  (users
report a busy signal when calling or being called)

A reboot will allow it to run for another day or maybe 2  or 3 till the
problem occurs again.


running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
2.6.32-5-686

i get the following errors:
pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2

(happens on all 4 spans)

and the following in dmesg:
[ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252
[ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:current
packet[0..2]: 55 55 FC
[ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71


Channel 0/1, span 1 got hangup, cause 18

According to my searching on the web the last two weeks i understand
that the first mentioned error can indicate a wrong configuration
setting or buggy hardware.

Have tried many options but the dahdi config as copied over from the
same machine (with elastix 1.4) seems the only one that seem to work

anybody has a clue?
much appreciated

some gory details below,

HARDWARE:
D201GLY2 motherboard
usb:001/002  xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware


DAHDI CONFIG:

dahdi-channels.conf

; Span 1: XBUS-00/XPD-00 "Xorcom XPD #00/00: BRI_TE" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel => 1-2
context = default
group = 63

; Span 2: XBUS-00/XPD-01 "Xorcom XPD #00/01: BRI_TE"
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel => 4-5
context = default
group = 63

; Span 3: XBUS-00/XPD-02 "Xorcom XPD #00/02: BRI_TE"
group=0,13
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel => 7-8
context = default
group = 63

; Span 4: XBUS-00/XPD-03 "Xorcom XPD #00/03: BRI_TE"
group=0,14
context=from-pstn
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel => 10-11
context = default
group = 63


chan_dahdi.conf

[channels]
context=from-pstn
signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
pridialplan=local
prilocaldialplan=local
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
;Uncomment these lines if you have problems with the disconection of
your analog lines
;busydetect=yes
;busycount=3
immediate=no



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Re: [asterisk-users] Asterisk 1.6.1 Won't Play Default ULAW Files

[Randall Degges]

Hi everyone,

Just wanted to give a quick update. After a lot of testing, it turns out
that my problem was that I didn't have `format_pcm.so` loaded in
modules.conf.

I was unaware that `format_pcm.so` handled ULAW audio files. I guess when I
was looking for unnecessary modules, it stood out as unnecessary, as I was
looking for stuff with 'ulaw' in the name.

Anyhow, thanks for all of your help. Maybe this will be of use to someone
else in the future.

-Randall

On Tue, Aug 24, 2010 at 5:59 PM, Randall Degges  wrote:

> Hi Paul,
>
> I don't actually have any asterisk.conf file currently. I haven't had one
> on any of my systems for a long time. Could that be the issue? I've had this
> work on other systems with different module configurations.
>
> -Randall
>
> On Tue, Aug 24, 2010 at 5:20 PM, Paul Belanger <
> paul.belan...@polybeacon.com> wrote:
>
>> On Tue, Aug 24, 2010 at 8:02 PM, Randall Degges 
>> wrote:
>> > That's all the debugging information I have, if you need anything else
>> > please let me know. I get the feeling that this is related to me not
>> loading
>> > a required module somewhere in modules.conf, but the modules that I've
>> > listed there don't seem to contain any other ULAW-type modules that
>> would be
>> > required to play ULAW files.
>> >
>> What is astvarlibdir set to in asterisk.conf?
>>
>> --
>> Paul Belanger | dCAP
>> Polybeacon | Consultant
>> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
>> blog.polybeacon.com
>>
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>
>
>
> --
> Randall Degges
> *http://rdegges.com/*
>
>


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Re: [asterisk-users] Asterisk 1.6.1 Won't Play Default ULAW Files

[Randall Degges]

Hi Paul,

I don't actually have any asterisk.conf file currently. I haven't had one on
any of my systems for a long time. Could that be the issue? I've had this
work on other systems with different module configurations.

-Randall

On Tue, Aug 24, 2010 at 5:20 PM, Paul Belanger  wrote:

> On Tue, Aug 24, 2010 at 8:02 PM, Randall Degges  wrote:
> > That's all the debugging information I have, if you need anything else
> > please let me know. I get the feeling that this is related to me not
> loading
> > a required module somewhere in modules.conf, but the modules that I've
> > listed there don't seem to contain any other ULAW-type modules that would
> be
> > required to play ULAW files.
> >
> What is astvarlibdir set to in asterisk.conf?
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
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[asterisk-users] Asterisk 1.6.1 Won't Play Default ULAW Files

[Randall Degges]

Hi everyone,

I'm having an odd issue. I've been doing some testing over the past couple
weeks on some Asterisk modules / utilities, but have bumped into a problem
which I can't seem to resolve.

Asterisk can't seem to play the default sound files (ULAW) in my
environment. All necessary debugging information is included below. I'd love
to get anyone else's thoughts on this, what I may be doing wrong, or how to
resolve the issue. Thanks so much!

