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

[Mark Engelhardt]

Hello,

I am installing Asterisk in a small office with just 4 lines and 8 Extensions. 

I have two choices from my local telco (Fairpoint): 

1) Old School ISDN BRI lines which I would connect to Asterisk with a OpenVOX 
B200P
2) Telco supplied SIP trunks over a service called EDIA which is 1MB ethernet 
over several pair of copper lines. 

The ISDN BRI solution is less than 1/2 the price of the SIP solution. 

Any recommendations? Pitfalls?  

Mark Engelhardt
- in snowy Vermont!








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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

[asterisk-users] ISDN Cause Code 47 Errors

[Dale Noll]

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?
Do I have something setup incorrectly?
Is there any way to get even lower level debugging on the PRI?
Has anyone ever had this problem and if so what was done to resolve it?

Thanks in advance for any insight,
Dale


=
general config info
=
There are two trunk groups for hunting by the provider  TG1 = spans 2 and
4, TG2 = spans 1 and 3
I have two dual port cards in the server. Each card has 1 span from each TG
so the lost of a card will keep the other span operational.  The provider
does the same across VWIC cards.
The spans do NOT use NFAS.


==
My equipment/software:
==
2x Digium TE210 dual T1/E1 with hw echo canceler and timing cable installed.
Asterisk 1.8.12
dahdi 2.5.0.1
libpri 1.4.12

==
Provider equipment:
==
Cisco 2821 with 2  VWIC2-@MFT-T1/E1


===
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 

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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[asterisk-users] ISDN outgoing caller id

[Gergo Csibra]

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?

-- 
Best regards,
 Gergo  mailto:csi...@gmail.com


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

[Dimitar Dimitrov]

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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[asterisk-users] ISDN incoming call disconnected after picking up phone

[Ivo]

Hi,

I have been trying to find solution to this problem by googleing and 
experimenting but without success so you are my last chance.


I have asterisk 1.8.11 installation on Debian squeeze with HFC-S PCI 
ISDN interface cards (dahdi channel, zaphfc kernel driver) , Huawei USB 
3G stick, Siemens Gigaset SIP phones and IAX PC soft phone.


The problem is that when I receive call through ISDN interface and pick 
up (SIP or IAX) phone, line immediately gets disconnected.


This is what I get with pri debug on :

PRI Span: 1 > Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) 
standard (0)  0: 0  Location: Private network serving the local user (1)
PRI Span: 1 >   Ext: 1  Progress 
Description: Called equipment is non-ISDN. (2) ]

PRI Span: 1
PRI Span: 1 < Protocol Discriminator: Q.931 (8)  len=4
PRI Span: 1 < TEI=90 Call Ref: len= 1 (reference 21/0x15) (Sent from 
originator)

PRI Span: 1 < Message Type: CONNECT ACKNOWLEDGE (15)
PRI Span: 1 Received message for call 0x89a4e68 on link 0x88a4238 
TEI/SAPI 90/0
PRI Span: 1 q931.c:8484 post_handle_q931_message: Call 21 enters state 
10 (Active).  Hold state: Idle

PRI Span: 1
PRI Span: 1 < Protocol Discriminator: Q.931 (8)  len=12
PRI Span: 1 < TEI=90 Call Ref: len= 1 (reference 21/0x15) (Sent from 
originator)

PRI Span: 1 < Message Type: DISCONNECT (69)
PRI Span: 1 < [08 02 81 90]
PRI Span: 1 < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 
 Spare: 0  Location: Private network serving the

 local user (1)
PRI Span: 1 <  Ext: 1  Cause: Normal Clearing (16), 
class = Normal Event (1) ]



So, after getting CONNECT ACKNOWLEDGE there is an immediate DISCONNECT 
received from telco.


