Re: [asterisk-users] ISDN BRI vs SIP Trunks over EDIA

[Doug Lytle]

>>> On Apr 27, 2016, at 12:12 PM, Mark Engelhardt 
>>> ma...@intuitiveengineering.com wrote:


>>> 1) Old School ISDN BRI lines which I would connect to Asterisk with a 
>>> OpenVOX B200P


I've never dealt with a BRI before, primarily PRI, but I'd go BRI instead of IP 
if they're doing any faxing.

Doug
 





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Re: [asterisk-users] ISDN Cause Code 47 Errors

[Dale Noll]

Thanks Richard and Andres.

I had come to the same conclusion, however the provider was fairly snarky
in saying is was my equipment.

We were able to replace the Cisco 2800 with a Cisco 2900 series and the
problem appears to have been resolved.

Thanks again, I always appreciate another set of eyes just in case I missed
something.

Dale



On Wed, Jan 22, 2014 at 11:57 AM, Andres  wrote:

>
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 < TEI=0 Call Ref: len= 2 (reference 6918/0x1B06) (Sent from
>> originator)
>> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 < Message Type: RELEASE COMPLETE (90)
>> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 < [08 02 80 af]
>> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare:
>> 0  Location: User (0)
>> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 <  Ext: 1  Cause: Resource unavailable, unspecified
>> (47), class = Network Congestion (resource unavailable) (2) ]
>>
>   My guess is your provider did not have a free voice channel to pass
> audio at some leg in the call.  There could be multiple legs in the call
> and one of them had 'Network Congestion'.
>
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 Received message for call 0xb750d598 on link 0x9b33f2c TEI/SAPI 0/0
>> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 -- Processing IE 8 (cs0, Cause)
>> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI
>> Span: 4 q931.c:8567 post_handle_q931_message: Call 6918 enters state 0
>> (Null).  Hold state: Idle
>> [Jan 14 12:56:04] VERBOSE[13262] sig_pri.c: [Jan 14 12:56:04] Span 4:
>> Processing event PRI_EVENT_HANGUP
>> [Jan 14 12:56:04] VERBOSE[13262] sig_pri.c: [Jan 14 12:56:04] -- Span
>> 4: Channel 0/2 got hangup, cause 47
>>
>
>  Richard
>
>
>
>
>
> --
> Technical Supporthttp://www.cellroute.net
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Re: [asterisk-users] ISDN Cause Code 47 Errors

[Andres]

[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 < TEI=0 Call Ref: len= 2 (reference 6918/0x1B06) (Sent
from originator)
[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 < Message Type: RELEASE COMPLETE (90)
[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 < [08 02 80 af]
[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU)
standard (0)  Spare: 0  Location: User (0)
[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 <  Ext: 1  Cause: Resource unavailable, unspecified
(47), class = Network Congestion (resource unavailable) (2) ]

My guess is your provider did not have a free voice channel to pass 
audio at some leg in the call.  There could be multiple legs in the call 
and one of them had 'Network Congestion'.


[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 Received message for call 0xb750d598 on link 0x9b33f2c
TEI/SAPI 0/0
[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 -- Processing IE 8 (cs0, Cause)
[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 q931.c:8567 post_handle_q931_message: Call 6918 enters
state 0 (Null).  Hold state: Idle
[Jan 14 12:56:04] VERBOSE[13262] sig_pri.c: [Jan 14 12:56:04] Span
4: Processing event PRI_EVENT_HANGUP
[Jan 14 12:56:04] VERBOSE[13262] sig_pri.c: [Jan 14 12:56:04]
-- Span 4: Channel 0/2 got hangup, cause 47



Richard






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Re: [asterisk-users] ISDN Cause Code 47 Errors

[Richard Mudgett]

On Mon, Jan 20, 2014 at 5:16 PM, Dale Noll  wrote:

> We fairly recently switched service providers for our 4 PRI circuits.
> Since that time, we started to notice some failed inbound calls. These
> calls terminate with an ISDN cause code 47 'resource unavailble'. Most of
> the time I see this error on the first or second channel on the second span
> in a trunk group (This is the providers trunk group for hunting, not an
> Asterisk trunk group).  All the PRIs are setup as individual spans, we are
> not using NFAS. If the provider sets the hunt method to 'least recently
> used channel', then I can receive calls on other channels on the secondary
> span, it is just the first 2 that consistently fail.
>
> We have had occasion where the error occurs on first span. If enough calls
> come in at the same time, callers who happen to land on channel 3 or above
> are OK. When the problem happens on the first span, if we physically
> disconnect the first span(RED alarm), the calls hunt to the second span and
> all calls seem to process properly. The only way to clear the cause 47
> errors from the first span is a power cycle on the provider equipment.
> Power cycling my equipment does not solve the problem, only when I cycle
> their equipment.
>
> The provider says the cause 47 is coming from my equipment, yet the 'core
> set debug on' log, unless I am reading it wrong, says it is coming from
> their side.
>
> I have a second server as a backup. Both servers have identical hardware
> and software. When switching to the backup server, the problem remains.
>
> I had the same setup on the previous provider, except using NFAS, and did
> not have this problem.
>
> Am I reading the log correctly?
>

Yes.  Asterisk has accepted the selected channel and CONNECTed the
call.  It is the peer that is disconnecting the call with cause 47.  This
really
appears to be a problem in the providers equipment.


> Do I have something setup incorrectly?
>

I don't see anything wrong.


> Is there any way to get even lower level debugging on the PRI?
>

Not really.  The problem is at layer 3 (Q.931) not layer 2 (Q.921).
Turning on
intense PRI debug will add nothing but noise to the debug log.



===
> Log sample with ISDN debug on
> ===
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 < Protocol Discriminator: Q.931 (8)  len=87
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 < TEI=0 Call Ref: len= 2 (reference 6918/0x1B06) (Sent from originator)
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 < Message Type: SETUP (5)
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 < [04 03 80 90 a2]
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 < Bearer Capability (len= 5) [ Ext: 1  Coding-Std: 0  Info transfer
> capability: Speech (0)
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 <  Ext: 1  Trans mode/rate: 64kbps,
> circuit-mode (16)
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4  [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 < [18 03 a9 83 82]
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 < Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0
>  Exclusive  Dchan: 0
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 <   ChanSel: As indicated in following octets
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 <   Ext: 1  Coding: 0  Number Specified  Channel
> Type: 3
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 <   Ext: 1  Channel: 2 Type: CPE]
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 < [1c 1d 9f 8b 01 00 a1 17 02 01 1f 02 01 00 80 0f 43 65 6c 6c 20 50 68
> 6f 6e 65 20 20 20 57 49]
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 < Facility (len=31, codeset=0) [ 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x17,
> 0x02, 0x01, 0x1F, 0x02, 0x01, 0x00, 0x80, 0x0F, 'Cell Phone   WI' ]
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 < [28 0f 43 65 6c 6c 20 50 68 6f 6e 65 20 20 20 57 49]
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 < Display (len=15) [ Cell Phone   WI ]
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 < [6c 0c 21 80 34 31 34 33 33 31 32 34 37 30]
> [Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04] PRI Span:
> 4 < Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI:
> ISDN/Teleph

Re: [asterisk-users] ISDN outgoing caller id

[Hans Witvliet]

-Original Message-
From: Gergo Csibra 
Reply-to: Gergo Csibra , Asterisk Users Mailing List -
Non-Commercial Discussion 
To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] ISDN outgoing caller id
Date: Tue, 27 Aug 2013 21:28:36 +0200

Tuesday, August 27, 2013, 8:41:18 PM, Patrick wrote:

> On 08/27/2013 08:04 PM, Gergo Csibra wrote:
>> Hi,
>>
>> is anybody out there who can set the outgoing caller id on ISDN (CAPI
>> or misdn) channels? I've tryed everything what I found in forums, os
>> voip-info.com but no luck. I use a fritz card with CAPI in my first
>> installation (1 BRI), and a hfc 4 port bri card with misdn on other.
>> The first installation have p-t-mp configuration, the second one is
>> p-t-p. Both configuration is EuroISDN in Hungary.
>>
>> So, can anybody help me?

> Have you checked with your Telco if they allow you to change the 
> callerid? If yes, are you setting the callerid to a number that you are 
> allowed to use? You can't just set callerid to any number you like. You 
> must "own" the number which you want to set callerid to. I have no 
> problem setting the callerid on outgoing calls via chan_capi to one of 
> the numbers that the telco assigned to me.

Yes, of course I want to set our assigned numbers, becuse the called
party sees "Unknown" now.
-Original Message-
It's been a while ago for me, but:

Besides the item mentioned above (hit that one also) two things come to mind..
1) is CLI-Display activated on that line? For some telco's it is a fascility 
that has to be enabled..
(you can check it by plugging in a isdn-handset, and try to make a call)

2) Perhaps accidentally activated the "HIDE CLI" activated?


hw

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Re: [asterisk-users] ISDN outgoing caller id

[Chad Wallace]

On Tue, 27 Aug 2013 21:28:36 +0200
Gergo Csibra  wrote:

> Tuesday, August 27, 2013, 8:41:18 PM, Patrick wrote:
> 
> > On 08/27/2013 08:04 PM, Gergo Csibra wrote:
> >> Hi,
> >>
> >> is anybody out there who can set the outgoing caller id on ISDN
> >> (CAPI or misdn) channels? I've tryed everything what I found in
> >> forums, os voip-info.com but no luck. I use a fritz card with CAPI
> >> in my first installation (1 BRI), and a hfc 4 port bri card with
> >> misdn on other. The first installation have p-t-mp configuration,
> >> the second one is p-t-p. Both configuration is EuroISDN in Hungary.
> >>
> >> So, can anybody help me?
> 
> > Have you checked with your Telco if they allow you to change the 
> > callerid? If yes, are you setting the callerid to a number that you
> > are allowed to use? You can't just set callerid to any number you
> > like. You must "own" the number which you want to set callerid to.
> > I have no problem setting the callerid on outgoing calls via
> > chan_capi to one of the numbers that the telco assigned to me.
> 
> Yes, of course I want to set our assigned numbers, becuse the called
> party sees "Unknown" now.

First you need to discuss with your telco whether you can do this on
your lines.  They may not allow it at all, but if they don't, they
should at least set it to something other than "Unknown" for you.  You
need to talk to them.

Then, if they do allow you to set it, and it still doesn't work for you,
please post your code to the list so we can see where you may have gone
wrong.


-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0


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Re: [asterisk-users] ISDN outgoing caller id

[Gergo Csibra]

Tuesday, August 27, 2013, 8:41:18 PM, Patrick wrote:

> On 08/27/2013 08:04 PM, Gergo Csibra wrote:
>> Hi,
>>
>> is anybody out there who can set the outgoing caller id on ISDN (CAPI
>> or misdn) channels? I've tryed everything what I found in forums, os
>> voip-info.com but no luck. I use a fritz card with CAPI in my first
>> installation (1 BRI), and a hfc 4 port bri card with misdn on other.
>> The first installation have p-t-mp configuration, the second one is
>> p-t-p. Both configuration is EuroISDN in Hungary.
>>
>> So, can anybody help me?

> Have you checked with your Telco if they allow you to change the 
> callerid? If yes, are you setting the callerid to a number that you are 
> allowed to use? You can't just set callerid to any number you like. You 
> must "own" the number which you want to set callerid to. I have no 
> problem setting the callerid on outgoing calls via chan_capi to one of 
> the numbers that the telco assigned to me.

Yes, of course I want to set our assigned numbers, becuse the called
party sees "Unknown" now.

-- 
Best regards,
 Gergomailto:csi...@gmail.com


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Re: [asterisk-users] ISDN outgoing caller id

[Patrick Lists]

On 08/27/2013 08:04 PM, Gergo Csibra wrote:

Hi,

is anybody out there who can set the outgoing caller id on ISDN (CAPI
or misdn) channels? I've tryed everything what I found in forums, os
voip-info.com but no luck. I use a fritz card with CAPI in my first
installation (1 BRI), and a hfc 4 port bri card with misdn on other.
The first installation have p-t-mp configuration, the second one is
p-t-p. Both configuration is EuroISDN in Hungary.

So, can anybody help me?


Have you checked with your Telco if they allow you to change the 
callerid? If yes, are you setting the callerid to a number that you are 
allowed to use? You can't just set callerid to any number you like. You 
must "own" the number which you want to set callerid to. I have no 
problem setting the callerid on outgoing calls via chan_capi to one of 
the numbers that the telco assigned to me.


Regards,
Patrick



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Re: [asterisk-users] ISDN- E1 PRI module in network side signaling

[Dimitar Dimitrov]

Thank you guys for the fast response.
I will try that.

Thanks.
Dimitar

On 03/31/2013 11:15 AM, Tony Mountifield wrote:

In article ,
Mitul Limbani  wrote:

On Mar 31, 2013 12:25 PM, "Dimitar Dimitrov"  wrote:


  Hello everyone.
I am looking for a E1 PRI card which supports network side signaling not
CPE. The main idea is to connect an plain old E1 compliant PBX which
doesn't have an VoIP module to the newly created VoIP infrastructure.
Could we use a Digium TE122P or something other to resolve this situation?

Use pri_net as signalling mode inside chan_dadhi.conf in /etc/asterisk/
folder.

You can set this up using any pri card thats supported on Asterisk.

And you may need to make an E1 crossover cable. These are different from
Ethernet crossover cables. You need to cross pair 1-2 with pair 4-5.

Cheers
Tony



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Re: [asterisk-users] ISDN- E1 PRI module in network side signaling

[Tony Mountifield]

In article ,
Mitul Limbani  wrote:
> On Mar 31, 2013 12:25 PM, "Dimitar Dimitrov"  wrote:
> 
> >  Hello everyone.
> > I am looking for a E1 PRI card which supports network side signaling not
> > CPE. The main idea is to connect an plain old E1 compliant PBX which
> > doesn't have an VoIP module to the newly created VoIP infrastructure.
> > Could we use a Digium TE122P or something other to resolve this situation?
> 
> Use pri_net as signalling mode inside chan_dadhi.conf in /etc/asterisk/
> folder.
> 
> You can set this up using any pri card thats supported on Asterisk.

And you may need to make an E1 crossover cable. These are different from
Ethernet crossover cables. You need to cross pair 1-2 with pair 4-5.

Cheers
Tony
-- 
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] ISDN- E1 PRI module in network side signaling

[Mitul Limbani]

Use pri_net as signalling mode inside chan_dadhi.conf in /etc/asterisk/
folder.

You can set this up using any pri card thats supported on Asterisk.

Mitul
On Mar 31, 2013 12:25 PM, "Dimitar Dimitrov"  wrote:

>  Hello everyone.
> I am looking for a E1 PRI card which supports network side signaling not
> CPE. The main idea is to connect an plain old E1 compliant PBX which
> doesn't have an VoIP module to the newly created VoIP infrastructure.
> Could we use a Digium TE122P or something other to resolve this situation?
>
> Thanks in advance.
> Dimitar
>
>
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Re: [asterisk-users] ISDN E1, after electrical disconnected is not becoming UP, IRQ misses: 1

[Shaun Ruffell]

On Tue, Jan 03, 2012 at 02:41:33PM -0800, bilal ghayyad wrote:
> Dear All;
> 
> I am afraid from IRQ misses: 1
> 
> The ISDN E1 was working fine on the machine, the electrical
> disconnected and then the Red Allarm. I checked the dahdi and I
> found that I have to reinstall dahdi again and I did. But still
> not becoming UP.
> 
> The output of the cat /proc/dahdi/1 is following (I am afraid from
> the IRQ misses: 1, so if it is a problem what is the solution)?
> 
> [root@CC asterisk]# cat /proc/dahdi/1
> Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS/CRC4 RED
> IRQ misses: 1

If it's just one IRQ miss, I don't think that by itself is anything to
worry about, although I would make sure you don't have a frame buffer,
slow serial console, etc.. configured on this system.

I am not sure why you would have to reinstall DAHDI after
disconnecting the electricity unless the disk was corrupted because
it was just installed without syncing to the disk.

I would make sure that that you can put the span in loopback and run
patlooptest: ie.

 /usr/src/dahdi-tools# dahdi_maint -s 1 --loopback localhost
 /usr/src/dahdi-tools# ./patlooptest /dev/dahdi/1 -t 10
 Using Timeout of 10 Seconds
 Going for it...
 Timeout achieved Ending Program
 Test ran 33 loops of 2039 bytes/loop with 0 errors
 /usr/src/dahdi-tools# dahdi_maint -s 1 --loopback off
 Span 1: loopback OFF

If that works you will most likely need to investigate with your
provider. They may have to reset things on their end.

-- 
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Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] ISDN PRI configuration

[Steve Edwards]

My favorite pinout.

http://org.against.org/how-to-create-a-ethernet-crossover-cable/

It's for an Ethernet crossover, but it does make making your own cables 
more enjoyable.


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Re: [asterisk-users] ISDN PRI configuration

[giovanni.v]

Il 08/12/2011 18.55, Lyle Giese ha scritto:


And one shield around all the pairs is not the same as ABAM.


I agree, if you run away from the demarc (NT) shielding make a great 
difference in crosstalk, also greater conductor size improve attenuation 
so a full specificaton cable must be used.
But if you are near the NT a 3 meters unshielded ethernet cable fits the 
job without issues.


A little biased but politically correct document:
- http://www.quabbin.com/page2027.html

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Re: [asterisk-users] ISDN PRI configuration

[Vieri]

Interesting:
"If you cannot obtain T1 specific cable, then use two runs of CAT 5.  Use one 
CAT5 cable for the Transmit (Tx) signal and one CAT5 cable for the Receive (Rx) 
signal.  It is necessary for the Tx and Rx signals to be in separate sheaths to 
prevent cross talk interference"

So pins 1 and 2 on one cable and pins 4 and 5 on another.