Asterisk Version: Asterisk 1.6.1.11
OS: ubuntu-sever 10.04, 64-bit
Kernel: 2.6.32-24-server #39-Ubuntu

Asterisk is running as `root` in the group `root`:

r...@tx-lu-dev:~# ps aux | grep asterisk
root  9671  0.0  0.1 250640  7420 ?Ssl  18:48   0:00
/usr/sbin/asterisk -U root -G root

Asterisk dialplan code used for testing (relevant snippet shown):

[tests]

exten => s,1,NoOp(***performing tests***)

exten => s,n,NoOp(language is: ${CHANNEL(language)})
exten => s,n,Playback(hello-world)
exten => s,n,Playback(tt-monkeys)
exten => s,n,Playback(jedi-extension-trick)

exten => s,n,NoOp(***ending tests***)
exten => s,n,Return()

Asterisk log (while running live):

-- Executing [...@tests:1] NoOp("SIP/TX-LU-CG-01-", "***performing
tests***") in new stack
-- Executing [...@tests:2] NoOp("SIP/TX-LU-CG-01-", "language is:
en") in new stack
-- Executing [...@tests:3] Playback("SIP/TX-LU-CG-01-",
"hello-world") in new stack
[2010-08-24 18:54:10] WARNING[9761]: file.c:650 ast_openstream_full: File
hello-world does not exist in any format
[2010-08-24 18:54:10] WARNING[9761]: file.c:953 ast_streamfile: Unable to
open hello-world (format 0x4 (ulaw)): No such file or directory
[2010-08-24 18:54:10] WARNING[9761]: app_playback.c:448 playback_exec:
ast_streamfile failed on SIP/TX-LU-CG-01- for hello-world
-- Executing [...@tests:4] Playback("SIP/TX-LU-CG-01-",
"tt-monkeys") in new stack
[2010-08-24 18:54:10] WARNING[9761]: file.c:650 ast_openstream_full: File
tt-monkeys does not exist in any format
[2010-08-24 18:54:10] WARNING[9761]: file.c:953 ast_streamfile: Unable to
open tt-monkeys (format 0x4 (ulaw)): No such file or directory
[2010-08-24 18:54:10] WARNING[9761]: app_playback.c:448 playback_exec:
ast_streamfile failed on SIP/TX-LU-CG-01- for tt-monkeys
-- Executing [...@tests:5] Playback("SIP/TX-LU-CG-01-",
"jedi-extension-trick") in new stack
[2010-08-24 18:54:10] WARNING[9761]: file.c:650 ast_openstream_full: File
jedi-extension-trick does not exist in any format
[2010-08-24 18:54:10] WARNING[9761]: file.c:953 ast_streamfile: Unable to
open jedi-extension-trick (format 0x4 (ulaw)): No such file or directory
[2010-08-24 18:54:10] WARNING[9761]: app_playback.c:448 playback_exec:
ast_streamfile failed on SIP/TX-LU-CG-01- for jedi-extension-trick
-- Executing [...@tests:6] NoOp("SIP/TX-LU-CG-01-", "***ending
tests***") in new stack
-- Executing [...@tests:7] Return("SIP/TX-LU-CG-01-", "") in new
stack

Here are my relevant sip.conf settings:

language=en
disallow=all
allow=ulaw

Here's a quick listing of the sound files on my system (only relevant ones
shown):

r...@tx-lu-dev:~# ls -la /var/lib/asterisk/sounds/en/hello-world.ulaw
-rw-r--r-- 1 root root 12021 2010-08-19 19:23
/var/lib/asterisk/sounds/en/hello-world.ulaw
r...@tx-lu-dev:~# ls -la /var/lib/asterisk/sounds/en/tt-monkeys.ulaw
-rw-r--r-- 1 root root 129440 2010-08-19 19:23
/var/lib/asterisk/sounds/en/tt-monkeys.ulaw
r...@tx-lu-dev:~# ls -la
/var/lib/asterisk/sounds/en/jedi-extension-trick.ulaw
-rw-r--r-- 1 root root 30400 2010-08-19 19:23
/var/lib/asterisk/sounds/en/jedi-extension-trick.ulaw

Here is my modules.conf file (I use autoload=no to prune out certain stuff).
Also, note that I have codec_ulaw.so loaded, which is, I think, the only
thing required in order to play ulaw files.

modules.conf: http://pastie.org/1113611

That's all the debugging information I have, if you need anything else
please let me know. I get the feeling that this is related to me not loading
a required module somewhere in modules.conf, but the modules that I've
listed there don't seem to contain any other ULAW-type modules that would be
required to play ULAW files.

Thanks for any help!

-Randall

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[asterisk-users] Removing `chan_dahdi.conf`

[Randall Degges]

Hi guys,

I'm currently playing around with optimizing an Asterisk install (trying to
remove as many possible configuration files as possible) for testing and
debugging purposes.

I've been able to remove most of the files and maintain an error-less
Asterisk full file, with a single exception: I can't seem to remove
`chan_dahdi.conf` without having Asterisk complain in the full logs:

ERROR[1182] chan_dahdi.c: Unable to load config chan_dahdi.conf

I'm only using DAHDI for dahdi_dummy timing, and nothing else. If I have an
empty `chan_dahdi.conf` file, then Asterisk won't complain at all, but if I
remove the empty file, Asterisk spits out the above error message.