This happens always if dialed extension is
-- SIP
   exten => ${MSN1},1,Dial(SIP/phone1)
or IAX
   exten => ${MSN1},1,Dial(IAX2/phone2)
or multiple extensions
   exten => ${MSN1},1,Dial(SIP/phone1&Dongle/dongle0/phone3)

but not if it is chan_dongle alone
   exten => ${MSN1},1,Dial(Dongle/dongle0/phone3)


Maybe it is worth mentioning that this problem started after switching 
telco. First one used simple NT device connected to the line and second 
one uses Patton IAD connected to the router.



Can you help me to find a solution to this problem?

Thanks in advance.

Best regards,

Ivo.

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[asterisk-users] ISDN, overlap and open dialing plans (Olivier)

[Mc GRATH Ricardo]

Dear Oliver

Well I never knew PBX could according to numbers pattern length  use enbloc and 
overlap, from my experiences dialling mode setting should be one or other, but 
it should be set on whole system one mode, by the way if number length pattern 
is variable component to use overlap mode.
Just in case when it use enbloc PBX send  whole number,  TDM phones it  press # 
or other key setting as send digits.
By the other way check dialplan rules to resolve receiving number length, a 
good practice is use and Asterisk extension to simulated call from PBX system.
Best regards

Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com

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Today's Topics:

   1. ISDN, overlap and open dialing plans (Olivier)
   2. DAHDI_SPANCONFIG failed on span 1: No such device or address
  (6) (rama...@gmx.de)
   3. Re: Capacity of single instance of Asterisk
  (Amit Patkar | Avhan Technologies Pvt Ltd)
   4. Re: DAHDI_SPANCONFIG failed on span 1: No such device or
  address (6) (Eric Wieling)
   5. Re: Capacity of single instance of Asterisk
  (Amit Patkar | Avhan Technologies Pvt Ltd)
   6. Re: how to show used "wrong password" (Kevin P. Fleming)
   7. Re: DAHDI_SPANCONFIG failed on span 1: No such device or
  address (6) (Shaun Ruffell)
   8. Re: Capacity of single instance of Asterisk (Kevin P. Fleming)
   9. Re: Capacity of single instance of Asterisk (Bryant Zimmerman)
  10. Re: DAHDI_SPANCONFIG failed on span 1: No such device or
  address (6) (Tzafrir Cohen)
  11. Re: DAHDI_SPANCONFIG failed on span 1: No such device or
  address (6) (rama...@gmx.de)
  12. Re: how to show used "wrong password" (Randall)
  13. Re: how to show used "wrong password" (Randall)


--

Message: 1
Date: Tue, 13 Mar 2012 14:37:06 +0100
From: Olivier 
Subject: [asterisk-users] ISDN, overlap and open dialing plans
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID:

Content-Type: text/plain; charset=ISO-8859-1

Hi,

I've got the following setup:

PSTN/ISDN < E1-> Asterisk  < E1-> Alcatel 4400 PBX
<> TDM phones

When a TDM phone is dialing out to a national number, it seems that
the PBX is using enbloc dialing.
When a TDM phone is dialing out to an international number (variable
length numbers), it seems that the PBX is using overlap dialing as
Asterisk is currently receiving truncated numbers.

What is the best way to deal with such situations ?
1. configure PSTN in enbloc dialing and tweak dialplan to mimic
overlap dialing ?
2. or configure both PTSN and PBX spans in overlap mode ?
Suggestions ?

Regards



--

Message: 2
Date: Tue, 13 Mar 2012 15:30:45 +0100
From: rama...@gmx.de
Subject: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No such
device or address (6)
To: asterisk-users@lists.digium.com
Message-ID: <20120313143045.18...@gmx.net>
Content-Type: text/plain; charset="utf-8"

Hi all,

I have problems starting dahdi.
dahdi_cfg -vvv allwasy comes back with:


DAHDI Tools Version - 2.2.1.1

DAHDI Version: 2.3.0.1
Echo Canceller(s):
Configuration
==

SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01)
Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02)
Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) 
(Slaves: 03)

3 channels to configure.