--- On Thu, 12/8/11, Lyle Giese  wrote:

> Try this instead:
> 
> http://www.ahk.com/t1_cable.html
> 
> That cisco link does not specify the cable itself, but only
> the pin 
> outs.  True T1 cable has a foil shield around each
> pair, also called 
> ABAM cable in the telco world.
> 
> Ethernet cable is twisted pair without any shielding
> between pairs.
> 
> And one shield around all the pairs is not the same as
> ABAM.
> 
> Lyle Giese
> LCR Computer Services, Inc.
> 
> On 12/08/11 10:53, Carlos Alvarez wrote:
> > A T1 cable according to this spec:
> >
> > http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml
> >
> > Crossing the 1/2 to 4/5 if needed.
> >
> >
> > On Thu, Dec 8, 2011 at 9:37 AM, Olivier  > >
> wrote:
> >
> >     2011/12/8, Carlos Alvarez
>  >     >:
> >      > I am not Kevin, but I'll tell
> you that I will not EVER use an
> >     Ethernet
> >      > cable for T1 again. 
> Kevin and I have discussed this at length,
> >     and the
> >      > "should work" plays out
> poorly in the real world, or at least
> >     mine.  I've
> >      > had it be fine, and had major
> problems.  I can't even find a
> >     pattern to it,
> >      > like length of cable.
> >      >
> >      > In a colo cabinet that was
> direct-connected to a carrier, it
> >     worked great
> >      > for years and then one
> day...no T1.  Just gone.  Go down there
> >     and put in a
> >      > real T1 cable, came right up,
> still up years later.
> >      >
> >      > I usually make my own,
> >
> >     which type of cable are you
> then using ?
> >
> >
> >      > since they are so expensive
> to buy.  I just connect
> >      > the four needed pins, pretty
> easy to do if you're not trying to
> >     stuff all
> >      > eight wires into the
> connector.
> >      >
> >      >
> >      >
> >      > On Thu, Dec 8, 2011 at 5:57
> AM, Tony Mountifield
> >      >
> wrote:
> >      >
> >      >> In article <4ee0b0e2.3050...@digium.com
> >     >,
> >      >> Kevin P. Fleming  >     >
> wrote:
> >      >> >
> >      >> > As I said before...
> an Ethernet cable will work nearly all the
> >     time, and
> >      >> > at a 5m length it's
> probably fine.
> >      >>
> >      >> Kevin, under what
> circumstances would an Ethernet cable
> >     potentially not
> >      >> work with T1/E1? And in
> those circumstances, what should be used
> >     instead?
> >      >> I'm wondering because I
> had never realised it was an issue until
> >     you said.
> >      >>
> >      >> Cheers
> >      >> Tony
> >      >> --
> >      >> Tony Mountifield
> >      >> Work: t...@softins.co.uk
> 
> -
> >     http://www.softins.co.uk
> >      >> Play: t...@mountifield.org
> 
> -
> >     http://tony.mountifield.org
> >      >>
> >      >> --
> >      >>
> > 
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> Colocation Provided by
> >     http://www.api-digital.com --
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> >      >>
> >      >
> >      >
> >      >
> >      > --
> >      > Carlos Alvarez
> >      > TelEvolve
> >      > 602-889-3003
> 
> >      >
> >
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Re: [asterisk-users] ISDN PRI configuration

[Lyle Giese]

Try this instead:

http://www.ahk.com/t1_cable.html

That cisco link does not specify the cable itself, but only the pin 
outs.  True T1 cable has a foil shield around each pair, also called 
ABAM cable in the telco world.


Ethernet cable is twisted pair without any shielding between pairs.

And one shield around all the pairs is not the same as ABAM.

Lyle Giese
LCR Computer Services, Inc.

On 12/08/11 10:53, Carlos Alvarez wrote:

A T1 cable according to this spec:

http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml

Crossing the 1/2 to 4/5 if needed.


On Thu, Dec 8, 2011 at 9:37 AM, Olivier mailto:oza_4...@yahoo.fr>> wrote:

2011/12/8, Carlos Alvarez mailto:car...@televolve.com>>:
 > I am not Kevin, but I'll tell you that I will not EVER use an
Ethernet
 > cable for T1 again.  Kevin and I have discussed this at length,
and the
 > "should work" plays out poorly in the real world, or at least
mine.  I've
 > had it be fine, and had major problems.  I can't even find a
pattern to it,
 > like length of cable.
 >
 > In a colo cabinet that was direct-connected to a carrier, it
worked great
 > for years and then one day...no T1.  Just gone.  Go down there
and put in a
 > real T1 cable, came right up, still up years later.
 >
 > I usually make my own,

which type of cable are you then using ?


 > since they are so expensive to buy.  I just connect
 > the four needed pins, pretty easy to do if you're not trying to
stuff all
 > eight wires into the connector.
 >
 >
 >
 > On Thu, Dec 8, 2011 at 5:57 AM, Tony Mountifield
mailto:t...@softins.co.uk>> wrote:
 >
 >> In article <4ee0b0e2.3050...@digium.com
>,
 >> Kevin P. Fleming mailto:kpflem...@digium.com>> wrote:
 >> >
 >> > As I said before... an Ethernet cable will work nearly all the
time, and
 >> > at a 5m length it's probably fine.
 >>
 >> Kevin, under what circumstances would an Ethernet cable
potentially not
 >> work with T1/E1? And in those circumstances, what should be used
instead?
 >> I'm wondering because I had never realised it was an issue until
you said.
 >>
 >> Cheers
 >> Tony
 >> --
 >> Tony Mountifield
 >> Work: t...@softins.co.uk  -
http://www.softins.co.uk
 >> Play: t...@mountifield.org  -
http://tony.mountifield.org
 >>
 >> --
 >>
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 >>
 >
 >
 >
 > --
 > Carlos Alvarez
 > TelEvolve
 > 602-889-3003 
 >

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--
Carlos Alvarez
TelEvolve
602-889-3003




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Re: [asterisk-users] ISDN PRI configuration

[Carlos Alvarez]

On Thu, Dec 8, 2011 at 10:48 AM, giovanni.v  wrote:

>
> This is wrong, in both T568A or T568B ethernet pins 1/2 and 4/5 runs on a
> properly twisted pair. Also 120 Ohm impedance is matching the ISDN pri
> specification.
> If a straight pri cable is needed then a straight ethernet cable fits the
> job (not the same for a pri cross cable vs an eth cross cable).
>
>

It was probably the crossover I was thinking of, which is what I almost
always end up needing.  I stopped analyzing the situation when I found
myself simply replacing them with the right cable and being successful.



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TelEvolve
602-889-3003
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Re: [asterisk-users] ISDN PRI configuration

[giovanni.v]

Il 08/12/2011 18.17, Carlos Alvarez ha scritto:

  If you use an ethernet cable, you are using a pair of wires that is
not twisted together, removing the electrical advantage of twisted-pair
cable.


This is wrong, in both T568A or T568B ethernet pins 1/2 and 4/5 runs on 
a properly twisted pair. Also 120 Ohm impedance is matching the ISDN pri 
specification.
If a straight pri cable is needed then a straight ethernet cable fits 
the job (not the same for a pri cross cable vs an eth cross cable).


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Re: [asterisk-users] ISDN PRI configuration

[Carlos Alvarez]

On Thu, Dec 8, 2011 at 10:14 AM, Olivier  wrote:

> In fact I was rather referring to the previous example in which a
> cable did run OK for years and suddenly stopped to.
>

My THEORY is that the driver chips on either end were wearing out and no
longer able to send or receive as well as they once did.  When you run the
correct pairs, the wires are twisted together.  This is important for a
variety of electrical reasons, too lengthy to cover here, but a quick
google search will give you a lot of info if you care.  If you use an
ethernet cable, you are using a pair of wires that is not twisted together,
removing the electrical advantage of twisted-pair cable.


> Obviously, the connector pins were still correctly set.
> If it stopped to work, then it must come from the electric signals and
> should explained through cable impedance or things like that.
>

Yes, exactly.


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Re: [asterisk-users] ISDN PRI configuration

[Carlos Alvarez]

On Thu, Dec 8, 2011 at 9:37 AM, Olivier  wrote:

> >
> > I usually make my own,
>
> which type of cable are you then using ?
>

I just realized that I may have not answered the right question.  Did you
mean what raw cable did I use to make T1 cables?  Cat-3 or above is fine.
 I use whatever I have around, which is typically Cat-5e.  Yes, I know that
solid conductors aren't meant to be pushed into those connectors, yet my
experience is 100% good doing that.


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TelEvolve
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Re: [asterisk-users] ISDN PRI configuration

[Olivier]

2011/12/8, Carlos Alvarez :
> A T1 cable according to this spec:
>
> http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml
>
> Crossing the 1/2 to 4/5 if needed.

In fact I was rather referring to the previous example in which a
cable did run OK for years and suddenly stopped to.

Obviously, the connector pins were still correctly set.
If it stopped to work, then it must come from the electric signals and
should explained through cable impedance or things like that.

My question was rather how could the replacement cable itself be
precisely described  (thickness, shield, category, ...) ?

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Re: [asterisk-users] ISDN PRI configuration

[Dan Austin]

Tony wrote:
>Kevin P. Fleming  wrote:
>> 
>> As I said before... an Ethernet cable will work nearly all the time, and 
>> at a 5m length it's probably fine.

> Kevin, under what circumstances would an Ethernet cable potentially not
> work with T1/E1? And in those circumstances, what should be used instead?
> I'm wondering because I had never realised it was an issue until you said.

I've never had an issue with using Cat5 cable, but I have run into telco/techs
that choose to use a pin out other than 1245, and of course defend it with
'That is our standard way to do it'.  So a standard Ethernet cable would fail,
but once one end was cut off an replaced with the required pin out it would
work fine (but no longer be an Ethernet cable, semantics but important).

Dan

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Re: [asterisk-users] ISDN PRI configuration

[Carlos Alvarez]

A T1 cable according to this spec:

http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml

Crossing the 1/2 to 4/5 if needed.


On Thu, Dec 8, 2011 at 9:37 AM, Olivier  wrote:

> 2011/12/8, Carlos Alvarez :
> > I am not Kevin, but I'll tell you that I will not EVER use an Ethernet
> > cable for T1 again.  Kevin and I have discussed this at length, and the
> > "should work" plays out poorly in the real world, or at least mine.  I've
> > had it be fine, and had major problems.  I can't even find a pattern to
> it,
> > like length of cable.
> >
> > In a colo cabinet that was direct-connected to a carrier, it worked great
> > for years and then one day...no T1.  Just gone.  Go down there and put
> in a
> > real T1 cable, came right up, still up years later.
> >
> > I usually make my own,
>
> which type of cable are you then using ?
>
>
> > since they are so expensive to buy.  I just connect
> > the four needed pins, pretty easy to do if you're not trying to stuff all
> > eight wires into the connector.
> >
> >
> >
> > On Thu, Dec 8, 2011 at 5:57 AM, Tony Mountifield 
> wrote:
> >
> >> In article <4ee0b0e2.3050...@digium.com>,
> >> Kevin P. Fleming  wrote:
> >> >
> >> > As I said before... an Ethernet cable will work nearly all the time,
> and
> >> > at a 5m length it's probably fine.
> >>
> >> Kevin, under what circumstances would an Ethernet cable potentially not
> >> work with T1/E1? And in those circumstances, what should be used
> instead?
> >> I'm wondering because I had never realised it was an issue until you
> said.
> >>
> >> Cheers
> >> Tony
> >> --
> >> Tony Mountifield
> >> Work: t...@softins.co.uk - http://www.softins.co.uk
> >> Play: t...@mountifield.org - http://tony.mountifield.org
> >>
> >> --
> >> _
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> >>
> >
> >
> >
> > --
> > Carlos Alvarez
> > TelEvolve
> > 602-889-3003
> >
>
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Re: [asterisk-users] ISDN PRI configuration

[Olivier]

2011/12/8, Carlos Alvarez :
> I am not Kevin, but I'll tell you that I will not EVER use an Ethernet
> cable for T1 again.  Kevin and I have discussed this at length, and the
> "should work" plays out poorly in the real world, or at least mine.  I've
> had it be fine, and had major problems.  I can't even find a pattern to it,
> like length of cable.
>
> In a colo cabinet that was direct-connected to a carrier, it worked great
> for years and then one day...no T1.  Just gone.  Go down there and put in a
> real T1 cable, came right up, still up years later.
>
> I usually make my own,

which type of cable are you then using ?


> since they are so expensive to buy.  I just connect
> the four needed pins, pretty easy to do if you're not trying to stuff all
> eight wires into the connector.
>
>
>
> On Thu, Dec 8, 2011 at 5:57 AM, Tony Mountifield  wrote:
>
>> In article <4ee0b0e2.3050...@digium.com>,
>> Kevin P. Fleming  wrote:
>> >
>> > As I said before... an Ethernet cable will work nearly all the time, and
>> > at a 5m length it's probably fine.
>>
>> Kevin, under what circumstances would an Ethernet cable potentially not
>> work with T1/E1? And in those circumstances, what should be used instead?
>> I'm wondering because I had never realised it was an issue until you said.
>>
>> Cheers
>> Tony
>> --
>> Tony Mountifield
>> Work: t...@softins.co.uk - http://www.softins.co.uk
>> Play: t...@mountifield.org - http://tony.mountifield.org
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Carlos Alvarez
> TelEvolve
> 602-889-3003
>

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Re: [asterisk-users] ISDN PRI configuration

[Carlos Alvarez]

I am not Kevin, but I'll tell you that I will not EVER use an Ethernet
cable for T1 again.  Kevin and I have discussed this at length, and the
"should work" plays out poorly in the real world, or at least mine.  I've
had it be fine, and had major problems.  I can't even find a pattern to it,
like length of cable.

In a colo cabinet that was direct-connected to a carrier, it worked great
for years and then one day...no T1.  Just gone.  Go down there and put in a
real T1 cable, came right up, still up years later.

I usually make my own, since they are so expensive to buy.  I just connect
the four needed pins, pretty easy to do if you're not trying to stuff all
eight wires into the connector.



On Thu, Dec 8, 2011 at 5:57 AM, Tony Mountifield  wrote:

> In article <4ee0b0e2.3050...@digium.com>,
> Kevin P. Fleming  wrote:
> >
> > As I said before... an Ethernet cable will work nearly all the time, and
> > at a 5m length it's probably fine.
>
> Kevin, under what circumstances would an Ethernet cable potentially not
> work with T1/E1? And in those circumstances, what should be used instead?
> I'm wondering because I had never realised it was an issue until you said.
>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: t...@softins.co.uk - http://www.softins.co.uk
> Play: t...@mountifield.org - http://tony.mountifield.org
>
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Re: [asterisk-users] ISDN PRI configuration

[Tony Mountifield]

In article <4ee0b0e2.3050...@digium.com>,
Kevin P. Fleming  wrote:
> 
> As I said before... an Ethernet cable will work nearly all the time, and 
> at a 5m length it's probably fine.

Kevin, under what circumstances would an Ethernet cable potentially not
work with T1/E1? And in those circumstances, what should be used instead?
I'm wondering because I had never realised it was an issue until you said.

Cheers
Tony
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Re: [asterisk-users] ISDN PRI configuration

[Kevin P. Fleming]

On 12/07/2011 05:06 PM, Vieri wrote:



--- On Wed, 12/7/11, Kevin P. Fleming  wrote:


Standard Ethernet cables do not always work for T-1/E-1
spans. They do work a rather large percentage of the time,
but not always. Distance between the NIU and the T-1/E-1
card can be a factor, among other things.

Many Digium products include span loopback devices, that
you can plug a cable into and generate a hard loopback
towards the card. If there is one of those on-site, have
someone unplug the cable from the NIU and plug it into the
loopback device instead; if the span goes green, then at
least your cabling/wiring are OK.


I bought several Digium products and for the site I'm managing now, there are 
at least these cards:
Wildcard TE120P single-span T1/E1/J1 card (rev 11)


A loopback connector should have been included with this card. It does 
not appear that our web store makes them (the T10i loopback connectors) 
available as individual items, although some distributors may sell them.



ISDN controller: Digium, Inc. Wildcard B410 quad-BRI card
and maybe more but right now I don't recall any "loopback device" although I 
won't be sure until I go to the site.
Can a loopback device be bought seperately?

What kind of cable should be used instead of an ethernet cable (I think they 
used a 5m long cat5 T-568B Straight-Through Ethernet Cable)?


As I said before... an Ethernet cable will work nearly all the time, and 
at a 5m length it's probably fine.


--
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Re: [asterisk-users] ISDN PRI configuration

[giovanni.v]

Il 07/12/2011 23.45, Vieri ha scritto:

As far as I know, E1 usually use 16 as D channel. Anyway, I tried as you 
suggested and set 1 as the D channel and 2-31 as B channels.
In the asterisk log I got these messages:

chan_dahdi.c: Channel 16 is reserved for D-channel.
chan_dahdi.c: Unable to register channel '2-31'

So doesn't this actually tell me that I should keep using 16 as the D channel?


I your line is provisioned as NET5 (ETSI/EuroISDN) you should use 
channel 16 as D channel, no one else.


As suggested from Kevin check your cabling using a loopback, if you 
don't have one make it using an rj45 socket simply wiring pin 1 to pin 4 
ad pin 2 to pin 5.


Also check you telco network termination, a standard one provide TX on 
pins 4/5 and RX on pins 1/2 but sometimes this isn't true (e.g. E1 
gateways).


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Re: [asterisk-users] ISDN PRI configuration

[giovanni.v]

Il 07/12/2011 23.45, Vieri ha scritto:

As far as I know, E1 usually use 16 as D channel. Anyway, I tried as you 
suggested and set 1 as the D channel and 2-31 as B channels.
In the asterisk log I got these messages:

chan_dahdi.c: Channel 16 is reserved for D-channel.
chan_dahdi.c: Unable to register channel '2-31'

So doesn't this actually tell me that I should keep using 16 as the D channel?


I your line is provisioned as NET5 (ETSI/EuroISDN) you should use 
channel 16 as D channel, no one else.


As suggested from Kevin check your cabling using a loopback, if you 
don't have one make it using an rj45 socket simply wiring pin 1 to pin 4 
ad pin 2 to pin 5.


Also check you telco network termination, a standard one provide TX on 
pins 4/5 and RX on pins 1/2 but sometimes this isn't true (e.g. E1 
gateways).


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Re: [asterisk-users] ISDN PRI configuration

[Andres]

and maybe more but right now I don't recall any "loopback device" although I 
won't be sure until I go to the site.
Can a loopback device be bought seperately?
   

Sure, we use the below device all the time:
http://www.amazon.com/Cables-Unlimited-TST-LOOP-003-Smartronix-Superlooper/dp/B000V6Y7IY

What kind of cable should be used instead of an ethernet cable (I think they 
used a 5m long cat5 T-568B Straight-Through Ethernet Cable)?

Thanks,

Vieri



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--
Technical Support
http://www.telesip.net


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Re: [asterisk-users] ISDN PRI configuration

[Vieri]

--- On Wed, 12/7/11, Kevin P. Fleming  wrote:

> Standard Ethernet cables do not always work for T-1/E-1
> spans. They do work a rather large percentage of the time,
> but not always. Distance between the NIU and the T-1/E-1
> card can be a factor, among other things.
> 
> Many Digium products include span loopback devices, that
> you can plug a cable into and generate a hard loopback
> towards the card. If there is one of those on-site, have
> someone unplug the cable from the NIU and plug it into the
> loopback device instead; if the span goes green, then at
> least your cabling/wiring are OK.