So my questions are:

1. By removing `chan_dahdi.conf`, am I breaking any functionality that would
cause dahdi_dummy to not provide timing (or some other critical feature) in
my Asterisk environment?

2. Is there someway to let Asterisk know that I *only* need DAHDI for the
dahdi_dummy driver, and that the configuration file is unnecessary? (This
way I don't get errors in my logs.)

The point of me pursuing this is because it *seems* as if I can currently
`get by` without having the `chan_dahdi.conf` file at all, even with the
error message I'm getting in the logs. I just worry that there may be
another consequence for not having this file that I will experience at some
point, and since I haven't found one yet, I wanted to check with the experts
to get a definitive answer.

I'm currently using Asterisk 1.6.1.1.

I plan to go through the source code for the module loader to see what I can
find, but I figured I'd pop this email off anyhow to see what you all have
to say.

Thanks for your time.

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[asterisk-users] Execute AGI, Then Continue

[Randall Degges]

Hi all,

I'm running Asterisk 1.6.2.7 using the following pseudo-dialplan (not actual
dialplan, because of complexity):

[something]
exten => s,1,Answer()
exten => s,n,AGI(blah,arg1,arg2)
exten => s,n,Playback(blah)
exten => s,n,DoMoreStuff()
exten => s,n,Hangup()

What I'd like to do, is have Asterisk launch my AGI script and continue
executing dialplan without waiting for the AGI to finish executing. I'm
aware that I can do this manually in my AGI by forking, but I'd like to
avoid doing it that way if possible.

I remember reading something a long time ago saying that there was a way to
do this, but I can't seem to find that documentation again. Am I crazy, or
is this possible to do without modifying my code?

Thanks for all help.

-Randall
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Re: [asterisk-users] forward incomming line to modem

[randall]

On Wed, 2010-02-10 at 12:02 +0200, Tzafrir Cohen wrote:
> On Wed, Feb 10, 2010 at 07:52:06AM +0100, randall wrote:
> > hi All,
> > 
> > its probably very simple but i can't find the way to it.
> > 
> > i have some b410p cards and use them to connect to ISDN2, this works OK
> > for calling but i need to have 1 line to be send to the fax machine.
> > 
> 
> BRI fax machine?

no, its an external analog modem

> 
> > the fax machine is a modem connected on another machine with hylafax.
> > 
> > as far as i can figure out i need to set 1 of the slots, the one leading
> > to the fax, in the b410p in NT mode by setting the jumpers in the
> > opposite direction as the default.
> > 
> > Not sure how to proceed after that but dahdi keeps showing this port as
> > TE mode.

right, followed the wrong manual that told me to use the 2 switches
instead of the jumper. i feel stupd.
at least it shows NT now.

> 
> Look in your logs and you'll find the message "How cool would it be if
> someone implemented this mode!  For now, sucks for you".

pfff, is it just me or is the readability of the asterisk project not
always up to par? , wouldn't want to say it sucked.

> 
> Use bri_net instead of bri_net_ptmp .

i've got bri_net as created by dahdi_genconf on the line connected to
the modem, the incoming lines are bri_cpe.
the bri description doesn't make sense to me for connecting to an analog
device.

whats next? all the info i can find on this subject relates to hylafax
being installed on the same server.

have set
 
 faxdetect=incoming
 faxdetect=outgoing 
 faxdetect=yes

so i guess it would detect an incoming fax automagically.

what kind of extension do you need to pass the signal to?"
 


> 
> > 
> > anybody has a pointer, i'm pretty much stuck and i guess its simply that
> > i'm not feeding google the right buzz word.

> 



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[asterisk-users] forward incomming line to modem

[randall]

hi All,

its probably very simple but i can't find the way to it.

i have some b410p cards and use them to connect to ISDN2, this works OK
for calling but i need to have 1 line to be send to the fax machine.

the fax machine is a modem connected on another machine with hylafax.

as far as i can figure out i need to set 1 of the slots, the one leading
to the fax, in the b410p in NT mode by setting the jumpers in the
opposite direction as the default.

Not sure how to proceed after that but dahdi keeps showing this port as
TE mode.

anybody has a pointer, i'm pretty much stuck and i guess its simply that
i'm not feeding google the right buzz word.

thanks,

Randall



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Re: [asterisk-users] connect problem unless when verbose

[randall]