DAHDI_SPANCONFIG failed on span 1: No such device or address (6)







I searched the internet but could not yet find a solution.
I already tried to exchange the zaphfc drivers as suggested, but they did not 
compile.

I actually did not find a new(er) tutorial how to build an Asterisk with a 
HFC-S card.

Any suggestions / hints / tutorials / links welcome.

Do I need some special drivers in the kernel ?
Modprobe ?
Anything else special I need ?

Thanks
Rainer







Details:

I run Debian 6.0.4 with a fresh 2.6.35 Kernel, m

[asterisk-users] ISDN, overlap and open dialing plans

[Olivier]

Hi,

I've got the following setup:

PSTN/ISDN < E1-> Asterisk  < E1-> Alcatel 4400 PBX
<> TDM phones

When a TDM phone is dialing out to a national number, it seems that
the PBX is using enbloc dialing.
When a TDM phone is dialing out to an international number (variable
length numbers), it seems that the PBX is using overlap dialing as
Asterisk is currently receiving truncated numbers.

What is the best way to deal with such situations ?
1. configure PSTN in enbloc dialing and tweak dialplan to mimic
overlap dialing ?
2. or configure both PTSN and PBX spans in overlap mode ?
Suggestions ?

Regards

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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.

-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] ISDN E1, after electrical disconnected is not becoming UP, IRQ misses: 1

[bilal ghayyad]

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

   1 TE2/0/1/1 Clear (In use) RED(SWEC: MG2)
   2 TE2/0/1/2 Clear (In use) RED(SWEC: MG2)
   3 TE2/0/1/3 Clear (In use) RED(SWEC: MG2)
   4 TE2/0/1/4 Clear (In use) RED(SWEC: MG2)
   5 TE2/0/1/5 Clear (In use) RED(SWEC: MG2)
   6 TE2/0/1/6 Clear (In use) RED(SWEC: MG2)
   7 TE2/0/1/7 Clear (In use) RED(SWEC: MG2)
   8 TE2/0/1/8 Clear (In use) RED(SWEC: MG2)
   9 TE2/0/1/9 Clear (In use) RED(SWEC: MG2)
  10 TE2/0/1/10 Clear (In use) RED(SWEC: MG2)
  11 TE2/0/1/11 Clear (In use) RED(SWEC: MG2)
  12 TE2/0/1/12 Clear (In use) RED(SWEC: MG2)
  13 TE2/0/1/13 Clear (In use) RED(SWEC: MG2)
  14 TE2/0/1/14 Clear (In use) RED(SWEC: MG2)
  15 TE2/0/1/15 Clear (In use) RED(SWEC: MG2)
  16 TE2/0/1/16 HDLCFCS (In use) RED
  17 TE2/0/1/17 Clear (In use) RED(SWEC: MG2)
  18 TE2/0/1/18 Clear (In use) RED(SWEC: MG2)
  19 TE2/0/1/19 Clear (In use) RED(SWEC: MG2)
  20 TE2/0/1/20 Clear (In use) RED(SWEC: MG2)
  21 TE2/0/1/21 Clear (In use) RED(SWEC: MG2)
  22 TE2/0/1/22 Clear (In use) RED(SWEC: MG2)
  23 TE2/0/1/23 Clear (In use) RED(SWEC: MG2)
  24 TE2/0/1/24 Clear (In use) RED(SWEC: MG2)
  25 TE2/0/1/25 Clear (In use) RED(SWEC: MG2)
  26 TE2/0/1/26 Clear (In use) RED(SWEC: MG2)
  27 TE2/0/1/27 Clear (In use) RED(SWEC: MG2)
  28 TE2/0/1/28 Clear (In use) RED(SWEC: MG2)
  29 TE2/0/1/29 Clear (In use) RED(SWEC: MG2)
  30 TE2/0/1/30 Clear (In use) RED(SWEC: MG2)
  31 TE2/0/1/31 Clear (In use) RED(SWEC: MG2)

Regards
Bilal

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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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-
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Newline  Fax: +1-760-731-3000

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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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> >     http://www.api-digital.com --
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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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 >

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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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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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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
> >>
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> >
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> > 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
>>
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>>   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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Play: t...@mountifield.org - http://tony.mountifield.org

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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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--
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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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[asterisk-users] ISDN PRI configuration

[Vieri]

Hi,

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.