I bought several Digium products and for the site I'm managing now, there are 
at least these cards:
Wildcard TE120P single-span T1/E1/J1 card (rev 11)
ISDN controller: Digium, Inc. Wildcard B410 quad-BRI card 
and maybe more but right now I don't recall any "loopback device" although I 
won't be sure until I go to the site.
Can a loopback device be bought seperately?

What kind of cable should be used instead of an ethernet cable (I think they 
used a 5m long cat5 T-568B Straight-Through Ethernet Cable)?

Thanks,

Vieri



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Re: [asterisk-users] ISDN PRI configuration

[Kevin P. Fleming]

On 12/07/2011 04:51 PM, Vieri wrote:



--- On Wed, 12/7/11, Kevin P. Fleming  wrote:


Vieri: You aren't even far enough along to worry about
D-channel assignments or anything like that. Your span is in
RED alarm; that means it can't see the far end at all. Until
you get that cured (layer 1 - physical layer) nothing above
it is going to work.

Since they mentioned HDB3 and CRC4, you most definitely
have an E1 span, and you will need to specify 'CCS' as well
because you are using ISDN signaling. If the line
coding/framing settings are wrong that *could* result in a
RED alarm, but doesn't always.

So, you need to start by getting the span to come out of
RED alarm (to go 'green'). This could be a cabling problem,
a hardware problem, or it could something as simple as the
fact that the telco hasn't actually 'turned up' the span
yet, because they don't usually do that until you have your
equipment plugged in and you call them to tell them that you
are ready for the span to be turned up.


They "should" have turned it up, or at least that's what one of the tech guys 
told me.
But I guess I'll have to check with them again.

The cable should be ok (standard ethernet cable) but I didn't actually install 
it myself (I'm in a remote location) so I'll have to check that too.


Standard Ethernet cables do not always work for T-1/E-1 spans. They do 
work a rather large percentage of the time, but not always. Distance 
between the NIU and the T-1/E-1 card can be a factor, among other things.


Many Digium products include span loopback devices, that you can plug a 
cable into and generate a hard loopback towards the card. If there is 
one of those on-site, have someone unplug the cable from the NIU and 
plug it into the loopback device instead; if the span goes green, then 
at least your cabling/wiring are OK.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] ISDN PRI configuration

[Vieri]

--- On Wed, 12/7/11, Kevin P. Fleming  wrote:

> Vieri: You aren't even far enough along to worry about
> D-channel assignments or anything like that. Your span is in
> RED alarm; that means it can't see the far end at all. Until
> you get that cured (layer 1 - physical layer) nothing above
> it is going to work.
> 
> Since they mentioned HDB3 and CRC4, you most definitely
> have an E1 span, and you will need to specify 'CCS' as well
> because you are using ISDN signaling. If the line
> coding/framing settings are wrong that *could* result in a
> RED alarm, but doesn't always.
> 
> So, you need to start by getting the span to come out of
> RED alarm (to go 'green'). This could be a cabling problem,
> a hardware problem, or it could something as simple as the
> fact that the telco hasn't actually 'turned up' the span
> yet, because they don't usually do that until you have your
> equipment plugged in and you call them to tell them that you
> are ready for the span to be turned up.

They "should" have turned it up, or at least that's what one of the tech guys 
told me.
But I guess I'll have to check with them again.

The cable should be ok (standard ethernet cable) but I didn't actually install 
it myself (I'm in a remote location) so I'll have to check that too.

Big thanks for the explanation!

Vieri



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Re: [asterisk-users] ISDN PRI configuration

[Vieri]

--- On Wed, 12/7/11, Steve Edwards  wrote:

> > A telco has recently installed a new line in our
> building and I need to connect it to my Asterisk server with
> a Digium PRI card.
> > 
> > It's not the first time I set up and configure a PRI
> link but I'm failing to make this one work.
> > 
> > chan_dahdi.c: No D-channels available!  Using
> Primary channel 16 as D-channel anyway!
> 
> We usually get D channels on the first channel of the first
> T1 in an NFAS group and the last channel of the last t1.
> 
> However, telcos don't always get the order right. I've
> spent hours trying configurations and varying the D channel.
> Sometimes it's just that they number things in a different
> order than we were expecting. Sometimes, it almost appears
> that they use a dartboard :)

As far as I know, E1 usually use 16 as D channel. Anyway, I tried as you 
suggested and set 1 as the D channel and 2-31 as B channels.
In the asterisk log I got these messages:

chan_dahdi.c: Channel 16 is reserved for D-channel.
chan_dahdi.c: Unable to register channel '2-31'

So doesn't this actually tell me that I should keep using 16 as the D channel? 
(so chan_dahdi actually knows about it on its own, I guess)

It's funny though that chan_dahdi tells me I have to use channel 16 as D 
channel whenever I try to use another one, but when I do use 16, it says that 
there are no D channels available.

Confusing.

Thanks anyway for the reply.

Vieri


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Re: [asterisk-users] ISDN PRI configuration

[Kevin P. Fleming]

On 12/07/2011 04:15 PM, Steve Edwards wrote:

On Wed, 7 Dec 2011, Vieri wrote:


A telco has recently installed a new line in our building and I need
to connect it to my Asterisk server with a Digium PRI card.

It's not the first time I set up and configure a PRI link but I'm
failing to make this one work.

chan_dahdi.c: No D-channels available! Using Primary channel 16 as
D-channel anyway!


We usually get D channels on the first channel of the first T1 in an
NFAS group and the last channel of the last t1.

However, telcos don't always get the order right. I've spent hours
trying configurations and varying the D channel. Sometimes it's just
that they number things in a different order than we were expecting.
Sometimes, it almost appears that they use a dartboard :)


Vieri: You aren't even far enough along to worry about D-channel 
assignments or anything like that. Your span is in RED alarm; that means 
it can't see the far end at all. Until you get that cured (layer 1 - 
physical layer) nothing above it is going to work.


Since they mentioned HDB3 and CRC4, you most definitely have an E1 span, 
and you will need to specify 'CCS' as well because you are using ISDN 
signaling. If the line coding/framing settings are wrong that *could* 
result in a RED alarm, but doesn't always.


So, you need to start by getting the span to come out of RED alarm (to 
go 'green'). This could be a cabling problem, a hardware problem, or it 
could something as simple as the fact that the telco hasn't actually 
'turned up' the span yet, because they don't usually do that until you 
have your equipment plugged in and you call them to tell them that you 
are ready for the span to be turned up.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] ISDN PRI configuration

[Steve Edwards]

On Wed, 7 Dec 2011, Vieri wrote:

A telco has recently installed a new line in our building and I need to 
connect it to my Asterisk server with a Digium PRI card.


It's not the first time I set up and configure a PRI link but I'm 
failing to make this one work.


chan_dahdi.c: No D-channels available!  Using Primary channel 16 as D-channel 
anyway!


We usually get D channels on the first channel of the first T1 in an NFAS 
group and the last channel of the last t1.


However, telcos don't always get the order right. I've spent hours trying 
configurations and varying the D channel. Sometimes it's just that they 
number things in a different order than we were expecting. Sometimes, it 
almost appears that they use a dartboard :)


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

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Re: [asterisk-users] ISDN-FAX with Asterisk

[Daniel Tryba]

On Thu, Nov 18, 2010 at 10:54:53PM +0100, Thorolf Godawa wrote:
> since some time I am looking for a current and reliable solution to send
> and receive faxes (probably fax-2-mail and mail-2-fax) in conjunction
> with Asterisk.
[snip]
> What are you using? mISDN, CAPI4linux, HylaFAX, IAXmodem, chan_misdn, ... ?

Hylafax/IAXmodem hasn't let me down so far, it works independent of
technology (it only needs alaw/ulaw). Jitter has the ability to kill
the transfers, but that shouldn't be any problem with ISDN.

Just create a bunch of iaxmodems and configure them in hylafax.

For incoming faxes to email I set the callerID name to the emailadress
in the dialplan and in etc/FaxDispatch set SENDTO to "$CIDNAME". For
outgoing faxes from email read the manpage of sendfax (save the
attachment, convert it when necessary, call sendfax with the senders
emailadress so notification get send back to the sender).

-- 

   Daniel Tryba

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Re: [asterisk-users] ISDN-FAX with Asterisk

[William Stillwell (Lists)]

Im using

asterisk-1.6.2.13
asterisk-addons-1.6.2.2
dahdi-linux-complete-2.4.0+2.4.0
libpri-1.4.11.4
spandsp-0.0.6
Sangoma Hardware, using wanpipe-3.5.17

Extensions.conf:

[fax-in]

exten => s,1,Answer()
exten => s,n,Wait(1)
exten =>
s,n,Set(BASEFILE=fax-${CALLERID(dnid)}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
;exten => s,n,Set(${LOCALSTATIONID})
exten => s,n,MixMonitor(/mnt/ramdisk/${BASEFILE}.wav)
exten => s,n,ReceiveFAX(/mnt/ramdisk/${BASEFILE}.tif)
exten => s,n,Hangup()
exten => h,1,System(/home/asterisk/dofax.sh "${EMAILADDRESS}" "${FAXSTATUS}"
"${CALLERID(num)}" "${REMOTESTATIONID}" "${CALLERID(dn$


[inbound-pri]

; 
exten => 00,1,Set(LOCALSTATIONID=${EXTEN})
exten => 00,2,Set(EMAILADDRESS="emailaddress")
exten => 00,3,Goto(fax-in,s,1)



the dofax.sh script checks if tif file exists, converts to pdf, emails, and
then archives on no errors,if missing tiff, or faxstatus <> success, it puts
the fax in a queue folder along with the mix monitor file for analysis. , of
the faxes the fail, you can usually here bad line quality from the sender.


/mnt/ramdisk is a 1gb ramdisk, the dofax script moves the tif/pdf/wavs to a
samba share, and deletes them out of the ramdisk folder.



William Stillwell



> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Thorolf Godawa
> Sent: Thursday, November 18, 2010 4:55 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] ISDN-FAX with Asterisk
> 
> Hi everybody,
> 
> since some time I am looking for a current and reliable solution to
> send
> and receive faxes (probably fax-2-mail and mail-2-fax) in conjunction
> with Asterisk.
> 
> For testing I am using a HFC-ISDN passive PCI-card, in production a
> Digium Dual T1/E1 PCI-card will be used.
> 
> I run CentOS 5.5 (Kernel 2.6.18) and Asterisk 1.4 (but I also can use
> 1.8) but did not find any solution where I think "that's it".
> 
> What are you using? mISDN, CAPI4linux, HylaFAX, IAXmodem, chan_misdn,
> ... ?
> 
> Can you point me to the correct direction, may be there are some more
> or
> less current howto's (more current than the ones from 2007 and earlier
> you find everywhere in the net)?
> 
> Thanks a lot,
> --
> 
> Chau y hasta luego,
> 
> Thorolf
> 
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Re: [asterisk-users] ISDN - Busy signal on 3rd call

[Paulo Santos]

Paulo Santos wrote:
> Hello,
> 
> Following my first mail about this issue [1], I think I know now what
> the problem is.
> 
> When I have both lines being used and a third call comes in, the person
> calling doesn't get a busy tone, he gets something like line unavailable.
> 
> I've been debugging mISDN and I think the reason is because asterisk is
> sending the release cause as 0.
> 
>   P[ 3]  --> channel:0 mode:TE cause:0 ocause:0 rad: cad:
> 
> The request from the telephone company's switch seems correct, a SETUP
> message (if 08 is Q.931, 05 is SETUP).
> 
>   02 ff 03 08  01 04 05 a1  04 03 80 90
>   a3 18 01 80  6c 0b 01 83  39 31 36 33
>   39 31 37 34  32 70 03 c1  38 34
> 
> I've changed misdn.conf so it sends a release cause as 17 (user busy),
> but I get the same behaviour - cause:0 ocause:0.
> 
> Anyone knows how can I force asterisk to send cause 16 or 17 in this
> situation?
> 
> Thanks in advance.
> 
> Best regards,
> Paulo Santos
> 
> misdn.conf: http://pastebin.com/FmgECqkU
> misdn debug: http://pastebin.com/Tg6wPKBD
> 
> [1]
> http://www.mail-archive.com/asterisk-users@lists.digium.com/msg244330.html
> 

Ok, I've encountered a similar issue on a different installation but
instead of being PTP it's PTMP. Plus, it's a setup with 2 BRI lines with
call forwarding between them - main number of BRI1 forwards to secondary
number of BRI2 when busy/unavailable and vice-versa.

I've called the phone company and confirmed that call waiting is
disabled, yet I get a message in misdn debug saying:

P[ 2]  --> Call Waiting on PMP sending RELEASE_COMPLETE

I don't know if this is the actual call waiting feature or if it is just
an information of some kind.

In the misdn debug I get this: http://pastebin.com/D7wv0qqm

The P[ 2] is the port of the BRI line I called in the first place, then
it is forwarded to P[ 1] where I get an error:

P[ 1] Decoding FACILITY failed! (-1)

And the same issue I said in the previews email:

P[ 1]  --> channel:0 mode:TE cause:0 ocause:0 rad: cad:

I changed isdn_lib.c and now I'm sending ocause:17 (user busy). I've
done this in the PTP line mentioned in the previews email as well.

For the PTP line it appears to have worked, I have the regular busy
signal. It worked only after the first time I tried to place a 3rd call.
Now the 3rd call doesn't even reach Asterisk, which was what I wanted
from the phone company in the first place.

On the PTMP line it didn't work, I still don't get the busy signal.

Maybe cause 17 isn't the right one? And what can be that "FACILITY"
mentioned in the debug?

Thanks in advance.

Best regards,
Paulo Santos

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Re: [asterisk-users] ISDN & SS7

[huu giang]

I'm planning to use SGM with Asterisk, it is a commercial product.
What is the different between SGM and libs77 and chan_ss7  ? Should I use SGM ?





From: Tzafrir Cohen 
To: asterisk-users@lists.digium.com
Sent: Tue, October 26, 2010 3:12:21 AM
Subject: Re: [asterisk-users] ISDN & SS7

On Sun, Oct 24, 2010 at 11:33:28AM -0500, Cary Fitch wrote:
> SS7 is an inter-telco system using a separate network for all signaling.
> 
>  
> 
> You must have an SS7 network connection before anything will work.
> 
>  
> 
> Then the T1 Spans run 24 64k audio paths.  The SS7 net exchanges the call
> data and connection info between the switches.
> 
>  
> 
> Asterisk doesn't support SS7 natively although I believe there are one or
> more add-on vendors.

The vendors of addons such as http://svn.asterisk.org/svn/libss7 and
http://svn.asterisk.org/svn/asterisk/trunk/channels/sig_ss7.c ?

-- 
              Tzafrir Cohen
icq#16849755              jabber:tzafrir.co...@xorcom.com
+972-50-7952406          mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] ISDN & SS7

[huu giang]

Are these solutions reliable and stable ?.
Have you used these solutions in production ? What about its quality ?





From: Tzafrir Cohen 
To: asterisk-users@lists.digium.com
Sent: Tue, October 26, 2010 3:12:21 AM
Subject: Re: [asterisk-users] ISDN & SS7

On Sun, Oct 24, 2010 at 11:33:28AM -0500, Cary Fitch wrote:
> SS7 is an inter-telco system using a separate network for all signaling.
> 
>  
> 
> You must have an SS7 network connection before anything will work.
> 
>  
> 
> Then the T1 Spans run 24 64k audio paths.  The SS7 net exchanges the call
> data and connection info between the switches.
> 
>  
> 
> Asterisk doesn't support SS7 natively although I believe there are one or
> more add-on vendors.

The vendors of addons such as http://svn.asterisk.org/svn/libss7 and
http://svn.asterisk.org/svn/asterisk/trunk/channels/sig_ss7.c ?

-- 
              Tzafrir Cohen
icq#16849755              jabber:tzafrir.co...@xorcom.com
+972-50-7952406          mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] ISDN & SS7

[Tzafrir Cohen]

On Sun, Oct 24, 2010 at 11:33:28AM -0500, Cary Fitch wrote:
> SS7 is an inter-telco system using a separate network for all signaling.
> 
>  
> 
> You must have an SS7 network connection before anything will work.
> 
>  
> 
> Then the T1 Spans run 24 64k audio paths.  The SS7 net exchanges the call
> data and connection info between the switches.
> 
>  
> 
> Asterisk doesn't support SS7 natively although I believe there are one or
> more add-on vendors.

The vendors of addons such as http://svn.asterisk.org/svn/libss7 and
http://svn.asterisk.org/svn/asterisk/trunk/channels/sig_ss7.c ?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] ISDN & SS7

[huu giang]

How can you caculate the number of call an SS7 channel can handle ?.
I have a E1 line, can I just need use 1 channel for SS7 signal, and other 29 
channels for data tranmission ?. Is it OK.

Giang





From: Cary Fitch 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Mon, October 25, 2010 12:34:19 AM
Subject: Re: [asterisk-users] ISDN & SS7


I do not have knowledge of the SS7 vendors for Asterisk.  Using redundant 56k 
data channels, we handle calls via 6 DS3s (672 X 6 calls) from the PSTN on a 
commercial telephone switch, with no issues at all.
 
SS7 can support any number of simultaneous calls depending only on the 
bandwidth 
of the SS7 channels.  SS7 is always done on a redundant channel basis since it 
is so important.  

 
Cary



From:asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang
Sent: Sunday, October 24, 2010 12:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ISDN & SS7
 
Hi cary,
 
Can you recommend me what add-on vendors I should use ?
Can a open source solution such as chan_ss7 or libss7 support many conncurrent 
calls (for example 240 calls) ?
 
Thanks
 



From:Cary Fitch 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Sun, October 24, 2010 9:33:28 AM
Subject: Re: [asterisk-users] ISDN & SS7
SS7 is an inter-telco system using a separate network for all signaling.
 
You must have an SS7 network connection before anything will work.
 
Then the T1 Spans run 24 64k audio paths.  The SS7 net exchanges the call data 
and connection info between the switches.
 
Asterisk doesn’t support SS7 natively although I believe there are one or more 
add-on vendors.
 
CaryFitch
 
 
 



From:asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang
Sent: Sunday, October 24, 2010 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN & SS7
 
Hi all,
 
I'm being requested to deploy an IVR service using SS7. 
I've deployed Asterisk before using ISDN connection, but never with SS7.
Can anyone explain me the different between using ISDN and SS7 ? What need I do 
now to change to use SS7 ?.
 
Many thanks,
Giang


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Re: [asterisk-users] ISDN & SS7

[Cary Fitch]

I do not have knowledge of the SS7 vendors for Asterisk.  Using redundant
56k data channels, we handle calls via 6 DS3s (672 X 6 calls) from the PSTN
on a commercial telephone switch, with no issues at all.