On 02/01/2010 08:38 AM, randall wrote:
> hi all,
>
> just had a terrible and sleepless weekend at the office trying to get
> asterisk going, its just tough love ;)
>
> have tried several asterisk versions but i currently have the following
> setup on debian lenny that kind of works.
> asterisk-1.6.2.0
> dahdi-linux-complete-2.2.0.2+2.2.0
> libpri-1.4.10.2
> freepbx-2.6.0
>
> setting up the sip devices is no problem at all, the difficulty i have
> is setting up 6xisdn2 lines with 2xb410p cards.
>
> besides the fact that i have no clue about what i'm doing i find the
> available documentation very very confusing, but i finally managed to
> make outgoing calls to my mobile this morning, sort off.
>
> when calling my mobile i hear a ringtone on my sip device and my mobile
> actually rings, YEAH!!!
> however, when i accept the call on my mobile my sip device keeps on
> ringing and my mobile gives no sound at all, when cancelling the call it
> simply cancels.
>
> except, and this i don't understand, i issue asterisk -rv (only with the
> v option), then i can connect and talk to myself, i often talk to myself
> when i spent a weekend at the office but this time its justifiable ;)
>
> anybody has a clue what could trigger this behavior???
> ,
>

update!!!

apparently it sometimes does work, randomly, guess the -v was a very 
lucky shot, i repeated it 20 times.
i do seem to get this message everytime a connection fails
[Feb  1 09:25:14] ERROR[2867] chan_dahdi.c: XXX Message longer than it 
should be?? XXX

after applying this patch below the problem seem to have dissapeared for now
https://issues.asterisk.org/view.php?id=16048



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[asterisk-users] connect problem unless when verbose

[randall]

hi all,

just had a terrible and sleepless weekend at the office trying to get 
asterisk going, its just tough love ;)

have tried several asterisk versions but i currently have the following 
setup on debian lenny that kind of works.
asterisk-1.6.2.0
dahdi-linux-complete-2.2.0.2+2.2.0
libpri-1.4.10.2
freepbx-2.6.0

setting up the sip devices is no problem at all, the difficulty i have 
is setting up 6xisdn2 lines with 2xb410p cards.

besides the fact that i have no clue about what i'm doing i find the 
available documentation very very confusing, but i finally managed to 
make outgoing calls to my mobile this morning, sort off.

when calling my mobile i hear a ringtone on my sip device and my mobile 
actually rings, YEAH!!!
however, when i accept the call on my mobile my sip device keeps on 
ringing and my mobile gives no sound at all, when cancelling the call it 
simply cancels.

except, and this i don't understand, i issue asterisk -rv (only with the 
v option), then i can connect and talk to myself, i often talk to myself 
when i spent a weekend at the office but this time its justifiable ;)

anybody has a clue what could trigger this behavior???

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Re: [asterisk-users] Virtual Asterisk Installation

[randall]

On 01/20/2010 11:28 PM, Felix Tiefenthaler wrote:
> Hi all!
>
> I've been reading this list for a few weeks and now this is my first
> post. :-)
>
> I'm planning to build a new VoIP telephone system at our company. It's
> just a small company with not more than 3-4 employees.
> The telephone system is not so important for us because each employee
> has it's own mobile phone.
>
> Because our company is a small one, we don't want to/we can't buy an
> expensive phone system. So we are going to use Asterisk.
> Additionally we don't want to obtain extra hardware. We have already a
> Server running with Linux (for Network monitoring).
> Because it's a waste to use this Server just for monitoring I thought
> about virtualizing. Now I want to run a machine with monitoring
> and a machine with Asterisk on this Server. I already bought a ISDN
> Card (berofix 400) with a S0 module.
>
> Now my big question: What kind of virtualization should I run on the
> Server? I have already used VMware ESXi and Proxmox.
> It would be very nice if there was a way to make snapshots (for
> "backup" purposes).
> I read about clock problems (physical time != virtual time) and so on.
> If I'm right this does not matter when using OpenVZ but when using
> KVM, XEN, ESX, ...
>
> Please tell me your opinion. I definitely want to run the Asterisk via
> virtualization - so we have to find a solution for this ;-)
>
> Thank you very much!
>
> felix
>
>
i never tried it for real, but i'm in the progress of setting up a PBX 
system and decided to go with a dedicated box since i need more pci 
slots then my regular virtualised box could provide.

However, i have heard that many have succesfully used asterisk on a 
VServer, which i usually use when installing new services.

have some old notes in the link below and the testing setup worked fine. 
VServer is more an advanced chroot so i don't think you will have the 
problems some mentioned when going for the heavier virtualisation types 
like Xen.
just make sure the card is reachable inside the guest

http://doku.songshu.org/doku.php?id=debian_lenny_asterisk


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Re: [asterisk-users] help with picking out a digium card.

[randall]

On 01/20/2010 06:00 PM, randall wrote:
> On 01/17/2010 09:25 PM, shawn bright wrote:
>
>> Hey all,
>>
>> We have been using a TDM400 card at work to provide our IVR.
>> We we have upgraded our server and now require the same capability,
>> but on a card that goes into a PCI Express.
>> Any suggestions would be greatly appreciated.
>>
>> oh, and it has to work with the zaptel drivers for linux.
>>
>> thanks all.
>>
>> sk
>>  
> i'm no expert on these cards nor of the drivers or even tried it myself,
> so please don't take my word on it.
>
> but... just happened to look for this last week, and i was under the
> assumption that PCI Express was backwards compatible with PCI
> http://en.wikipedia.org/wiki/PCI_Express
>
> Randall
>
>
i might take that back, eventhough still confused, guess my assumption 
above was wrong.

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Re: [asterisk-users] help with picking out a digium card.

[randall]

On 01/17/2010 09:25 PM, shawn bright wrote:
> Hey all,
>
> We have been using a TDM400 card at work to provide our IVR.
> We we have upgraded our server and now require the same capability, 
> but on a card that goes into a PCI Express.
> Any suggestions would be greatly appreciated.
>
> oh, and it has to work with the zaptel drivers for linux.
>
> thanks all.
>
> sk
i'm no expert on these cards nor of the drivers or even tried it myself, 
so please don't take my word on it.

but... just happened to look for this last week, and i was under the 
assumption that PCI Express was backwards compatible with PCI
http://en.wikipedia.org/wiki/PCI_Express

Randall

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Re: [asterisk-users] 10/100 voip phones and gigabit connection

[randall]

On 01/15/2010 05:01 PM, David Backeberg wrote:
> On Fri, Jan 15, 2010 at 1:54 AM, randall  wrote:
>
>> does anybody know of another solution to this or is my conclusion above
>> simply all the choice there is?
>>  
> So let me get this straight.
>
> You're planning on buying multiple Gigabit, PoE switches,

no, only 1 would be enough, think i will get this one in the link for 
the phones seperately
http://www.salland.eu/product/350730

> and you're
> quibbling over the price

yes, i like to quibble over price, i'm dutch ;)

> The gigabit
> PoE switches are not cheap,

i noticed

> at least if you're buying enterprise
> switches that actually deliver real gigabit, with full cross-sectional
> bandwidth. The cable isn't very much money, and if you double-wire
> now, you're ready when you have twice as many employees in the same
> space.
>

i already have 3 cat5e wires running to each workplace, at the moment 2 
are occupied with respectively 1 for a desktop and 1 for the "old" phone

> Next, you don't say what this office is like, but I'm going to let you
> in on a little secret. Most people in an office rarely spike to a full
> 100Mbit connection. Do some bandwidth monitoring on your network and
> you'll discover that.

we use a lot of email which is IMAP based, for normal text like this 
email it will not be a problem but we are send huge picture attachments, 
when using 100mbit it can get real sluggish at times especially when you 
are in a hurry to forward them, plus i like to have /home directories 
mounted on the server. a little extra never hurts.

> A gigabit ethernet phone is a nice thing to
> have, but it's more a marketing thing than an actual necessity.
>

i don't care for the phone to have gigabit connection, its about the 
desktops not losing gigabit connection

> Anybody that can afford a gigabit ethernet switching phone and true
> gigabit ethernet PoE backend can afford a second wire to every desk.
>

its not the wires at the desk i wanted to get rid off, i was hoping to 
use less in the server room, from the patch panel to the switch.

> Please let me know the use case if you find people can't be happy with
> a 100Mbit connection for the typical Windoze office environment.
>

windoze? people still use that? ;)
we mostley have Xubuntu based desktops running here, not that it matters 
or is absolutely necessary to have, but its a terrible thing to loose if 
you are used to it, also i use the LAN here sometimes for cluster testing.





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Re: [asterisk-users] 10/100 voip phones and gigabit connection

[randall]

On 01/15/2010 02:54 PM, Jeff LaCoursiere wrote:
>
> On Fri, 15 Jan 2010, randall wrote:
>
>
>>> Sure. My point was just that IF you only got one connection in the wall,
>>> its cheaper to get a switch than getting a phone with dual 1Gbit ports.
>>>
>>> Leif
>>>
>>>
>> OK, point taken.
>>
>> but i have 6xisdn2 and already 2x24 gigabit switches (will need to replace
>> one with a PoE version ) these connections include both desktops and current
>> phones.
>>
>> i was just hoping to cut back the amount of cabling with 50%, and when i
>> found out that most phones with 10/100/1000 connection cost about 250,-
>> euro's a piece instead of 90,- for a decent version with 10/100 it was a real
>> bummer, it would mean about doubling my budget.
>>
>>  
> I'm not sure you get it - he is saying you can eliminate the extra cable
> run to the desk, and place a small 5 port gigabit switch under the desk
> and drive both your PC and the phone from it.  Total cost per desk - 90 +
> 17 euros.  Significantly less than 250 euros for a dual gigabit port
> phone.  No change to your switching infrastructure in your machine room.
>
> j
>
i did get it,

its a good idea itself and i considered doing this a few years back, but 
as you can read from my reply i already have the separate cabling lying 
around and my old phones that i need to replace are plugged in there.