 

SS7 can support any number of simultaneous calls depending only on the
bandwidth of the SS7 channels.  SS7 is always done on a redundant channel
basis since it is so important.  

 

Cary

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang
Sent: Sunday, October 24, 2010 12:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ISDN & SS7

 

Hi cary,

 

Can you recommend me what add-on vendors I should use ?

Can a open source solution such as chan_ss7 or libss7 support many
conncurrent calls (for example 240 calls) ?

 

Thanks

 

  _  

From: Cary Fitch 
To: Asterisk Users Mailing List - Non-Commercial Discussion

Sent: Sun, October 24, 2010 9:33:28 AM
Subject: Re: [asterisk-users] ISDN & SS7

SS7 is an inter-telco system using a separate network for all signaling.

 

You must have an SS7 network connection before anything will work.

 

Then the T1 Spans run 24 64k audio paths.  The SS7 net exchanges the call
data and connection info between the switches.

 

Asterisk doesn't support SS7 natively although I believe there are one or
more add-on vendors.

 

Cary Fitch

 

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang
Sent: Sunday, October 24, 2010 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN & SS7

 

Hi all,

 

I'm being requested to deploy an IVR service using SS7. 

I've deployed Asterisk before using ISDN connection, but never with SS7.

Can anyone explain me the different between using ISDN and SS7 ? What need I
do now to change to use SS7 ?.

 

Many thanks,

Giang

 

 

 

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Re: [asterisk-users] ISDN & SS7

[huu giang]

Hi cary,

Can you recommend me what add-on vendors I should use ?
Can a open source solution such as chan_ss7 or libss7 support many conncurrent 
calls (for example 240 calls) ?

Thanks





From: Cary Fitch 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Sun, October 24, 2010 9:33:28 AM
Subject: Re: [asterisk-users] ISDN & SS7


SS7 is an inter-telco system using a separate network for all signaling.
 
You must have an SS7 network connection before anything will work.
 
Then the T1 Spans run 24 64k audio paths.  The SS7 net exchanges the call data 
and connection info between the switches.
 
Asterisk doesn’t support SS7 natively although I believe there are one or more 
add-on vendors.
 
Cary Fitch
 
 
 



From:asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang
Sent: Sunday, October 24, 2010 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN & SS7
 
Hi all,
 
I'm being requested to deploy an IVR service using SS7. 
I've deployed Asterisk before using ISDN connection, but never with SS7.
Can anyone explain me the different between using ISDN and SS7 ? What need I do 
now to change to use SS7 ?.
 
Many thanks,
Giang


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Re: [asterisk-users] ISDN & SS7

[Cary Fitch]

SS7 is an inter-telco system using a separate network for all signaling.

 

You must have an SS7 network connection before anything will work.

 

Then the T1 Spans run 24 64k audio paths.  The SS7 net exchanges the call
data and connection info between the switches.

 

Asterisk doesn't support SS7 natively although I believe there are one or
more add-on vendors.

 

Cary Fitch

 

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang
Sent: Sunday, October 24, 2010 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN & SS7

 

Hi all,

 

I'm being requested to deploy an IVR service using SS7. 

I've deployed Asterisk before using ISDN connection, but never with SS7.

Can anyone explain me the different between using ISDN and SS7 ? What need I
do now to change to use SS7 ?.

 

Many thanks,

Giang

 

 

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Re: [asterisk-users] ISDN - Busy signal on 3rd call

[Gopalakrishnan A.N]

Still I am also facing the call disconnection when there is a third call. I
am using Netmod BRI router and the output of the BRI router lines are
connected to FXO ports in Asterisk.

Where in Asterisk I am facing the call disconnection when there is a third
call..

On Tue, Sep 28, 2010 at 4:22 PM, Paulo Santos wrote:

> Hello,
>
> Following my first mail about this issue [1], I think I know now what
> the problem is.
>
> When I have both lines being used and a third call comes in, the person
> calling doesn't get a busy tone, he gets something like line unavailable.
>
> I've been debugging mISDN and I think the reason is because asterisk is
> sending the release cause as 0.
>
>P[ 3]  --> channel:0 mode:TE cause:0 ocause:0 rad: cad:
>
> The request from the telephone company's switch seems correct, a SETUP
> message (if 08 is Q.931, 05 is SETUP).
>
>02 ff 03 08  01 04 05 a1  04 03 80 90
>a3 18 01 80  6c 0b 01 83  39 31 36 33
>39 31 37 34  32 70 03 c1  38 34
>
> I've changed misdn.conf so it sends a release cause as 17 (user busy),
> but I get the same behaviour - cause:0 ocause:0.
>
> Anyone knows how can I force asterisk to send cause 16 or 17 in this
> situation?
>
> Thanks in advance.
>
> Best regards,
> Paulo Santos
>
> misdn.conf: http://pastebin.com/FmgECqkU
> misdn debug: http://pastebin.com/Tg6wPKBD
>
> [1]
> http://www.mail-archive.com/asterisk-users@lists.digium.com/msg244330.html
>
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>



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Re: [asterisk-users] ISDN - Busy signal on 3rd call

[Paulo Santos]

Hello,

Gopalakrishnan A.N wrote:
> I am also facing the call disconnection if there is a third call. I
> tried disable call waiting in the BRI router, but now it has been
> reduced, it means call disconnection is not permanent but seems to be
> occasion, let say per day two times there is a call disconnection.

In the call disconnections after disabling call waiting, do you still
get the following error as well?

P[ 3]  --> !! lib: No free channel!

I've called the telephone company and they told me they had already
disabled call waiting and answering machine, but because they pretty
much have no idea what they're talking about I'll call them again and
confirm those features are actually disabled.

Best regards,
Paulo Santos

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Re: [asterisk-users] ISDN - Busy signal on 3rd call

[Gopalakrishnan A.N]

I am also facing the call disconnection if there is a third call. I tried
disable call waiting in the BRI router, but now it has been reduced, it
means call disconnection is not permanent but seems to be occasion, let say
per day two times there is a call disconnection.

On Wed, Sep 29, 2010 at 3:20 PM, Paulo Santos wrote:

> I'm resending this email to the list, apparently the first one didn't go
> through. If it did, I apologize for the re-post.
>
> Hello,
>
> Following my first mail about this issue [1], I think I know now what
> the problem is.
>
> When I have both lines being used and a third call comes in, the person
> calling doesn't get a busy tone, he gets something like line unavailable.
>
> I've been debugging mISDN and I think the reason is because asterisk is
> sending the release cause as 0.
>
>P[ 3]  --> channel:0 mode:TE cause:0 ocause:0 rad: cad:
>
> The request from the telephone company's switch seems correct, a SETUP
> message (if 08 is Q.931, 05 is SETUP).
>
>02 ff 03 08  01 04 05 a1  04 03 80 90
>a3 18 01 80  6c 0b 01 83  39 31 36 33
>39 31 37 34  32 70 03 c1  38 34
>
> I've changed misdn.conf so it sends a release cause as 17 (user busy),
> but I get the same behaviour - cause:0 ocause:0.
>
> Anyone knows how can I force asterisk to send cause 16 or 17 in this
> situation?
>
> Thanks in advance.
>
> Best regards,
> Paulo Santos
>
> misdn.conf: http://pastebin.com/FmgECqkU
> misdn debug: http://pastebin.com/Tg6wPKBD
>
> [1]
> http://www.mail-archive.com/asterisk-users@lists.digium.com/msg244330.html
>
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>



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Re: [asterisk-users] ISDN -> SIP

[Gergo Csibra]

Friday, June 11, 2010, 12:27:08 AM, Tzafrir wrote:

> On Fri, Jun 11, 2010 at 12:19:37AM +0200, Gergo Csibra wrote:
>> Okay. There's some problems with mISDN v2: I'm unable to compile
>> zaphfc, because there's no source for it. mISDN v2 works with hfcpci
>> too?

> Certainly there is.

> It's also part of the standard dahdi-extra patch. See
> http://git.tzafrir.org.il/?p=dahdi-extra.git;a=tree
> http://svn.debian.org/viewsvn/pkg-voip/dahdi-linux/trunk/debian/patches/dahdi_linux_extra

OK. Last time I checked (2009. dec) there wasn't :)

I downloaded dahdi-extra snapshot, and dahdi from asterisk.org,
untared, I have two directories:

dahdi-extra
dahdi-linux-complete-2.3.0.1+2.3.0

What's next?

I don't understand where to start make with MODULES_EXTRA and
SUBDIRS_EXTRA parameters, and how can I configure drivers...


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Re: [asterisk-users] ISDN -> SIP

[Philipp von Klitzing]

Hi!

> But if i try to establish ISDN->SIP-Dialout, the redirection ist not
> working.

Your logs are very sketchy and difficult to understand because you 
stripped them of some details and cut out lines in between.

  > From: "5" ;tag=as1ec770c5

This line does not make much sense.

> exten => 123456,1,Dial(SIP/987...@sip)
> exten => 123457,1,Dial(SIP/33)
> ; both not working. Do i need to accept the call before?

What is the CLI output of:
  "sip show peer sip" and
  "sip show peer 33"?

Note: It it not good practice to define local sip peers (phones) with
numbers only (like 33). Use alphanumeric names like "phone1" or
"mac11223344566".

> The Call is rejected whith the message "No Connection" (de: "kein
> Anschluss unter dieser Nummer").
...
> -- chan_capi queue frame:[ TYPE: Control (4) SUBCLASS: Hangup (1) ]

Yes, that is what you get: A hangup cause code of "1", which means 
"number not allocated". Use the dialplan variables ${HANGUPCAUSE} and
${DIALSTATUS} to process accordingly this in extensions.conf.

So: Obviously you dialed the wrong number. ;->

> INVITE sip:987...@sip SIP/2.0
> To: 

> What is wrong. An why SIP-to internal SIP-Phone(/33)

See above "sip show peer 33". Maybe you haven't registered the phone, or
you have forgotten to give it a static IP in sip.conf.

Philipp


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Re: [asterisk-users] ISDN -> SIP

[Stefan Dreyer]

On 06/10/10 23:19, Philipp von Klitzing wrote:
> Hi!
> 
>> i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
>> CentOS 5.5. The only thing, i want to do is a call-redirection from an
>> isdn-call to my mobile via sip-account.
> 
> Unless you are using mISDN v2: Do yourself a favour and switch to CAPI 
> with chan_capi and fcpci. mISDN v1 is a guarantee for headaches (and 
> unstable systems).

After a little torture to install fcpci, SIP->ISDN-Dialout is working.
But if i try to establish ISDN->SIP-Dialout, the redirection ist not
working.

[isdn-in]
; MSN 123456 -> 987...@sip
exten => 123456,1,Dial(SIP/987...@sip)
exten => 123457,1,Dial(SIP/33)
; both not working. Do i need to accept the call before?

[misdnOut]
; DIAL-Out-Working
exten => _0X.,1,Dial(CAPI/contr1/${EXTEN})

[default]
include => misdnOut

The Call is rejected whith the message "No Connection" (de: "kein
Anschluss unter dieser Nummer"). But the outgoing SIP-Call is made. The
log shows:


-- CONNECT_IND
(PLCI=0x101,DID=12345,CID=5,CIP=0x10,CONTROLLER=0x1)
  == Started pbx on channel CAPI/ISDN1#02/12345-10
   -- Executing [12...@isdn-in:1] Dial("CAPI/ISDN1#02/12345-10",
"SIP/87...@sip,45,t") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 212.x.y.z port 15256
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to a.b.c.d:5060:

INVITE sip:987...@sip SIP/2.0
Via: SIP/2.0/UDP 212.x.y.z:5060;branch=z9hG4bK5a55a76e;rport

Max-Forwards: 70
From: "5" ;tag=as1ec770c5

To: 
Contact: 
Call-ID: 1979cd9a3c3cb9013e9cd9660cd33...@sip
CSeq: 102 INVITE

...
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer
...
v=0
o=root 1971852647 1971852647 IN IP4 212.x.y.z
s=Asterisk PBX 1.6.2.8
c=IN IP4 212.x.y.z
t=0 0
m=audio 15256 RTP/AVP 8 3 0
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called 987...@sip
<--- SIP read from UDP:a.b.c.d:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 212.x.y.z:5060;branch=z9hG4bK5a55a76e;rport
From: "5" ;tag=as1ec770c5
To: 
Contact: sip:987...@a.b.c.d:5060
Call-ID: 1979cd9a3c3cb9013e9cd9660cd33...@sip
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm="sip...",nonce="3042653437",algorithm=MD5
Content-Length: 0
...
---
Audio is at 212.x.y.z port 15256
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to a.d.c.d:5060:
INVITE sip:987...@sip SIP/2.0
Via: SIP/2.0/UDP 212.x.y.z:5060;branch=z9hG4bK51f5e20e;rport
Max-Forwards: 70
From: "5" ;tag=as1ec770c5
To: 
Contact: 
Call-ID: 1979cd9a3c3cb9013e9cd9660cd33...@sip.voipdiscount.com
CSeq: 103 INVITE
...
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8
(alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)


Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port a.b.c.d:41302

-- SIP/sip-0007 is making progress passing it to
CAPI/ISDN1#02/12345-10
-- chan_capi queue frame:[ TYPE: Control (4) SUBCLASS: Hangup (1) ]
[ISDN1#02]
Scheduling destruction of SIP dialog '19@sip' in 32000 ms (Method:
INVITE)
Reliably Transmitting (no NAT) to 77.72.169.134:5060:

Scheduling destruction of SIP dialog '1...@sip' in 32000 ms (Method: INVITE)
  == Spawn extension (isdn-in, 12345, 1) exited non-zero on
'CAPI/ISDN1#02/12345-10'
  == ISDN1#02: Interface cleanup PLCI=0xdead

What is wrong. An why SIP-to internal SIP-Phone(/33) is not working.
>From internal SIP to ISDN and internal SIP to external SIP is working.

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Re: [asterisk-users] ISDN -> SIP

[Gergo Csibra]

Thursday, June 10, 2010, 11:19:16 PM, Philipp wrote:

>> i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
>> CentOS 5.5. The only thing, i want to do is a call-redirection from an
>> isdn-call to my mobile via sip-account.

> Unless you are using mISDN v2: Do yourself a favour and switch to CAPI 
> with chan_capi and fcpci. mISDN v1 is a guarantee for headaches (and 
> unstable systems).

Okay. There's some problems with mISDN v2: I'm unable to compile
zaphfc, because there's no source for it. mISDN v2 works with hfcpci
too?

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Re: [asterisk-users] ISDN -> SIP

[Philipp von Klitzing]

Hi!

> i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
> CentOS 5.5. The only thing, i want to do is a call-redirection from an
> isdn-call to my mobile via sip-account.

Unless you are using mISDN v2: Do yourself a favour and switch to CAPI 
with chan_capi and fcpci. mISDN v1 is a guarantee for headaches (and 
unstable systems).

Philipp


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Re: [asterisk-users] ISDN -> SIP

[Tzafrir Cohen]

On Fri, Jun 11, 2010 at 12:19:37AM +0200, Gergo Csibra wrote:
> Thursday, June 10, 2010, 11:19:16 PM, Philipp wrote:
> 
> >> i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
> >> CentOS 5.5. The only thing, i want to do is a call-redirection from an
> >> isdn-call to my mobile via sip-account.
> 
> > Unless you are using mISDN v2: Do yourself a favour and switch to CAPI 
> > with chan_capi and fcpci. mISDN v1 is a guarantee for headaches (and 
> > unstable systems).
> 
> Okay. There's some problems with mISDN v2: I'm unable to compile
> zaphfc, because there's no source for it. mISDN v2 works with hfcpci
> too?

Certainly there is.

It's also part of the standard dahdi-extra patch. See
http://git.tzafrir.org.il/?p=dahdi-extra.git;a=tree
http://svn.debian.org/viewsvn/pkg-voip/dahdi-linux/trunk/debian/patches/dahdi_linux_extra

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Re: [asterisk-users] ISDN config: LBO values

[Jaap Winius]

Quoting Tilghman Lesher :

> http://www.digchip.com/datasheets/parts/datasheet/222/82V2088-pdf.php
>
> See pages 17-18 of the associated PDF.  While this is not the T1 framer chip
> used, the values are identical, which leads me to believe that these values
> are actually industry standard.

Well, maybe more like a defacto standard. But, it still doesn't  
explain when to use the different values in a software configuration,  
e.g. with Asterisk.

As a term, DSX-1 is confusing. One description can be found in the  
Wikipedia article for T-carrier, which says it stands for Digital  
Signal Crossconnect: "DS1 signals are interconnected typically at  
Central Office locations at a common metallic cross-connect point  
known as a DSX-1. ..."

On the other hand, articles like the following use DSX-1 to describe  
customer site connections:

* Adtran NetVanta T1 Access Router
   http://www.arcelect.com/netvanta_access_t1_router.htm

The diagram shows how two different NetVanta models can be used to  
connect a T-1 line to a PBX.

There's also this page:

* Primary Rate Interface ISDN Line Port
   http://www22.verizon.com/wholesale/solutions/solution/pri+rate+isdn.html

Near the end, under Detailed Information, it says:

 "PRI service consists of a 4-wire DSX-1 port associated
 with a local switching system and the 4-wire DSX-1
 cross-connect between the OTC DSX-1 termination and the
 local switching system DSX-1 termination.

 "PRI ports are DSX-1 interfaces that meet the electrical
 specifications in ANSI T1.102. PRI service and use B8ZS
 line code and the Extended Superframe Format (ESF)
 described in ANSI T1.403."

Again, the term DSX-1 is used to describe a CPE port. In such cases, I  
think it will probably be appropriate to use the "DSX-1" column in the  
LBO table.

Still, what's the difference between "CSU" and "DSX-1"??

Speculation:

Could it be that "CSU" refers to situations where there is no  
equipment of any kind between the demarcation point and the ISDN card?  
In such cases, the ISDN card will have an integrated CSU, and the  
length of the cable will be unknown (thousands of feet), but you can  
know the attenuation value in dB; either by measuring it, or by  
getting it from the telco.

This scenario may only occur in the United States.

On the other hand, "DSX-1" will refer to situations where the ISDN  
card is connected -- via a DSX-1 port and a cable of a known length --  
to an external CSU and/or DSU. In turn, this equipment is connected to  
the demarc.

This scenario may apply in all other situations, e.g. ISDN BRI cards  
that connect to an NT-1.

Does this sound reasonable?