Since i have these already it would mean adding extra switches and 
untangling the huge amounts of cluttered wires under the desks, and i'm 
not sure if thats worth the trouble since it usually is a little dusty 
there and the ladies in the office always tend to get a little nervous 
when i stay down there too long ;)





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Re: [asterisk-users] 10/100 voip phones and gigabit connection

[randall]

On 01/15/2010 02:19 PM, Leif Neland wrote:


- Original Message -
*From:* randall <mailto:rand...@songshu.org>
*To:* asterisk-users@lists.digium.com
<mailto:asterisk-users@lists.digium.com>
*Sent:* Friday, January 15, 2010 2:11 PM
*Subject:* Re: [asterisk-users] 10/100 voip phones and gigabit
connection

On 01/15/2010 02:00 PM, Leif Neland wrote:


- Original Message -
*From:* randall <mailto:rand...@songshu.org>
*To:* asterisk-users@lists.digium.com
<mailto:asterisk-users@lists.digium.com>
*Sent:* Friday, January 15, 2010 7:54 AM
*Subject:* [asterisk-users] 10/100 voip phones and gigabit
connection
list.

i noticed that a lot of VOIP phones have a double network
interface
allowing you to use only 1 LAN cable for both the phone and your
desktop, a really nice feature that can save a lot of cable,
but most
are 10/100 connections while i have a gigabit network. Off
course there
are phones available with a Gigabit connection but these are
at least 3
to 4 times as expensive.

In a pinch, the cheapest 1Gbit switch I could find is 17 Eur with
5 ports.
Leif



its not the network switch that i'm worried about, its the build
in switch of the phones with the double network card

Sure. My point was just that IF you only got one connection in the 
wall, its cheaper to get a switch than getting a phone with dual 1Gbit 
ports.


Leif


OK, point taken.

but i have 6xisdn2 and already 2x24 gigabit switches (will need to 
replace one with a PoE version ) these connections include both desktops 
and current phones.


i was just hoping to cut back the amount of cabling with 50%, and when i 
found out that most phones with 10/100/1000 connection cost about 250,- 
euro's a piece instead of 90,- for a decent version with 10/100 it was a 
real bummer, it would mean about doubling my budget.



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Re: [asterisk-users] 10/100 voip phones and gigabit connection

[randall]

On 01/15/2010 02:00 PM, Leif Neland wrote:


- Original Message -
*From:* randall <mailto:rand...@songshu.org>
*To:* asterisk-users@lists.digium.com
<mailto:asterisk-users@lists.digium.com>
*Sent:* Friday, January 15, 2010 7:54 AM
*Subject:* [asterisk-users] 10/100 voip phones and gigabit connection

hi all,

just subscribed to the list and first mail, nice to be here.

Hopefully i'm in the right place for this question since i'm
planning a
little VOIP implementation at the moment and ran in to something
while
going through the shopping list.

i noticed that a lot of VOIP phones have a double network interface
allowing you to use only 1 LAN cable for both the phone and your
desktop, a really nice feature that can save a lot of cable, but most
are 10/100 connections while i have a gigabit network. Off course
there
are phones available with a Gigabit connection but these are at
least 3
to 4 times as expensive.

In a pinch, the cheapest 1Gbit switch I could find is 17 Eur with 5 ports.
Leif


its not the network switch that i'm worried about, its the build in 
switch of the phones with the double network card
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Re: [asterisk-users] 10/100 voip phones and gigabit connection

[randall]

On 01/15/2010 12:41 PM, Rob Hillis wrote:
> On 01/15/10 17:54, randall wrote:
>
>> hi all,
>>
>> i noticed that a lot of VOIP phones have a double network interface
>> allowing you to use only 1 LAN cable for both the phone and your
>> desktop, a really nice feature that can save a lot of cable, but most
>> are 10/100 connections while i have a gigabit network. Off course there
>> are phones available with a Gigabit connection but these are at least 3
>> to 4 times as expensive.
>>
>> another option would be to have both desktop and voip phone each a
>> dedicated line ( basically having 2 seperate networks ), already have
>> these in place from the old/current situation but i was hoping to clear
>> some cables.
>>
>> does anybody know of another solution to this or is my conclusion above
>> simply all the choice there is?
>>
>>
>>  
> You've hit the nail on the head.  A VoIP phone with two network ports is
> probably best thought of as a two port switch.  Like any switch, if you
> connect a gigabit NIC to a 10/100 switch, you'll end up with a 100
> megabit connection.  The only way to get a gigabit connection to your PC
> is via a phone that has gigabit ports, or have a separate cable back to
> the switch.
>
> Best practice is usually to segregate phone and PC networks anyway - it
> helps avoid degradation of VoIP quality when the LAN becomes heavily loaded.
>
>
i'll folow the best practice then in that case.

thanks for the confirmation Rob,


Randall


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[asterisk-users] 10/100 voip phones and gigabit connection

[randall]

hi all,

just subscribed to the list and first mail, nice to be here.

Hopefully i'm in the right place for this question since i'm planning a 
little VOIP implementation at the moment and ran in to something while 
going through the shopping list.

i noticed that a lot of VOIP phones have a double network interface 
allowing you to use only 1 LAN cable for both the phone and your 
desktop, a really nice feature that can save a lot of cable, but most 
are 10/100 connections while i have a gigabit network. Off course there 
are phones available with a Gigabit connection but these are at least 3 
to 4 times as expensive.

another option would be to have both desktop and voip phone each a 
dedicated line ( basically having 2 seperate networks ), already have 
these in place from the old/current situation but i was hoping to clear 
some cables.

does anybody know of another solution to this or is my conclusion above 
simply all the choice there is?