Thanks,

Jaap

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Re: [asterisk-users] ISDN config: LBO values

[Tilghman Lesher]

On Sunday 16 May 2010 17:32:09 Jaap Winius wrote:
> Quoting Tilghman Lesher :
> >> The value selected should almost always be zero. However, if the cable
> >> is of a significant length, another value must be selected, but which
> >> one? There are two columns: CSU and DSX-1. When is it appropriate to
> >> use the one or the other to determine the correct LBO value?
> >
> > Each LBO value is a different amount of loss to be expected on the
> > line, and therefore the signal is amplified a commensurate amount.
> > What it really comes down to is what works for you.
>
> That's the usual approach, but if I was still happy with it I would
> not have asked the question. According to the manual, the values are
> found in a table, but what good is that if you can't make any sense of
> it?

http://www.digchip.com/datasheets/parts/datasheet/222/82V2088-pdf.php

See pages 17-18 of the associated PDF.  While this is not the T1 framer chip
used, the values are identical, which leads me to believe that these values
are actually industry standard.  The values used are merely inputs into the
firmware and the T1 framer does the rest.  BTW, you can find the datasheet
for the actual T1 framer chip, but it's less helpful than the one above.

-- 
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Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] ISDN config: LBO values

[Jaap Winius]

Quoting Tilghman Lesher :

>> The value selected should almost always be zero. However, if the cable
>> is of a significant length, another value must be selected, but which
>> one? There are two columns: CSU and DSX-1. When is it appropriate to
>> use the one or the other to determine the correct LBO value?
>
> Each LBO value is a different amount of loss to be expected on the
> line, and therefore the signal is amplified a commensurate amount.
> What it really comes down to is what works for you.

That's the usual approach, but if I was still happy with it I would  
not have asked the question. According to the manual, the values are  
found in a table, but what good is that if you can't make any sense of  
it?

In the mean time, I've googled some more and found one text that  
suggests CSU and DSX-1 are both T1 trunk interface types, while  
another suggests that a DSX-1 is an interface that a CSU is attached to.

It seems to me that the table refers to two situations that used to  
(or maybe still do) occur in North America in which an ISDN card is be  
attached to a T1 trunk line via a CSU/DSU (the "DSX-1"), or only a  
CSU. In the latter case, the ISDN card must also act as a DSU.

Can anyone say is this is correct? Any further explanation would be welcome.

Cheers,

Jaap

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Re: [asterisk-users] ISDN config: LBO values

[Tilghman Lesher]

On Saturday 15 May 2010 19:28:37 Jaap Winius wrote:
> When configuring Asterisk with an ISDN card, it will at one point
> become necessary to select the LBO (Line Build-Out) value. This is an
> integer (0-7) that is determined by the length of the cable and is
> selected from the following table. Many of us are familiar with it:
>
>  CSU (dB)   DSX-1 (feet)
> ---
> 00  0?133
> 1   133?266
> 2   266?399
> 3   399?533
> 4   533?655
> 5-7.5
> 6-15
> 7-22.5
>
> The value selected should almost always be zero. However, if the cable
> is of a significant length, another value must be selected, but which
> one? There are two columns: CSU and DSX-1. When is it appropriate to
> use the one or the other to determine the correct LBO value?

Each LBO value is a different amount of loss to be expected on the line, and
therefore the signal is amplified a commensurate amount.  What it really comes
down to is what works for you.

-- 
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Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] ISDN: Inband DTMF doesn't trigger transfer feature

[Christian Theune]

Hi,

(posting again as my previous log attachments were too large. Sorry if 
this should end up as a double posting.)


On 12/14/2009 09:31 PM, Tzafrir Cohen wrote:

On Mon, Dec 14, 2009 at 05:52:40PM +0100, Christian Theune wrote:

Hi there,

I just upgraded a relatively old Asterisk installation (1.2) in our
office to a relatively new version (1.6svn from last wednesday) which
runs a Junghans QuadBRI card [1].

To get this flying I got dahdi-linux, dahdi-tools and libpri from SVN as
well.

After a while of juggling it "works".

What doesn't work: connected ISDN devices (Gigaset phones connected to
QuadBRI with a port in NT mode, signalling via bri_net_ptmp) can not
transfer calls by pressing "#". SIP phones can.

I'm at a point where I can't even get *any* output to debug.


Is overlap dialing used?


Yes and no: I retried with overlap dialing disabled (overlapdial=no in 
chan_dahdi.conf) but that didn't change anything.



Wher ecan we see your pri debug trace?


I attached a sanitized version. The incoming number is me calling from 
my mobile phone through the trunk (DAHDI/g1) to one of the DECT handsets 
(DAHDI/g4).


Also, find the chan_dahdi.conf attached.

Christian

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chan_dahdi.conf.gz
Description: application/gzip


pridebug.log.gz
Description: application/gzip
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Re: [asterisk-users] ISDN: Inband DTMF doesn't trigger transfer feature

[Christian Theune]

On 12/14/2009 09:31 PM, Tzafrir Cohen wrote:
> On Mon, Dec 14, 2009 at 05:52:40PM +0100, Christian Theune wrote:
>> Hi there,
>>
>> I just upgraded a relatively old Asterisk installation (1.2) in our
>> office to a relatively new version (1.6svn from last wednesday) which
>> runs a Junghans QuadBRI card [1].
>>
>> To get this flying I got dahdi-linux, dahdi-tools and libpri from SVN as
>> well.
>>
>> After a while of juggling it "works".
>>
>> What doesn't work: connected ISDN devices (Gigaset phones connected to
>> QuadBRI with a port in NT mode, signalling via bri_net_ptmp) can not
>> transfer calls by pressing "#". SIP phones can.
>>
>> I'm at a point where I can't even get *any* output to debug.
>
> Is overlap dialing used?
>
> Wher ecan we see your pri debug trace?

Somewhere in the queue of this mailing list ... is there an 
administrator around who can review my posting that has 41kb instead of 
40kb with the logs attached?

Thanks.


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Re: [asterisk-users] ISDN: Inband DTMF doesn't trigger transfer feature

[Tzafrir Cohen]

On Mon, Dec 14, 2009 at 05:52:40PM +0100, Christian Theune wrote:
> Hi there,
> 
> I just upgraded a relatively old Asterisk installation (1.2) in our 
> office to a relatively new version (1.6svn from last wednesday) which 
> runs a Junghans QuadBRI card [1].
> 
> To get this flying I got dahdi-linux, dahdi-tools and libpri from SVN as 
> well.
> 
> After a while of juggling it "works".
> 
> What doesn't work: connected ISDN devices (Gigaset phones connected to 
> QuadBRI with a port in NT mode, signalling via bri_net_ptmp) can not 
> transfer calls by pressing "#". SIP phones can.
> 
> I'm at a point where I can't even get *any* output to debug. 

Is overlap dialing used?

Wher ecan we see your pri debug trace?

> I went 
> through the relevant configs as good as I can a couple of times but am 
> out of ideas. I'm guessing that this is somehow related to either 
> Asterisk not being able to monitor the call for inband DTMF (which I can 
> prove how?) or something in the relatively new bri_net_ptmp code that 
> breaks it (which I can't prove either).
> 
> A pointer how to debug this further would be very appreciated.
> 
> Best regards,
> Christian
> 
> [1] lspci output of QuadBRI
> 
> 02:0c.0 ISDN controller: Cologne Chip Designs GmbH ISDN network 
> Controller [HFC-4S] (rev 01)

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Re: [asterisk-users] ISDN: Inband DTMF doesn't trigger transfer feature

[Christian Theune]

On 12/14/2009 06:45 PM, Olivier wrote:
>
>
> 2009/12/14 Christian Theune mailto:c...@gocept.com>>
>
> Hi there,
>
> I just upgraded a relatively old Asterisk installation (1.2) in our
> office to a relatively new version (1.6svn from last wednesday) which
> runs a Junghans QuadBRI card [1].
>
> To get this flying I got dahdi-linux, dahdi-tools and libpri from SVN as
> well.
>
> After a while of juggling it "works".
>
> What doesn't work: connected ISDN devices (Gigaset phones connected to
> QuadBRI with a port in NT mode, signalling via bri_net_ptmp)
>
>
> I would be very pleased to be corrected but I don't think NT/ptmp mode
> is supported now (I think this is an ongoing work but it's not complete)
> through Dahdi.

Right, it's probably ongoing. Revision 225692 added it relatively recently.

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Re: [asterisk-users] ISDN: Inband DTMF doesn't trigger transfer feature

[Olivier]

2009/12/14 Christian Theune 

> Hi there,
>
> I just upgraded a relatively old Asterisk installation (1.2) in our
> office to a relatively new version (1.6svn from last wednesday) which
> runs a Junghans QuadBRI card [1].
>
> To get this flying I got dahdi-linux, dahdi-tools and libpri from SVN as
> well.
>
> After a while of juggling it "works".
>
> What doesn't work: connected ISDN devices (Gigaset phones connected to
> QuadBRI with a port in NT mode, signalling via bri_net_ptmp)


I would be very pleased to be corrected but I don't think NT/ptmp mode is
supported now (I think this is an ongoing work but it's not complete)
through Dahdi.




> can not
> transfer calls by pressing "#". SIP phones can.
>
> I'm at a point where I can't even get *any* output to debug. I went
> through the relevant configs as good as I can a couple of times but am
> out of ideas. I'm guessing that this is somehow related to either
> Asterisk not being able to monitor the call for inband DTMF (which I can
> prove how?) or something in the relatively new bri_net_ptmp code that
> breaks it (which I can't prove either).
>
> A pointer how to debug this further would be very appreciated.
>
> Best regards,
> Christian
>
> [1] lspci output of QuadBRI
>
> 02:0c.0 ISDN controller: Cologne Chip Designs GmbH ISDN network
> Controller [HFC-4S] (rev 01)
>
> --
> Christian Theune · c...@gocept.com
> gocept gmbh & co. kg · forsterstraße 29 · 06112 halle (saale) · germany
> http://gocept.com · tel +49 345 1229889 0 · fax +49 345 1229889 1
> Zope and Plone consulting and development
>
>
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Re: [asterisk-users] ISDN from Macau CTM

[Si Tai Fan]

Come on! Anyone? How about anyone doing Asterisk in Macau (China)?

Si Tai Fan wrote:

Hi

Has anyone successfully connected to Macau CTM using the E1 TE110P 
card? They are using the R2 signaling for their IDAP connection.


Thanks,
Si


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Re: [asterisk-users] ISDN Timer T309

[Afonso Zimmermann]

Martin escreveu:

  Hi,

You're right. I wasn't aware of this patch getting into the code.
In the version you're running the code is already present.

The only problem I see is that some other timer kicks in here and the
T309 cannot be scheduled.

q931.c has this ...
/* For a call in Active state, activate T309 only if there is no timer
already running. */

You'd have to probably dig deeper in it to find out more. But this is
the "latest" explanation I see.
That would explain why the call is disconnected/hanged up right when
the alarm happens.

One way to fix it for you would be to remove the already running timer
so the T309 could be scheduled since
anyways all other timers do not matter since without T309 the call is
hanged up anyways.

Martin

On Mon, Apr 6, 2009 at 12:23 PM, Afonso Zimmermann  wrote:
  
  
Martin escreveu:

Based on the Asterisk logs you posted the Asterisk doesn't have it
implemented per:

"The implementation of timer T309 in the user side is optional"

Martin

On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann 
wrote:


Martin escreveu:

What is the specification for T309 ? I'm too lazy to look it up.

The default behaviour when the alarm of layer 1 (electrical T1/E1) is
detected is to assume
all calls dropped on both sides and that's what Asterisk does.

The timer is simply deactivated since all the calls are supposed to
drop. I believe that agrees with Q921/Q931 specs.

Martin

On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann 
wrote:


Hi everione,

I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri
1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the
timer fail with a telco link in this scenario:

Telco Phone <--> Telco <---> Asterisk <> Sip
Phone

When i make a call from Telco Phone to Sip Phone, the call complete, but
when i disconnect the link and reconnect in few seconds, the Asterisk clear
call:

[Apr  3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 1: Red Alarm
[Apr  3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got
event: Alarm (4) on Primary D-channel of span 1
  == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on
'DAHDI/1-1'
[Apr  3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No
D-channels available!  Using Primary channel 16 as D-channel anyway!
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 2: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 2: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 3: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 3: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 4: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 4: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 5: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 5: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 6: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 6: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 7: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 7: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 8: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 8: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 9: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 9: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 10: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 10: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 11: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 11: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 12: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable

Re: [asterisk-users] ISDN Timer T309

[Martin]

Hi,

You're right. I wasn't aware of this patch getting into the code.
In the version you're running the code is already present.

The only problem I see is that some other timer kicks in here and the
T309 cannot be scheduled.

q931.c has this ...
/* For a call in Active state, activate T309 only if there is no timer
already running. */

You'd have to probably dig deeper in it to find out more. But this is
the "latest" explanation I see.
That would explain why the call is disconnected/hanged up right when
the alarm happens.

One way to fix it for you would be to remove the already running timer
so the T309 could be scheduled since
anyways all other timers do not matter since without T309 the call is
hanged up anyways.

Martin

On Mon, Apr 6, 2009 at 12:23 PM, Afonso Zimmermann  wrote:
> Martin escreveu:
>
> Based on the Asterisk logs you posted the Asterisk doesn't have it
> implemented per:
>
> "The implementation of timer T309 in the user side is optional"
>
> Martin
>
> On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann 
> wrote:
>
>
> Martin escreveu:
>
> What is the specification for T309 ? I'm too lazy to look it up.
>
> The default behaviour when the alarm of layer 1 (electrical T1/E1) is
> detected is to assume
> all calls dropped on both sides and that's what Asterisk does.
>
> The timer is simply deactivated since all the calls are supposed to
> drop. I believe that agrees with Q921/Q931 specs.
>
> Martin
>
> On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann 
> wrote:
>
>
> Hi everione,
>
> I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri
> 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the
> timer fail with a telco link in this scenario:
>
> Telco Phone <--> Telco <---> Asterisk <> Sip
> Phone
>
> When i make a call from Telco Phone to Sip Phone, the call complete, but
> when i disconnect the link and reconnect in few seconds, the Asterisk clear
> call:
>
> [Apr  3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 1: Red Alarm
> [Apr  3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got
> event: Alarm (4) on Primary D-channel of span 1
>   == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on
> 'DAHDI/1-1'
> [Apr  3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No
> D-channels available!  Using Primary channel 16 as D-channel anyway!
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 2: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 2: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 3: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 3: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 4: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 4: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 5: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 5: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 6: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 6: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 7: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 7: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 8: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 8: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 9: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 9: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 10: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 10: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 11: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 11: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_

Re: [asterisk-users] ISDN Timer T309

[Afonso Zimmermann]

Martin escreveu:

  Based on the Asterisk logs you posted the Asterisk doesn't have it
implemented per:

"The implementation of timer T309 in the user side is optional"

Martin

On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann  wrote:
  
  
Martin escreveu:

What is the specification for T309 ? I'm too lazy to look it up.

The default behaviour when the alarm of layer 1 (electrical T1/E1) is
detected is to assume
all calls dropped on both sides and that's what Asterisk does.

The timer is simply deactivated since all the calls are supposed to
drop. I believe that agrees with Q921/Q931 specs.

Martin

On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann 
wrote:


Hi everione,

I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri
1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the
timer fail with a telco link in this scenario:

Telco Phone <--> Telco <---> Asterisk <> Sip
Phone

When i make a call from Telco Phone to Sip Phone, the call complete, but
when i disconnect the link and reconnect in few seconds, the Asterisk clear
call:

[Apr  3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 1: Red Alarm
[Apr  3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got
event: Alarm (4) on Primary D-channel of span 1
  == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on
'DAHDI/1-1'
[Apr  3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No
D-channels available!  Using Primary channel 16 as D-channel anyway!
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 2: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 2: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 3: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 3: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 4: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 4: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 5: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 5: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 6: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 6: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 7: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 7: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 8: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 8: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 9: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 9: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 10: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 10: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 11: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 11: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 12: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 12: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 13: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 13: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 14: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 14: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 15: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 15: Invalid argument
[Apr  3 

Re: [asterisk-users] ISDN Timer T309

[Martin]

Based on the Asterisk logs you posted the Asterisk doesn't have it
implemented per:

"The implementation of timer T309 in the user side is optional"

Martin

On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann  wrote:
> Martin escreveu:
>
> What is the specification for T309 ? I'm too lazy to look it up.
>
> The default behaviour when the alarm of layer 1 (electrical T1/E1) is
> detected is to assume
> all calls dropped on both sides and that's what Asterisk does.
>
> The timer is simply deactivated since all the calls are supposed to
> drop. I believe that agrees with Q921/Q931 specs.
>
> Martin
>
> On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann 
> wrote:
>
>
> Hi everione,
>
> I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri
> 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the
> timer fail with a telco link in this scenario:
>
> Telco Phone <--> Telco <---> Asterisk <> Sip
> Phone
>
> When i make a call from Telco Phone to Sip Phone, the call complete, but
> when i disconnect the link and reconnect in few seconds, the Asterisk clear
> call:
>
> [Apr  3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 1: Red Alarm
> [Apr  3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got
> event: Alarm (4) on Primary D-channel of span 1
>   == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on
> 'DAHDI/1-1'
> [Apr  3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No
> D-channels available!  Using Primary channel 16 as D-channel anyway!
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 2: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 2: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 3: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 3: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 4: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 4: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 5: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 5: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 6: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 6: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 7: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 7: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 8: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 8: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 9: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 9: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 10: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 10: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 11: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 11: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 12: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 12: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 13: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 13: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 14: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 14: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 15: Red Alarm
> [Apr 

Re: [asterisk-users] ISDN Timer T309

[Afonso Zimmermann]

Martin escreveu:

  What is the specification for T309 ? I'm too lazy to look it up.

The default behaviour when the alarm of layer 1 (electrical T1/E1) is
detected is to assume
all calls dropped on both sides and that's what Asterisk does.

The timer is simply deactivated since all the calls are supposed to
drop. I believe that agrees with Q921/Q931 specs.

Martin

On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann  wrote:
  
  
Hi everione,

I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri
1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the
timer fail with a telco link in this scenario:

Telco Phone <--> Telco <---> Asterisk <> Sip
Phone

When i make a call from Telco Phone to Sip Phone, the call complete, but
when i disconnect the link and reconnect in few seconds, the Asterisk clear
call:

[Apr  3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 1: Red Alarm
[Apr  3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got
event: Alarm (4) on Primary D-channel of span 1
  == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on
'DAHDI/1-1'
[Apr  3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No
D-channels available!  Using Primary channel 16 as D-channel anyway!
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 2: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 2: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 3: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 3: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 4: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 4: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 5: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 5: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 6: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 6: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 7: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 7: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 8: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 8: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 9: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 9: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 10: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 10: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 11: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 11: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 12: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 12: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 13: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 13: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 14: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 14: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 15: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 15: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
alarm on channel 17: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
to disable echo cancellation on channel 17: Invalid argument

Re: [asterisk-users] ISDN Timer T309

[Martin]

What is the specification for T309 ? I'm too lazy to look it up.

The default behaviour when the alarm of layer 1 (electrical T1/E1) is
detected is to assume
all calls dropped on both sides and that's what Asterisk does.

The timer is simply deactivated since all the calls are supposed to
drop. I believe that agrees with Q921/Q931 specs.

Martin

On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann  wrote:
> Hi everione,
>
> I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri
> 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the
> timer fail with a telco link in this scenario:
>
> Telco Phone <--> Telco <---> Asterisk <> Sip
> Phone
>
> When i make a call from Telco Phone to Sip Phone, the call complete, but
> when i disconnect the link and reconnect in few seconds, the Asterisk clear
> call:
>
> [Apr  3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 1: Red Alarm
> [Apr  3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got
> event: Alarm (4) on Primary D-channel of span 1
>   == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on
> 'DAHDI/1-1'
> [Apr  3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No
> D-channels available!  Using Primary channel 16 as D-channel anyway!
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 2: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 2: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 3: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 3: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 4: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 4: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 5: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 5: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 6: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 6: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 7: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 7: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 8: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 8: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 9: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 9: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 10: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 10: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 11: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 11: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 12: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 12: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 13: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 13: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 14: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 14: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 15: Red Alarm
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
> to disable echo cancellation on channel 15: Invalid argument
> [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
> alarm on channel 17: Red Alarm
> [Apr  3 10:44:4

Re: [asterisk-users] ISDN PRI settings for Telus BC network

[Gondar Monn]

Thanks a lot, will let you know how things are going when I get them to turn
on two way dialing .
By the way any pointers on how to connect to Portmaster ?
Looks like we are going to have to share some PRIs lines with portmaster
(dialup)

FYI, I am sticking with Asterisk 1.4 for now ...

Thanks!

Gondar

On Sat, Dec 6, 2008 at 3:49 PM, Trevor Peirce <[EMAIL PROTECTED]> wrote:

> Gondar Monn wrote:
> > Hi there!
> > Does anyone deal with Telus in BC ? We have some PRI lines that were
> > used for dialup, would like to convert them for pbx system, talked
> > with some technicians @ Telus, but the information given was not
> > clear, kind of: "try this see if it works" Does anyone here have
> > the settings required to talk to there equipment ?
>
> A few years ago I had a PRI from TELUS.  The winning zaptel.conf line:
>
> span=1,1,0,esf,b8zs
> bchan=1-23
> dchan=24
>
> And in zapata.conf it was
>
> switchtype=national
> pridialplan=unknown
> prilocaldialplan=national
> signalling=pri_cpe
>
>
> These are both from the asterisk 1.2 days so a lot may have changed
> between now and then...
>
> Hope this helps,
> Trevor
>
>
>
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Re: [asterisk-users] ISDN PRI settings for Telus BC network

[Trevor Peirce]

Gondar Monn wrote:
> Hi there!
> Does anyone deal with Telus in BC ? We have some PRI lines that were 
> used for dialup, would like to convert them for pbx system, talked 
> with some technicians @ Telus, but the information given was not 
> clear, kind of: "try this see if it works" Does anyone here have 
> the settings required to talk to there equipment ?

A few years ago I had a PRI from TELUS.  The winning zaptel.conf line:

span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

And in zapata.conf it was

switchtype=national
pridialplan=unknown
prilocaldialplan=national
signalling=pri_cpe


These are both from the asterisk 1.2 days so a lot may have changed 
between now and then...

Hope this helps,
Trevor



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Re: [asterisk-users] ISDN Cause codes

[Martin Smith]

Hi Robert & all,

Maybe someone else can speak to using Progress(), but I don't know if it
is required or not. On our system, we didn't need it, and these settings
below (plus a call to the telco to tell them to turn on operator
messages, don't eat them) did the trick.

Good luck,

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Robert Boardman
> Sent: Saturday, November 22, 2008 11:43 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] ISDN Cause codes
> 
> I have found that the messages are not played as the hangup 
> cause clears 
> down the channel and passed hangup to the other end
> 
> should I have progress() before the dial command?
> 
> Robb
> 
> Martin Smith wrote:
> > Hi Robert,
> >
> > I'd recommend the following options for Dial() so that you 
> corroborate
> > operator messages w/ cause codes:
> >
> >  1. remove R and r - we've found this can supress operator 
> recordings on
> > early audio
> >  2. likewise, remove m to disable MOH
> >
> > Also, check the values of DIALSTATUS to compare to HANGUPCAUSE.
> >
> > Good luck,
> >
> > Martin Smith, Systems Developer
> > [EMAIL PROTECTED]
> > Bureau of Economic and Business Research
> > University of Florida
> > (352) 392-0171 Ext. 221 
> >
> >  
> >
> >   
> >> -Original Message-----
> >> From: [EMAIL PROTECTED] 
> >> [mailto:[EMAIL PROTECTED] On Behalf Of 
> >> Robert Boardman
> >> Sent: Friday, November 21, 2008 3:07 PM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] ISDN Cause codes
> >>
> >> Thanks for the reply
> >>
> >> Could you be a little more specific?
> >>
> >> Thanks
> >> Robb
> >>
> >> Martin Smith wrote:
> >> 
> >>> Hi Robert,
> >>>
> >>> I'd suggest tweaking the Dial() arguments so that you (1) 
> >>>   
> >> allow early
> >> 
> >>> audio, (2) don't force it play ringing to the calling 
> party, and (3)
> >>> modify any other options to be as relaxed as possible. if 
> >>>   
> >> you make those
> >> 
> >>> changes, you'll start hearing the operator message 
> >>>   
> >> recordings and those
> >> 
> >>> are sometimes easier to reference against the cause codes.
> >>>
> >>> Cheers,
> >>>
> >>>
> >>> Martin Smith, Systems Developer
> >>> [EMAIL PROTECTED]
> >>> Bureau of Economic and Business Research
> >>> University of Florida
> >>> (352) 392-0171 Ext. 221 
> >>>
> >>>  
> >>>
> >>>   
> >>>   
> >>>> -Original Message-
> >>>> From: [EMAIL PROTECTED] 
> >>>> [mailto:[EMAIL PROTECTED] On Behalf Of 
> >>>> Robert Boardman
> >>>> Sent: Thursday, November 20, 2008 5:56 PM
> >>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >>>> Subject: [asterisk-users] ISDN Cause codes
> >>>>
> >>>> Hi All
> >>>>
> >>>> Just been looking at stats for one of my sites, and I'm 
> >>>> conserned about 
> >>>> the number of error cause codes being returned from the telco
> >>>>
> >>>> for example
> >>>>
> >>>> 12000 calls processed
> >>>>
> >>>> 131 are cause code 31* normal. unspecified.*
> >>>>
> >>>> 139 are cause code 28 * invalid number format (address 
> >>>> 
> >> incomplete).*
> >> 
> >>>> 112 are cause code 1 *Unallocated (unassigned) number.
> >>>>
> >>>> *this adds up to about 3% of calls not completing.
> >>>>
> >>>> there are various other codes including 17 busy 34 channel 
> >>>> unavaliable 
> >>>> and 44 requested channel unavaliable, which add up to 
> another 1%.*
> >>>> *
> >>>> the telco says there is no problem with the line, I'm tryi

Re: [asterisk-users] ISDN Cause codes

[Robert Boardman]

I have found that the messages are not played as the hangup cause clears 
down the channel and passed hangup to the other end

should I have progress() before the dial command?

Robb

Martin Smith wrote:
> Hi Robert,
>
> I'd recommend the following options for Dial() so that you corroborate
> operator messages w/ cause codes:
>
>  1. remove R and r - we've found this can supress operator recordings on
> early audio
>  2. likewise, remove m to disable MOH
>
> Also, check the values of DIALSTATUS to compare to HANGUPCAUSE.
>
> Good luck,
>
> Martin Smith, Systems Developer
> [EMAIL PROTECTED]
> Bureau of Economic and Business Research
> University of Florida
> (352) 392-0171 Ext. 221 
>
>  
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] 
>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>> Robert Boardman
>> Sent: Friday, November 21, 2008 3:07 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] ISDN Cause codes
>>
>> Thanks for the reply
>>
>> Could you be a little more specific?
>>
>> Thanks
>> Robb
>>
>> Martin Smith wrote:
>> 
>>> Hi Robert,
>>>
>>> I'd suggest tweaking the Dial() arguments so that you (1) 
>>>   
>> allow early
>> 
>>> audio, (2) don't force it play ringing to the calling party, and (3)
>>> modify any other options to be as relaxed as possible. if 
>>>   
>> you make those
>> 
>>> changes, you'll start hearing the operator message 
>>>   
>> recordings and those
>> 
>>> are sometimes easier to reference against the cause codes.
>>>
>>> Cheers,
>>>
>>>
>>> Martin Smith, Systems Developer
>>> [EMAIL PROTECTED]
>>> Bureau of Economic and Business Research
>>> University of Florida
>>> (352) 392-0171 Ext. 221 
>>>
>>>  
>>>
>>>   
>>>   
>>>> -Original Message-
>>>> From: [EMAIL PROTECTED] 
>>>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>>>> Robert Boardman
>>>> Sent: Thursday, November 20, 2008 5:56 PM
>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>> Subject: [asterisk-users] ISDN Cause codes
>>>>
>>>> Hi All
>>>>
>>>> Just been looking at stats for one of my sites, and I'm 
>>>> conserned about 
>>>> the number of error cause codes being returned from the telco
>>>>
>>>> for example
>>>>
>>>> 12000 calls processed
>>>>
>>>> 131 are cause code 31* normal. unspecified.*
>>>>
>>>> 139 are cause code 28 * invalid number format (address 
>>>> 
>> incomplete).*
>> 
>>>> 112 are cause code 1 *Unallocated (unassigned) number.
>>>>
>>>> *this adds up to about 3% of calls not completing.
>>>>
>>>> there are various other codes including 17 busy 34 channel 
>>>> unavaliable 
>>>> and 44 requested channel unavaliable, which add up to another 1%.*
>>>> *
>>>> the telco says there is no problem with the line, I'm trying to 
>>>> understand what the problem could be
>>>>
>>>> now  alot of calls complete OK so I don't think is my configs
>>>>
>>>> Any advice would be appriciated
>>>>
>>>> Versions
>>>> asterisk 1.4.21.1
>>>> zaptel 1.4.12.1
>>>>
>>>>
>>>> Robb
>>>>
>>>> ___
>>>> -- Bandwidth and Colocation Provided by 
>>>> 
>> http://www.api-digital.com --
>> 
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>> 
>>>> 
>>> ___
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>>> asterisk-users mailing list
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>>>http://lists.digium.com/mailman/listinfo/asterisk-users
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>>>   
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>> 
>
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Re: [asterisk-users] ISDN Cause codes

[Martin Smith]

Hi Robert,

I'd recommend the following options for Dial() so that you corroborate
operator messages w/ cause codes:

 1. remove R and r - we've found this can supress operator recordings on
early audio
 2. likewise, remove m to disable MOH

Also, check the values of DIALSTATUS to compare to HANGUPCAUSE.

Good luck,

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Robert Boardman
> Sent: Friday, November 21, 2008 3:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] ISDN Cause codes
> 
> Thanks for the reply
> 
> Could you be a little more specific?
> 
> Thanks
> Robb
> 
> Martin Smith wrote:
> > Hi Robert,
> >
> > I'd suggest tweaking the Dial() arguments so that you (1) 
> allow early
> > audio, (2) don't force it play ringing to the calling party, and (3)
> > modify any other options to be as relaxed as possible. if 
> you make those
> > changes, you'll start hearing the operator message 
> recordings and those
> > are sometimes easier to reference against the cause codes.
> >
> > Cheers,
> >
> >
> > Martin Smith, Systems Developer
> > [EMAIL PROTECTED]
> > Bureau of Economic and Business Research
> > University of Florida
> > (352) 392-0171 Ext. 221 
> >
> >  
> >
> >   
> >> -Original Message-
> >> From: [EMAIL PROTECTED] 
> >> [mailto:[EMAIL PROTECTED] On Behalf Of 
> >> Robert Boardman
> >> Sent: Thursday, November 20, 2008 5:56 PM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: [asterisk-users] ISDN Cause codes
> >>
> >> Hi All
> >>
> >> Just been looking at stats for one of my sites, and I'm 
> >> conserned about 
> >> the number of error cause codes being returned from the telco
> >>
> >> for example
> >>
> >> 12000 calls processed
> >>
> >> 131 are cause code 31* normal. unspecified.*
> >>
> >> 139 are cause code 28 * invalid number format (address 
> incomplete).*
> >>
> >> 112 are cause code 1 *Unallocated (unassigned) number.
> >>
> >> *this adds up to about 3% of calls not completing.
> >>
> >> there are various other codes including 17 busy 34 channel 
> >> unavaliable 
> >> and 44 requested channel unavaliable, which add up to another 1%.*
> >> *
> >> the telco says there is no problem with the line, I'm trying to 
> >> understand what the problem could be
> >>
> >> now  alot of calls complete OK so I don't think is my configs
> >>
> >> Any advice would be appriciated
> >>
> >> Versions
> >> asterisk 1.4.21.1
> >> zaptel 1.4.12.1
> >>
> >>
> >> Robb
> >>
> >> ___
> >> -- Bandwidth and Colocation Provided by 
> http://www.api-digital.com --
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >> 
> >
> > ___
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> http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >   
> 
> 
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Re: [asterisk-users] ISDN Cause codes

[Robert Boardman]

Thanks for the reply

Could you be a little more specific?

Thanks
Robb

Martin Smith wrote:
> Hi Robert,
>
> I'd suggest tweaking the Dial() arguments so that you (1) allow early
> audio, (2) don't force it play ringing to the calling party, and (3)
> modify any other options to be as relaxed as possible. if you make those
> changes, you'll start hearing the operator message recordings and those
> are sometimes easier to reference against the cause codes.
>
> Cheers,
>
>
> Martin Smith, Systems Developer
> [EMAIL PROTECTED]
> Bureau of Economic and Business Research
> University of Florida
> (352) 392-0171 Ext. 221 
>
>  
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] 
>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>> Robert Boardman
>> Sent: Thursday, November 20, 2008 5:56 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] ISDN Cause codes
>>
>> Hi All
>>
>> Just been looking at stats for one of my sites, and I'm 
>> conserned about 
>> the number of error cause codes being returned from the telco
>>
>> for example
>>
>> 12000 calls processed
>>
>> 131 are cause code 31* normal. unspecified.*
>>
>> 139 are cause code 28 * invalid number format (address incomplete).*
>>
>> 112 are cause code 1 *Unallocated (unassigned) number.
>>
>> *this adds up to about 3% of calls not completing.
>>
>> there are various other codes including 17 busy 34 channel 
>> unavaliable 
>> and 44 requested channel unavaliable, which add up to another 1%.*
>> *
>> the telco says there is no problem with the line, I'm trying to 
>> understand what the problem could be
>>
>> now  alot of calls complete OK so I don't think is my configs
>>
>> Any advice would be appriciated
>>
>> Versions
>> asterisk 1.4.21.1
>> zaptel 1.4.12.1
>>
>>
>> Robb
>>
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>> 
>
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Re: [asterisk-users] ISDN Cause codes

[Martin Smith]

Hi Robert,

I'd suggest tweaking the Dial() arguments so that you (1) allow early
audio, (2) don't force it play ringing to the calling party, and (3)
modify any other options to be as relaxed as possible. if you make those
changes, you'll start hearing the operator message recordings and those
are sometimes easier to reference against the cause codes.

Cheers,


Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Robert Boardman
> Sent: Thursday, November 20, 2008 5:56 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] ISDN Cause codes
> 
> Hi All
> 
> Just been looking at stats for one of my sites, and I'm 
> conserned about 
> the number of error cause codes being returned from the telco
> 
> for example
> 
> 12000 calls processed
> 
> 131 are cause code 31* normal. unspecified.*
> 
> 139 are cause code 28 * invalid number format (address incomplete).*
> 
> 112 are cause code 1 *Unallocated (unassigned) number.
> 
> *this adds up to about 3% of calls not completing.
> 
> there are various other codes including 17 busy 34 channel 
> unavaliable 
> and 44 requested channel unavaliable, which add up to another 1%.*
> *
> the telco says there is no problem with the line, I'm trying to 
> understand what the problem could be
> 
> now  alot of calls complete OK so I don't think is my configs
> 
> Any advice would be appriciated
> 
> Versions
> asterisk 1.4.21.1
> zaptel 1.4.12.1
> 
> 
> Robb
> 
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Re: [asterisk-users] ISDN Cause codes

[Robert Boardman]

Some are mis dialed but most work one day but not the next
they are all dialed manually

Robb

Don Kelly wrote:
> What is the source of the numbers you are calling? Are they
> previously-verified numbers from your database? Are some of them
> fumble-fingered manually-dialed calls? I'm pretty sure that I goof on more
> than 3% of calls that I manually call. Have you researched some of the
> failures (examining the numbers that were attempted to be called)? I don't
> really see a problem with what you're reporting.
>
>   --Don
>
> Don Kelly
> PCF Corp
> Real Support for your Virtual Office TM
> 651 842-1000
> 888 Don Kell(y)
> 651 842-1001 fax
>
>  
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Robert
> Boardman
> Sent: Thursday, November 20, 2008 4:56 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] ISDN Cause codes
>
> Hi All
>
> Just been looking at stats for one of my sites, and I'm conserned about 
> the number of error cause codes being returned from the telco
>
> for example
>
> 12000 calls processed
>
> 131 are cause code 31* normal. unspecified.*
>
> 139 are cause code 28 * invalid number format (address incomplete).*
>
> 112 are cause code 1 *Unallocated (unassigned) number.
>
> *this adds up to about 3% of calls not completing.
>
> there are various other codes including 17 busy 34 channel unavaliable 
> and 44 requested channel unavaliable, which add up to another 1%.*
> *
> the telco says there is no problem with the line, I'm trying to 
> understand what the problem could be
>
> now  alot of calls complete OK so I don't think is my configs
>
> Any advice would be appriciated
>
> Versions
> asterisk 1.4.21.1
> zaptel 1.4.12.1
>
>
> Robb
>
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Re: [asterisk-users] ISDN Cause codes

[Don Kelly]

What is the source of the numbers you are calling? Are they
previously-verified numbers from your database? Are some of them
fumble-fingered manually-dialed calls? I'm pretty sure that I goof on more
than 3% of calls that I manually call. Have you researched some of the
failures (examining the numbers that were attempted to be called)? I don't
really see a problem with what you're reporting.