Much obliged,

Randall

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Re: [asterisk-users] simultaneous ring problem

[Randall Smith]

I think it may have been a NAT and/or reinvite issue.  I've now 
forwarded udp 1-2 to my pbx and turned off reinvite for all. 
NAT issues can be so difficult to diagnose sometimes.  I'll be glad to 
see it go with ipv6.

Randall

Randall Smith wrote:
> [macro-stdexten]
> exten => s,1,Dial(${ARG2},30,p)
> 
> exten => 
> 601555,1,Macro(stdexten,200,SIP/200&SIP/201&SIP/203&SIP/${VOICEPULSE_GATEWAY_OUT_A}/+1504555)
> 
> Where the real numbers have been replaced with 555.  What I'm trying 
> to do is ring my cell phone in addition to the local extensions. Funny 
> thing is the cell phone rings and I can press 1 to connect the call, but 
> if I remove the 'p' from the Dial command the cell phone rings, but when 
> I answer it the call doesn't connect.  I'm just talking to myself.  So 
> what am I doing wrong?


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[asterisk-users] simultaneous ring problem

[Randall Smith]

I've got this in extensions.conf:

[macro-stdexten]
exten => s,1,Dial(${ARG2},30,p)

exten => 
601555,1,Macro(stdexten,200,SIP/200&SIP/201&SIP/203&SIP/${VOICEPULSE_GATEWAY_OUT_A}/+1504555)

Where the real numbers have been replaced with 555.  What I'm trying 
to do is ring my cell phone in addition to the local extensions. Funny 
thing is the cell phone rings and I can press 1 to connect the call, but 
if I remove the 'p' from the Dial command the cell phone rings, but when 
I answer it the call doesn't connect.  I'm just talking to myself.  So 
what am I doing wrong?

Randall


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[asterisk-users] forward call intended for another domain

[Randall Smith]

I'm new here.  For a registered SIP 'friend' that dials an address not 
handled by my server (say [EMAIL PROTECTED]), how do I get Asterisk to 
forward that call to ekiga.net?

Thanks.

Randall


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[asterisk-users] Ever donate Software to Digium? If you did your a fool.

[Randall H.]

If you gave software to Digium then you helped Mark become very rich.

http://abcnews.go.com/Technology/wireStory?id=2290152
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[asterisk-users] Inc.com Names Mark Spencer o f Digium to its “30 Under 30: America’s Coo lest Young Entrepreneurs”

[Randall H.]

Congrats Mark !
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Re: [asterisk-users] FW: $3,000 server

[Randall H.]

Both of you are a huge pain in the ass. Just go do something so you
can make money and pay your bills. The Jeremy NuFone roadshow is back
in reruns again. Damn.

On 7/12/06, Alex Robar <[EMAIL PROTECTED]> wrote:

His work ethic is fine... He either couldn't do the job you wanted, or he
didn't want to, so he sent you to someone else. If you paid Greg instead of
NuFone, then that's really tough shit for you for not following the
instructions Jeremy gave you. Furthermore, Jeremy gets to wash his hands of
the issue because you never paid him... If I hire a tech to go out to your
site and install a server, but you pay the tech instead of my company, then
you're dealing with the tech, not me. If something breaks, you don't get to


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RE: [Asterisk-Users] Echo cancellation over satellite link

[Randall Prentice]

Title: Message



2 
Things...  You probably know anyway...
 
Early 
echo cancel for satellite just used half duplex switching as a way to get around 
the echo (This led to the echo suppess tone being used for faxes and modems to 
allow full duplex).
 
From 
my Telephony days the echo comes mostly from a mismatch in the hybrid at the 
other end of the link from where the echo is heard.
 
Regards
Randall Prentice

  
  Just wondering, is the echo canceller in the TE411P capable of 
  cancelling the echo caused by the delay over satellite link (i.e. approx 
  400 ms delay)? 
   
  Does anyone have any success story to share? 
   
  I'm kinda stuck with a really2 annoying echo... adjusting the gain didn't 
  help... and what should my zapata.conf look like for effective echo 
  cancellation?
   
  Thanks in advance ^_^
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[Asterisk-Users] list administrator.....???

[Randall Shimizu]

I keep recieving multiple digests per day. Need to find out if there is a way 
to limit the number digests that are being sent to me. Tried contacting the 
list administrator, but I have not recieved a response. Does anyone have a 
alternate email for him...???
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[Asterisk-Users] Asterisk MTBF studies

[Randall Shimizu]

I was wondering if anyone has seen any MTBF studies on Asterisk. Since Asterisk 
is a softswitch there is reason to be concerned about uptime. Some people have 
mentioned using a T-1 switch. Is there a DSL failover using DSL...??? 
Alternatively speaking one could use a product like Vmware ESX to failover 
posssibly.
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[Asterisk-Users] OT: list digest: receiving multiple digests per day....????