  --Don

Don Kelly
PCF Corp
Real Support for your Virtual Office TM
651 842-1000
888 Don Kell(y)
651 842-1001 fax

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Boardman
Sent: Thursday, November 20, 2008 4:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN Cause codes

Hi All

Just been looking at stats for one of my sites, and I'm conserned about 
the number of error cause codes being returned from the telco

for example

12000 calls processed

131 are cause code 31* normal. unspecified.*

139 are cause code 28 * invalid number format (address incomplete).*

112 are cause code 1 *Unallocated (unassigned) number.

*this adds up to about 3% of calls not completing.

there are various other codes including 17 busy 34 channel unavaliable 
and 44 requested channel unavaliable, which add up to another 1%.*
*
the telco says there is no problem with the line, I'm trying to 
understand what the problem could be

now  alot of calls complete OK so I don't think is my configs

Any advice would be appriciated

Versions
asterisk 1.4.21.1
zaptel 1.4.12.1


Robb

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Re: [asterisk-users] ISDN Cause Code 100, Bosch Integral Management Connection

[Johann Steinwendtner]

Wolfgang Pichler wrote:
> Hi all,
> 

> we have the following setup
> 
> PSTN 3 PRI Lines   <--->  Asterisk (1.4.22)   <--->  Siemens  HiCom   
> <--->   Bosch Integral
> 
> The Asterisk Machine does play the man in the middle - and adds some 
> extra functionality to the system (SIP users...) - the normal calls are 
> getting 1:1 through the system (incoming calls from PSTN are handled by 
> a simple Dial(ZAP/g1/${EXTEN}) (g1 = Siemens side) - so no special 
> handling here...
> 
> Everything is working as it should - beside of one little thing. The 
> Bosch Integral PBX does have a special extension (99) which is used to 
> remote manage the machine - this managment connection is working fine 
> without asterisk, as soon as asterisk is connected in the middle the 
> management connection wont work any more - getting back isdn cause code 
> 100. I have already tried dial options d und c (make it digital - clear 
> channel) - no success.
> 
Can you also post the incoming setup message to your asterisk system ?
They should be almost identical.

Best regards

Hans




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Re: [asterisk-users] ISDN - BRI

[Tzafrir Cohen]

On Thu, Oct 30, 2008 at 12:44:40PM -0600, Wilton Helm wrote:
> Subsequent to some previous E-Mails, I've been trying to dig into the ISDN - 
> BRI situation a bit more.  I have determined that I have a HFC card with 
> Winbond chip, but I'm not sure what combination of drivers is best or usable.
> 
> zaphfc is out because it only supports the cologne chip.

The HFC chip is "the cologne chip".

The HFC-S chip is used in various common single-port ISDN cards.

> 
> misdn is a possibility.  I haven't determined if it supports the card 
> natively, or needs a card specific driver under it.
> 
> capi is a possibility, but again, I don't know what driver, if any needs to 
> be under it.
> 
> capi can support misdn under it, but I don't know if this is an advantage or 
> not, and again whether a card driver needs to be under misdn
> 
> libpri 1.4.4 is supposed to work, provided you unpatch the bad patch in the 
> source and compile it--again, I'm not sure what driver, if any needs to be 
> under it.

This is if you have zaphfc.

> 
> F9 detected the card and loaded some sort of driver support for it, but I 
> don't know if that covers the lower levels appropriately.
> 
> Can anyone provide more specific information or suggest which of these 
> approaches is most likely to work, most likely to be stable, or supported for 
> the future, or support the most features?  It would appear there are at least 
> four possible ways that might work, but I can't determine which is best.
> 
> Thanks,
> Wilton

-- 
   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] ISDN PRI Caller ID problem

[A.R. Nasir Qureshi]

Dear Matthew,

Thank you for your reply.

Please tell me if there is any way I can see the actual q931 message received 
from the card without any translation / filtration / alteration by the software 
or the driver ?

How much the switch type or other configuration variables affect the SETUP 
message being received ? What I am concerned is that may be the wrong switch 
type or other parameter may be causing the problem.

This is my first attempt at configuring an ISDN, and I want to be sure before I 
go after the telco.

--
Regards,


Nasir.

A.R. Nasir Qureshi wrote:


> Dear All,
> 
> I am trying to setup an ISDN line from local telco on a digium card. The 
> problem I am facing is that I am not getting any caller id from the 
> telco. They say that they have enabled caller id.
  


Dear Matthew,

Thank you for your reply.

Please tell me if there is any way I can see the actual q931 message received 
from the card without any translation / filtration / alteration by the software 
or the driver ?

How much the switch type or other configuration variables affect the SETUP 
message ?

This is my first attempt at configuring an ISDN, and I want to be sure before I 
go after the telco.


Tell them they are wrong.  There is no calling party number IE in that 
SETUP message below.  :-) 


Matthew Fredrickson
Digium, Inc.


> 
> Please help me out.
> 
> My zapata.conf

> 

> [trunkgroups]
> 
> [channels]

> context=pstnincoming
> pridialplan=local
> prilocaldialplan=local
> 
> usecallerid=yes

> cidsignalling=v23
> cidstart=ring
> hidecallerid=no
> callwaiting=no
> usecallingpres=yes
> sendcalleridafter=1
> echocancel=no
> echocancelwhenbridged=no
> rxgain=0.0
> txgain=0.0
> callgroup=1
> pickupgroup=1
> 
> immediate=no

> callerid=asreceived
> busydetect=no
> busycount=6
> callprogress=no
> faxdetect=incoming
> 
> 
> switchtype = national

> signalling = pri_cpe
> group = 1
> channel => 1-15,17-31
> channel => 32-46,48-62
> 

> 
> The information I get from using "pri intense debug span 1" is:

> 

> < [ 02 01 16 9c 08 02 15 01 05 04 03 80 90 a3 18 03 a1 83 81 70 08 c1 34 
> 33 39 32 38 34 32 a1 ]
> 
> < Informational frame:

> < SAPI: 00  C/R: 1 EA: 0
> <  TEI: 000EA: 1
> < N(S): 011   0: 0
> < N(R): 078   P: 0
> < 26 bytes of data
> Handling message for SAPI/TEI=0/0
> -- ACKing all packets from 77 to (but not including) 78
> -- Since there was nothing left, stopping T200 counter
> -- Stopping T203 counter since we got an ACK
> -- Nothing left, starting T203 counter
> < Protocol Discriminator: Q.931 (8)  len=26
> < Call Ref: len= 2 (reference 5377/0x1501) (Originator)
> < Message type: SETUP (5)
> < [04 03 80 90 a3]
> < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
> capability: Speech (0)
> <  Ext: 1  Trans mode/rate: 64kbps, 
> circuit-mode (16)

>  < [18 03 a1 83 81]
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  
> Preferred  Dchan: 0

>  <   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
> <   Ext: 1  Channel: 1 ]
> < [70 08 c1 34 33 39 32 38 34 32]
> < Called Number (len=10) [ Ext: 1  TON: Subscriber Number (4)  NPI: 
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '4392842' ]

> < [a1]
> < Sending Complete (len= 1)
> -- Making new call for cr 5377
> -- Processing Q.931 Call Setup
> -- Processing IE 4 (cs0, Bearer Capability)
> -- Processing IE 24 (cs0, Channel Identification)
> -- Processing IE 112 (cs0, Called Party Number)
> -- Processing IE 161 (cs0, Sending Complete)
> q931.c:3509 q931_receive: call 5377 on channel 1 enters state 6 (Call 
> Present)

> Sending Receiver Ready (12)
> 
> 
> 
  

--
Regards,


Nasir.

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Re: [asterisk-users] ISDN PRI Caller ID problem

[Matthew Fredrickson]

A.R. Nasir Qureshi wrote:
> Dear All,
> 
> I am trying to setup an ISDN line from local telco on a digium card. The 
> problem I am facing is that I am not getting any caller id from the 
> telco. They say that they have enabled caller id.

Tell them they are wrong.  There is no calling party number IE in that 
SETUP message below. :-)

Matthew Fredrickson
Digium, Inc.

> 
> Please help me out.
> 
> My zapata.conf
> 
> [trunkgroups]
> 
> [channels]
> context=pstnincoming
> pridialplan=local
> prilocaldialplan=local
> 
> usecallerid=yes
> cidsignalling=v23
> cidstart=ring
> hidecallerid=no
> callwaiting=no
> usecallingpres=yes
> sendcalleridafter=1
> echocancel=no
> echocancelwhenbridged=no
> rxgain=0.0
> txgain=0.0
> callgroup=1
> pickupgroup=1
> 
> immediate=no
> callerid=asreceived
> busydetect=no
> busycount=6
> callprogress=no
> faxdetect=incoming
> 
> 
> switchtype = national
> signalling = pri_cpe
> group = 1
> channel => 1-15,17-31
> channel => 32-46,48-62
> 
> 
> The information I get from using "pri intense debug span 1" is:
> 
> < [ 02 01 16 9c 08 02 15 01 05 04 03 80 90 a3 18 03 a1 83 81 70 08 c1 34 
> 33 39 32 38 34 32 a1 ]
> 
> < Informational frame:
> < SAPI: 00  C/R: 1 EA: 0
> <  TEI: 000EA: 1
> < N(S): 011   0: 0
> < N(R): 078   P: 0
> < 26 bytes of data
> Handling message for SAPI/TEI=0/0
> -- ACKing all packets from 77 to (but not including) 78
> -- Since there was nothing left, stopping T200 counter
> -- Stopping T203 counter since we got an ACK
> -- Nothing left, starting T203 counter
> < Protocol Discriminator: Q.931 (8)  len=26
> < Call Ref: len= 2 (reference 5377/0x1501) (Originator)
> < Message type: SETUP (5)
> < [04 03 80 90 a3]
> < Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
> capability: Speech (0)
> <  Ext: 1  Trans mode/rate: 64kbps, 
> circuit-mode (16)
>  < [18 03 a1 83 81]
> < Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  
> Preferred  Dchan: 0
>  <   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
> <   Ext: 1  Channel: 1 ]
> < [70 08 c1 34 33 39 32 38 34 32]
> < Called Number (len=10) [ Ext: 1  TON: Subscriber Number (4)  NPI: 
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '4392842' ]
> < [a1]
> < Sending Complete (len= 1)
> -- Making new call for cr 5377
> -- Processing Q.931 Call Setup
> -- Processing IE 4 (cs0, Bearer Capability)
> -- Processing IE 24 (cs0, Channel Identification)
> -- Processing IE 112 (cs0, Called Party Number)
> -- Processing IE 161 (cs0, Sending Complete)
> q931.c:3509 q931_receive: call 5377 on channel 1 enters state 6 (Call 
> Present)
> Sending Receiver Ready (12)
> 
> 
> 


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Re: [asterisk-users] ISDN

[Steve Totaro]

On Tue, Oct 14, 2008 at 4:59 PM, Steve Totaro <
[EMAIL PROTECTED]> wrote:

>
>
> On Mon, Oct 13, 2008 at 7:52 PM, Tzafrir Cohen <[EMAIL PROTECTED]>wrote:
>
>> On Mon, Oct 13, 2008 at 04:55:01PM -0500, Michael Graves wrote:
>> > I had converations with both Pika and Xorcom wherein the thought that
>> > it should be possible using their interface hardware. There might be
>> > some software changes to be made in their drivers, but BRI should be
>> > usable in the US.
>>
>> Or actually: I suppose that now that Asterisk finally knows that spans
>> can be of size 2B+1D (as of 1.4.22), chan_dahdi will support US BRI will
>> with any BRI device that has a Zaptel/DAHDI driver. That should be
>> either ours (Xorcom) BRI module of the Astribank, Junghanns
>> quad/octo/duo BRI cards and compatible, or the simple HFC-S -based
>> single port PCI cards. Sangoma A500 cards should have Zaptel drivers as
>> well.
>>
>> However this is based on quite a few cases and mostly remains to be
>> tested.
>>
>> (The relevant patch to Asterisk is trivial to apply to earlier versions
>> of 1.4 / 1.2)
>>
>>
> Can you define the SPIDs and what to do with them, otherwise it is fairly
> useless.
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>

In addition, do the BRI cards/drivers use the US Standards for BRI or the
R.O.W (fest of the world)?

http://www.smallnetbuilder.com/content/view/30444/84/
Written by Michael Graves, May 20th 2008

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] ISDN

[Steve Totaro]

On Mon, Oct 13, 2008 at 7:52 PM, Tzafrir Cohen <[EMAIL PROTECTED]>wrote:

> On Mon, Oct 13, 2008 at 04:55:01PM -0500, Michael Graves wrote:
> > I had converations with both Pika and Xorcom wherein the thought that
> > it should be possible using their interface hardware. There might be
> > some software changes to be made in their drivers, but BRI should be
> > usable in the US.
>
> Or actually: I suppose that now that Asterisk finally knows that spans
> can be of size 2B+1D (as of 1.4.22), chan_dahdi will support US BRI will
> with any BRI device that has a Zaptel/DAHDI driver. That should be
> either ours (Xorcom) BRI module of the Astribank, Junghanns
> quad/octo/duo BRI cards and compatible, or the simple HFC-S -based
> single port PCI cards. Sangoma A500 cards should have Zaptel drivers as
> well.
>
> However this is based on quite a few cases and mostly remains to be
> tested.
>
> (The relevant patch to Asterisk is trivial to apply to earlier versions
> of 1.4 / 1.2)
>
>
Can you define the SPIDs and what to do with them, otherwise it is fairly
useless.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] ISDN

[Wilton Helm]

>There's a HFC-S winbond card. How does that card show on lspci?


Network controller: Dynalink IS64PH [0675:1702] ISDN Adapter
kernel modules: hisax

Does that tell you anything useful?  Do you want more details?  Would you 
like to borrow one for a while (I have two)?

Wilton 


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Re: [asterisk-users] ISDN

[Wilton Helm]

>Why didn't BRI catch-on in the US?
I a word--greed.  It arrived shortly after divestiture when there was a lot of 
competition in the market and a dozen independent regional telcos.  Apparently 
they saw a huge cash cow for this data service and yet another competitive 
advantage to proprietary implementation details.  I used to live in GTE 
territory (now Verizon) and they were charging 2 cents per minute of connect 
time!  This was on top of monthly fees that were unreasonable to begin with.  I 
don't know if they have all caught on now and fixed it, but its too late now 
because most of the US has DSL which is 10x the data rate and supports VoIP for 
voice.

I'm fortunate here that Qwest has offered it for at least 10 years at rates 
comparable to two POTS lines and no per minute charges.  When I started using 
it, in my situation I got more features for less money than two POTS lines, 
which I would have needed instead.

The irony is that the direct cost (equipment) of ISDN is less per B channel 
than POTS.  Anyone who has ever compared the cost of digital versus analog 
station cards for a PABX, knows painfully the cost of supporting A/D 
conversion, 90 V 20 Hz ringing and even DTMF registers.  If they had wanted to 
the telcos could have made ISDN cheaper than POTS, still made money and moved 
technology forward in the process.

But to illustrate the mentality, GTE, who I mentioned above was serving the 
affluent community I was in with stepper switches in the CO until well into the 
1980s!  They put tone to pulse converters in front of them (initially for an 
extra fee) so they could support DTMF.  Ironically they only put 600 ms 
interdigit time in the converters and the steppers could take up to 800 ms to 
find a link for the next digit, so the call failure rate ran 10 - 50%!  I ran a 
PABX at the time and we did our own tone to pulse conversion just to avoid 
that.  With 800 ms interdigit time we had at least 99% completion!

Wilton
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Re: [asterisk-users] ISDN

[Joe Greco]

> On Mon, 13 Oct 2008, Steve Totaro wrote:
> > I have done this.  Why BRIs exist in the US is beyond me.  If you can, 
> > don't go with BRI.
> 
> Why didn't BRI catch-on in the US?

Stupidity.

Okay, well, many reasons.

It was targetted as a business service, and the pricing models (at least
locally) didn't offer a reasonable residential offering until ... I'm
thinking almost the mid 1990's (1994-1995).  That's when I first recall
hearing about the availability of residential BRI that didn't charge by
the minute, I think.

The telcos had this huge amount of legacy plant installed, much of which
wasn't particularly great copper.  Good enough for voice.  There was also
lots of CO and remote equipment for which BRI simply wasn't available.  I
fondly remember Ameritech running BRI lines out of MILWWI10, using a bunch
of repeaters, because they didn't have the capabilities (or capacity, can't
recall for sure) at MILWWI45, 

(you know what's interesting, Google Maps knows CLLI codes!)

which resulted in about 11 extra miles on a BRI circuit, and I guess it was
much worse in wire miles.  It's useful to remember that the phone company
expects their equipment to be good for decades, and so there's a huge
amount of resistance to upgrading "just for newfangled data services."

Now, I know that there was this huge "vision" in the '60's and '70's of the
possibility of things like videophones, and ISDN might well have been the
ideal platform for delivering something like that, but the flip side is
that the Carterfone decision resulted in a booming non-Ma-Bell CPE business
and Ma Bell mostly realizing that they were getting shut out of that market.

Now, prior to that point, you had a situation where it was Only The Phone
Guy who would bring CPE to your house and hook it up, you might dare to
move the phone line or jack, but usually not.  This meant that a service
that was more complex to provision would still be relatively easy for Ma
Bell to deploy, since it was just training and equipping their techs that
was important.  The end user would have had no idea what the underlying
technology was.

After that point, though, it would have been really hard to sell a
residential BRI, unless you had equipment capable of automatic
configuration, because it's hard enough for Joe Sixpack to plug in a
POTS line correctly as it is.  Configuring more-complex stuff, especially
in the days before nifty little GUI interfaces (which requires electronic
capabilities not really present until recent years) would have been rough.

Despite all this, there was a renaissance with BRI in the 1990's.  We had
reached a point where the electronics were reasonable.  The Internet did
not yet exist for most people, and modem technology was 9600 or 14.4.  As
corporate networking and the Internet exploded, there was a willingness to
pay premium prices for ISDN gear that would allow relatively inexpensive
BRI circuits to attach at speeds far beyond POTS.

Then we saw that fall apart, as speeds increased to 28.8 and then 56K,
and for most users, that was close enough.  DSL was right on the heels of
that, offering greater-than-ISDN speeds.  ISDN BRI was back to dead status
by about 2000.  You can *see* this in terms of CPE devices that supported
ISDN BRI.