[Randall Shimizu]

OT: list digest: receiving multiple digests per day

I selected a single digest to be sent daily to my email account. But for some reason I 
keep recieving multiple digests?? Is there a way to force mailman to send me on 
digest per day and several per day??

Thanks

Randy
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[Asterisk-Users] help

[Randall Shimizu]

help
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[Asterisk-Users] Asterisk fault tolerance and a embedded hardware solution.....??

[Randall Shimizu]

Hello,

IMHO you have a problem with the hardware that Asterisk runs on.
You should really look around because there are a number of companies
selling intel based systems with a cPCI bus fully hot swap capable.
I think the only problem would be getting network adapters compatible with
* but then this is only a problem of drivers easily solved by a good
programmer.

If you test out Asterisk on a fully redundant box and you find problems I
think you'd be welcome to send in a patch to fix them so that * could
be used in "enterprise computing" instead of sending in a two page e-mail
with the problems we all know about !

Ok Sorry, I if the email was a bit provocative. I was just trying to get some 
suggestions & thoughts about hardware and fault tolerance with Asterisk. 

Regards

Kiss Karoly

On Thu, 4 Mar 2004, Randall Shimizu wrote:

> Asterisk fault tolerance and a embedded hardware solution.??
>
> Has anyoone tried implement Asterisk as a hardware based solution similar to Soekris 
> firewall?
>
>
> Asterisk & fault tolerance: I ran across this posting about Asterisk
> and here is some interesting thoughts to ponder
>
>

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[Asterisk-Users] Asterisk fault tolerance and a embedded hardware solution.....??

[Randall Shimizu]

Asterisk fault tolerance and a embedded hardware solution.??

Has anyoone tried implement Asterisk as a hardware based solution similar to Soekris 
firewall?


Asterisk & fault tolerance: I ran across this posting about Asterisk
and here is some interesting thoughts to ponder


http://groups.google.com/groups?hl=en&lr=&ie=UTF-8&oe=UTF-8&selm=aca5dd1d9141c07addd9d3414e934380%40free.teranews.com&rnum=14


Not blow anyone's ASTERISK bubble BUT,,,

"Show me an Asterisk system that can:

1) Have a communication bus that can survive the removal of the CPU,
and
still have calls in progress that remain active until the calling
parties
hang up.

Difficult problem to solve. One would have to have some sort of
parallel network connection. Perhaps one could have a buffering or
cache solution.

The CPU problem could be solved by a blade server or failover.

2) I have yet to hear of any Asterisk box running a fully redundant
CPU
configuration. I bet this is possible.  Especially with the newer hot
swap
cPCI bus systems and slave CPU cards.  Even better if the chassis has
and
embedded H.110, or equivalent in LAN/memory, switching bus.

Yes could be solved.

3) A redundant configuration where either CPU can talk to the
communications
boards (T1/E1), and LAN interfaces.  And which can address all boards
in the
system redundantly.

Sounds like a job for Infiniband or a platform that has a switched
crossbar architecture like IBM P-Series or Sun.

4) A redundant configuration that has either shared system memory
between
the CPU's, or at least table copies between memory that hold all
static and
dynamic call information.

5) A redundant configuration that can swap between system CPU's in
less than
20 seconds.

6) A redundant configuration that can synchronize on, and share one,
two ,
and more network clocking signals.  Plus synchronize on a independent
stratum 3 or greater clock source.

7) And can support 1,000 or more endpoints (TDM and/or IP) without
choking
on it's own guts.

8) A redundant configuration that can synchronize on, and share one,
two ,
and more network clocking signals."

Well it's a lot to ask, but enterprise computing demands a lot.


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RE: [Asterisk-Users] Dell 1750 server and Asteriks...

[Robert R. Randall]

Thanks!  Any idea how many ports it can support as a gateway?

I was hoping it could support 2 4 port cards.
If configured with 2 CPUs and enough ram.

It has 2 PCI buses - one for each card.

Is this just too many ports for the CPUs to handle the 
encode/decode task?

Be gentle with me, I'm still learning ;-)

Robert
 
> -Original Message-
> From: Pertti Pikkarainen [mailto:[EMAIL PROTECTED] 
> Sent: Wednesday, February 18, 2004 2:23 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Dell 1750 server and Asteriks...
> 
> 
> Yes,
> works ok with TE410P and E400P.
> The server has both slot types.
> 
> 
> -- Pertti
> 
> 
> Robert R. Randall wrote:
> 
> >  
> > Has anyone tried the Dell 1750 server as an Asterisks 
> server with one 
> > of the 4 port Digium cards?
> > I'm just looking for a reference point on this.  Thanks.
> >  
> >
> > Robert
> >
> 
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[Asterisk-Users] Dell 1750 server and Asteriks...

[Robert R. Randall]

 
Has anyone tried the Dell 1750 server as an Asterisks 
server with one of the 4 port Digium cards?
I'm just looking for a reference point on this.  
Thanks.
 

Robert