In the meantime, the ILEC's began to truly understand the difference
between switched circuit and packet data services.  Many people had been
using ISDN BRI as a faster and more flexible alternative to 56K DDS 
lines, which tended to tie up switch capacity.  More people were ordering
"second lines," and leaving them connected to local ISP's for hours at a
time, which created trunk, network, and switch capacity challenges for
the ILEC's.  This was devastating to the ILEC's, which typically plan
capital expenditures to be good for many years, but in this case, I have
to assume that the ILEC planners knew that Internet would be provided
over circuits other than their switched POTS/DS0's in the near future,
which would dump capacity requirements back down.  They even got smacked
harder than maybe they expected, as some people gave up land lines
entirely, in favor of cell...

In the meantime, many of the major "data" uses that had been envisioned
for ISDN BRI have been done, better, cheaper, on the Internet.
Videophone?  Easy on a PC with speakers, mic, and an Internet connection,
but hopelessly challenging to someone with a POTS or BRI line.  Private
network interconnection?  VPN over Internet.  Etc.

This has meant that BRI has "evolved" towards a way to deliver telephone
network with no loss, or, rather, most of the envisioned benefits are no
longer likely to be exploited via BRI.  So you don't see many BRI lines,
and it is pretty common for those that you do see to be hooked up to a
PBX, automated call handling system, etc.

Further, telco departments to handle BRI's have suffered mightily.  We had
a problem maybe a year ago where we had suddenly "broken" and were unable
to dial... local?  ld?  I can't 

Re: [asterisk-users] ISDN

[Joe Greco]

> >With ISDN, the conversion is done in your phone
>
> Exactly.  Or in the case of Asterisk, it is a 4 wire digital right into =
> the switch--no degradation.  Even converting back and forth between =
> analog and digital multiple times compromises quality.  Try doing a =
> dial-up modem across such a path.  The best you will get is 20 - 30 K.

A single D/A hop destroys the ability to do 56k.  Successful 56k requires
that there be a single A/D hop at the far end (the user's POTS interface)
and then digital delivery of signal the remainder of the way to the
terminating equipment. (modem -> phone co A/D -> digital to ISP modem
bank).

If you stick an extra D/A (maybe plus A/D) transformation in there, you 
will probably get fairly clean speeds in the upper ranges of 28.8-33.6,
but that'll be it (modem -> phone co A/D -> digital network -> phone co
D/A -> your buddy's modem).  If you're unlucky enough to get some crummy
phone co arrangement where they punt you back and forth from digital to
analog and back to digital within their network, that's even worse.

>From an Asterisk point of view, it's interesting that you can get digital
delivery of the signal, and route the signal around internally digitally,
if you have ISDN.  This means, for example, that our USR Courier I-Modem,
which can terminate a 56K call *digitally*, results in my being able to
make a 56K connection from most modern cities here in the US, without
wasting an ILEC ISDN BRI line dedicated to that purpose, by having the 
PBX connect an extension to the I-Modem.  I just dial into a general
purpose number, and dial the appropriate extension, and voila, I'm on our
network at "high speed."

This is clearly obvious to you, but I thought I'd expand for the others
who might be reading along and didn't understand the implications of all-
digital.

> >IF you can get a PRI-line for the same price.
>
> Not to mention that the interfaces for PRI are about five times as =
> expensive.  I'm not sure why.  It doesn't seem like it ought to take a =
> lot of electronics to break down the bit stream.

It may not take a lot of electronics.  However, the sad truth of it all is
that any electronic device produced at low volume tends to be expensive to
produce.  This is largely the result of costs such as retooling, and in
most cases the significantly higher cost of small-run integrated circuits.

For example, a PC board manufacturing house (I'll use the following shop,
no affiliation, as an example, because they have transparent pricing)
http://www.expresspcb.com/ExpressPCBHtm/Specs4LayerStandard.htm
http://www.expresspcb.com/ExpressPCBHtm/Specs4LayerProduction.htm
for a 30 sq in board.  To produce 10 boards would cost $404, or $40/board.
To produce 50 boards would cost $1109, or $22/board.  To produce 1000 
boards would cost $15516, or $15/board.  Even 1000 isn't really a large
run, though.  You're paying premium board rates for small runs, because
the shop has to stop and retool for your run.  I haven't bothered to get
a large-run quote, but I bet you can get that down to well under $10/board
if you're ordering a hundred thousand at a time...

You then have to add on assembly costs, which are typically higher than
the PCB costs.  It could very easily end up costing $50/board *just* for
PCB and assembly, no parts included, for runs in the hundreds of cards
range. 

The problem with telephony stuff, especially in this market, will be that
the demand for a T1(/PRI/etc) interface is going to be very low.  You would
need to be a relatively big shop to be able to buy by the thousand, as even
at one bulk buy per year, that translates to several cards departing 
inventory daily.

I expect that some of the ISDN BRI interfaces are dirt cheap because they're
popular over in Europe.  I've been told that in many places, they're sold in
lieu of a modem.  Once you are moving product in high volumes, the pricing
tends to come down.

It stinks, yes.

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.

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Re: [asterisk-users] ISDN

[Gordon Henderson]

On Mon, 13 Oct 2008, Steve Totaro wrote:

> I have done this.  Why BRIs exist in the US is beyond me.  If you can, 
> don't go with BRI.

Why didn't BRI catch-on in the US? It's been in-use in the UK and Europe 
for a long time (especially Germany AIUI). I have several sites with 
ISDN2e (BRI) in the UK. In some locations it was cheaper to run in more 
ISDN2e lines than get BT to provide an ISDN30e (BT 'forcing' the customer 
to pay for the fibre, but they'd happily run ISDN2e over copper). One site 
has 8 ISDN2e ports (16 channels), although I think only 6 are "lit" right 
now.

I use mISDN FWIW.

Gordon

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Re: [asterisk-users] ISDN

[Tzafrir Cohen]

On Mon, Oct 13, 2008 at 11:53:33PM +0200, Hans Witvliet wrote:
> On Mon, 2008-10-13 at 17:37 -0400, Steve Totaro wrote:
> > I have done this.  Why BRIs exist in the US is beyond me. 
> Much of the idea's behind ISDN are hopelesly outdated, except for one:

BRI is actually ISDN BRI. PRI is ISDN PRI. Uses very much the same
concepts, standards, etc. BRI is supported in chan_zap/chan_dahdi 
through libpri :-)

In most parts of the world BRI adds some extra layers of complication to
make the CPE units behave a bit more like analog phones (PtMP). In the
US, AFAIK, this is not used.

Again I'll note that I have met very few cases, so I might be wrong.

-- 
   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] ISDN

[Tzafrir Cohen]

On Mon, Oct 13, 2008 at 09:44:30PM -0600, Wilton Helm wrote:
> The card I have has no name but is based on the Winbond W6692CF 
> chip and ships with RVS, which I think is for Windows and of no 
> use to me.  I'm not sure about whether it is supported by DAHDI or not.

There's a HFC-S winbond card. How does that card show on lspci?

I started adapting the driver of zaphfc ot dahdi. And got to the point
that I have a compiling but oopsing driver due to the fact that I don't
have such a card myself.

-- 
   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] ISDN

[Wilton Helm]

The card I have has no name but is based on the Winbond W6692CF chip and ships 
with RVS, which I think is for Windows and of no use to me.  I'm not sure about 
whether it is supported by DAHDI or not.

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Re: [asterisk-users] ISDN

[Wilton Helm]

>I have done this.
Good

>Why BRIs exist in the US is beyond me.
I'm not sure why you say that.  It is the only way I know of two get two 
digital voice grade circuits at prices competitive with POTS.  The better 
question is why the LECs used such poor judgment when they introduced this.  
Most were charging outrageous prices that had no technical justification and 
per minute usage fees.  They destroyed the market before it got off the ground. 
 I don't use it for data any more now that I have DSL, but it is still a very 
viable voice channel.

  If you can, don't go with BRI.
Why?  (Not that I don't already have it and have been using it for seven years.)

Who is the carrier. 
Qwest (formerly US West or US Worst as some used to call it).

So do you have some information about what you did or where to get 
configuration information?

Wilton
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Re: [asterisk-users] ISDN

[Wilton Helm]

>With ISDN, the conversion is done in your phone
Exactly.  Or in the case of Asterisk, it is a 4 wire digital right into the 
switch--no degradation.  Even converting back and forth between analog and 
digital multiple times compromises quality.  Try doing a dial-up modem across 
such a path.  The best you will get is 20 - 30 K.

>IF you can get a PRI-line for the same price.
Not to mention that the interfaces for PRI are about five times as expensive.  
I'm not sure why.  It doesn't seem like it ought to take a lot of electronics 
to break down the bit stream.



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Re: [asterisk-users] ISDN

[Tzafrir Cohen]

On Mon, Oct 13, 2008 at 04:55:01PM -0500, Michael Graves wrote:
> I had converations with both Pika and Xorcom wherein the thought that
> it should be possible using their interface hardware. There might be
> some software changes to be made in their drivers, but BRI should be
> usable in the US.

Or actually: I suppose that now that Asterisk finally knows that spans
can be of size 2B+1D (as of 1.4.22), chan_dahdi will support US BRI will 
with any BRI device that has a Zaptel/DAHDI driver. That should be
either ours (Xorcom) BRI module of the Astribank, Junghanns
quad/octo/duo BRI cards and compatible, or the simple HFC-S -based
single port PCI cards. Sangoma A500 cards should have Zaptel drivers as
well.

However this is based on quite a few cases and mostly remains to be
tested.

(The relevant patch to Asterisk is trivial to apply to earlier versions
of 1.4 / 1.2)

-- 
   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] ISDN

[Joe Greco]

> I'm in the process of setting up Asterisk in a SOHO environment using =
> ISDN for trunking.  More specifically a BRI 2B+D circuit where one SPID =
> is used for the business and the other is used for personal.  The =
> circuit already exists, but is presently being interfaced to POTS phones =
> via a TA.
> 
> This configuration is not very common in the US, but we are fortunate =
> that our LEC offers it price competitively with equivalent POTS services =
> and it makes more sense, both in terms of voice quality (4 wire digital =
> to the PABX) and flexibility.
> 
> Ideally it would allow any combination of two calls, identified by SPID.
> 
> If anyone has done anything similar, or has any experience with BRI =
> ISDN, I would appreciate input and direction.
> 
> If anyone knows where documentation exists on configuring ISDN, that =
> information would also be greatly appreciated.  Asterisk has a bit of a =
> learning curve, and ISDN BRI isn't the most widely used or covered =
> aspect of it.  BTW, I have a strong telecom background, so the theory =
> part of it will not be a problem, only the necessary documentation to =
> apply it to Asterisk.

The one solution I've heard, on and off again, that works with Asterisk
here in the US is the Eicon Diva cards.

There are other solutions.  Where I am, we're unreasonably close to a 
local radio conglomerate that has a number of high power antennas.  We
found early on that RF interference was a killer, which caused me to run
a lot of our telecom and data wiring in conduit.

Unfortunately, we discovered that POTS lines were a hell of a mess when
connected to anything more complex than a phone or two.  Lots of RF
interference.  Church radio music on Sundays, even.  So, we brought our
lines in on BRI, which we've used for data and voice elsewhere.

Being eternally frustrated with the lack of ISDN support after maybe 2000
here in the US (we have a bunch of interesting ISDN gear from the 90's!),
I set out to see what I could do to interface BRI to Asterisk.  I *didn't*
go the Eicon route, because at the time it was considered relatively
unreliable.

Instead, we picked up an Adtran Atlas 550, which can handle ISDN BRI, PRI,
POTS, etc.  We have been using the Atlas as a translator to convert BRI
to T1, which works moderately well, but we've seen some issues, mostly in
the capabilities of the Adtran (such as an inability to select the desired
SPID/DN for outgoing calls).

The Adtran has some other amazing capabilities, such as providing FXO/FXS
ports, and even ISDN BRI ports for other devices we'd liked hooked into
our PBX.

Despite that, I'd love to see an ISDN BRI solution for the US.  I might
be willing to test the Eicon Diva Server card...  hm.

... JG
-- 
Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net
"We call it the 'one bite at the apple' rule. Give me one chance [and] then I
won't contact you again." - Direct Marketing Ass'n position on e-mail spam(CNN)
With 24 million small businesses in the US alone, that's way too many apples.

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Re: [asterisk-users] ISDN

[Steve Totaro]

The documentation is in my head, two solid days worth.

The issue is the SPID code that Marcin Pyco claimed he had the only code,
and way to make it work in the US..

You may need this "code" if you are using SPIDs to route calls.  In my
situation, they were just a hunt group, two BRIs, and I was tasked with
adding a quad port Sangoma analog card.  Absolutely NO difference in audio,
but talk about a mish mash of equipment.  Luckily  Sangoma drivers for
Zaptel 1.4 do not require Zaptel to be patched.

It absolutely refused to work with 1.2 so it became my first 1.4
installation out of necessity, I am sure 1.2 didn't work because of
conflicting patches (BRIStuff and Sangoma)

That is why Xorcom was so happy to help me with a US BRI, and I just thought
Tzafrir was a nice guy trying to help out...

Marcin Pyco claimed that BRI would not work without his code in the US and
went so far as to call me a liar.

I proved him wrong, but he is not very good at admitting he is wrong, he
blamed Verizon rather than apologizing.  He is very good at calling people
liars but not so good at apologizing and admitting he is wrong.

Whatever the rub, using BRIStuff, Zaptel 1.4.X and a Junghanns' card or
knock-off (and even Sangoma drivers), it will work with Verizon.

I have pages upon pages of all the emails and IRC chats where I am called a
liar, and where Tzafrir admits his true motives (to his credit).

And finally the revelation that you do not need any additional code for
SPIDs (at least with Verizon) in the US, and around here everyone resells
Verizon anyways.

One thing to note is that inbound calls work immediately when the spans come
up BUT it takes ten to fifteen minutes for outbound calls to work.

I am not sure if the time starts at loading qozap or Asterisk but it works
beyond a shadow of a doubt, so don't pay for "code" that makes it work.

I am convinced that the conversations you had with Xorcom and probably Pika
(since Marcin works or worked there (LinkedIN)) came as a direct result of
my work.

Anyways, in this area, everything is close to a CO and I BET that calling
from a regular phone, you could never guess which is ISDN and which is POTS,
unless you cheat somehow, but not by voice quality.  I am not sure why OP
thinks that two pair for voice is better than two unless he is afraid of
echo, which was absolutely no issue with the Sangoma cards with onboard EC.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

On Mon, Oct 13, 2008 at 5:55 PM, Michael Graves <[EMAIL PROTECTED]> wrote:

> I had converations with both Pika and Xorcom wherein the thought that it
> should be possible using their interface hardware. There might be some
> software changes to be made in their drivers, but BRI should be usable in
> the US.
>
> I abandoned the idea for being more expensive when all costs are
> considered.
>
> Michael
>
> --Original Message Text---
> *From:* Steve Totaro
> *Date:* Mon, 13 Oct 2008 17:37:37 -0400
>
>
> I have done this. Why BRIs exist in the US is beyond me. If you can, don't
> go with BRI.
>
> Who is the carrier. There is someone on the list that will tell you it is
> impossible unless you use his code, which is not true.
>
> Thanks,
> Steve Totaro
>
> On Mon, Oct 13, 2008 at 4:57 PM, Michael Graves <[EMAIL PROTECTED]>
> wrote:
> I had considered something like this as well, but was convinced to go
> another direction.
>
> I wrote something up about it at the time.
>
> *http://www.smallnetbuilder.com/content/view/30444/84/*
>
> Michael
>
>
>
> --Original Message Text---
> *From:* Wilton Helm
> *Date:* Mon, 13 Oct 2008 14:44:26 -0600
>
>
> Hi,
>
> I'm in the process of setting up Asterisk in a SOHO environment using ISDN
> for trunking. More specifically a BRI 2B+D circuit where one SPID is used
> for the business and the other is used for personal. The circuit already
> exists, but is presently being interfaced to POTS phones via a TA.
>
> This configuration is not very common in the US, but we are fortunate that
> our LEC offers it price competitively with equivalent POTS services and it
> makes more sense, both in terms of voice quality (4 wire digital to the
> PABX) and flexibility.
>
> Ideally it would allow any combination of two calls, identified by SPID.
>
> If anyone has done anything similar, or has any experience with BRI ISDN, I
> would appreciate input and direction.
>
> If anyone knows where documentation exists on configuring ISDN, that
> information would also be greatly appreciated. Asterisk has a bit of a
> learning curve, and ISDN BRI isn't the most widely used or covered aspect of
> it. BTW, I have a strong telecom background, so the theory part of it will
> not be a problem, only the necessary documentation to apply it to Asterisk.
>
> Thanks,
> Wilton Helm
> Embedded System Resources
>
>
>
>
>
> --
> Michael Graves
> mgraves*mstvp.com*
> *http://blog.mgraves.org*
> o713-861-4005
> c713-201-1262
> *sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]>*

Re: [asterisk-users] ISDN

[Michael Graves]

I had converations with both Pika and Xorcom wherein the thought that
it should be possible using their interface hardware. There might be
some software changes to be made in their drivers, but BRI should be
usable in the US.

I abandoned the idea for being more expensive when all costs are
considered.

Michael

--Original Message Text---
From: Steve Totaro
Date: Mon, 13 Oct 2008 17:37:37 -0400

I have done this.  Why BRIs exist in the US is beyond me.  If you can,
don't go with BRI.

Who is the carrier.  There is someone on the list that will tell you it
is impossible unless you use his code, which is not true.

Thanks,
Steve Totaro

On Mon, Oct 13, 2008 at 4:57 PM, Michael Graves <[EMAIL PROTECTED]>
wrote:
I had considered something like this as well, but was convinced to go
another direction. 

I wrote something up about it at the time.

http://www.smallnetbuilder.com/content/view/30444/84/

Michael



--Original Message Text---
From: Wilton Helm
Date: Mon, 13 Oct 2008 14:44:26 -0600


Hi, 

I'm in the process of setting up Asterisk in a SOHO environment using
ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID
is used for the business and the other is used for personal. The
circuit already exists, but is presently being interfaced to POTS
phones via a TA. 

This configuration is not very common in the US, but we are fortunate
that our LEC offers it price competitively with equivalent POTS
services and it makes more sense, both in terms of voice quality (4
wire digital to the PABX) and flexibility. 

Ideally it would allow any combination of two calls, identified by
SPID. 

If anyone has done anything similar, or has any experience with BRI
ISDN, I would appreciate input and direction. 

If anyone knows where documentation exists on configuring ISDN, that
information would also be greatly appreciated. Asterisk has a bit of a
learning curve, and ISDN BRI isn't the most widely used or covered
aspect of it. BTW, I have a strong telecom background, so the theory
part of it will not be a problem, only the necessary documentation to
apply it to Asterisk. 

Thanks, 
Wilton Helm 
Embedded System Resources 





--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245

 
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-- 
Thanks,
Steve Totaro 
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)



--
Michael Graves
mgravesmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245


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