The only information I got from the telco is:
"
Line Coding [HDB3] 
Framing [CRC4]
Encapsultation [hdlc 
Isdn switch-type primary-[net5]
"

Is "crc4" actually a "framing" parameter as stated by the telco, or is it just 
an "optional line coding parameter"?

I searched the web and not knowing exactly which parameters to use, I tried the 
following zaptel/dahdi config:

# TE120P (PRI):
span=1,1,0,ccs,hdb3,crc4
# as E1
bchan=1-15
dchan=16
bchan=17-31

switchtype = euroisdn
signalling = pri_cpe

However, the link doesn't work and I get this:

*CLI show status:
Description  Alarms IRQbpviol CRC4
Wildcard TE120P Card 0   RED1  0  0

# cat /proc/zaptel/1
Span 1: WCT1/0 "Wildcard TE120P Card 0" (MASTER) HDB3/CCS/CRC4 RED
IRQ misses: 1

   1 WCT1/0/1 Clear (In use) RED
   2 WCT1/0/2 Clear (In use) RED
   3 WCT1/0/3 Clear (In use) RED
   4 WCT1/0/4 Clear (In use) RED
   5 WCT1/0/5 Clear (In use) RED
   6 WCT1/0/6 Clear (In use) RED
   7 WCT1/0/7 Clear (In use) RED
   8 WCT1/0/8 Clear (In use) RED
   9 WCT1/0/9 Clear (In use) RED
  10 WCT1/0/10 Clear (In use) RED
  11 WCT1/0/11 Clear (In use) RED
  12 WCT1/0/12 Clear (In use) RED
  13 WCT1/0/13 Clear (In use) RED
  14 WCT1/0/14 Clear (In use) RED
  15 WCT1/0/15 Clear (In use) RED
  16 WCT1/0/16 HDLCFCS (In use) RED
  17 WCT1/0/17 Clear (In use) RED
  18 WCT1/0/18 Clear (In use) RED
  19 WCT1/0/19 Clear (In use) RED
  20 WCT1/0/20 Clear (In use) RED
  21 WCT1/0/21 Clear (In use) RED
  22 WCT1/0/22 Clear (In use) RED
  23 WCT1/0/23 Clear (In use) RED
  24 WCT1/0/24 Clear (In use) RED
  25 WCT1/0/25 Clear (In use) RED
  26 WCT1/0/26 Clear (In use) RED
  27 WCT1/0/27 Clear (In use) RED
  28 WCT1/0/28 Clear (In use) RED
  29 WCT1/0/29 Clear (In use) RED
  30 WCT1/0/30 Clear (In use) RED
  31 WCT1/0/31 Clear (In use) RED

Placing a call through the Zap/Dahdi trunk in Asterisk doesn't work and I get 
the following message in the log:

chan_dahdi.c: No D-channels available!  Using Primary channel 16 as D-channel 
anyway!
logger.c: -- Attempting call on Zap/g1/999xx for 
999xx@custom-TESTCALL:1 (Retry 1)
channel.c: Unable to request channel Zap/g1/999xx
pbx_spool.c: Call failed to go through, reason (8) Congestion (circuits busy)
chan_dahdi.c: No D-channels available!  Using Primary channel 16 as D-channel 
anyway!

Am I missing some information here?
I'm *supposing* it should be E1 (and that I can use 16 as dchan), euroisdn (not 
"national"), but my telco states "hdlc Isdn switch-type primary-[net5]" and I 
don't know how to translate it to zaptel/dahdi...

Also, my telco hasn't mentioned anything about ccs but I tried it anyway 
because I wouldn't know what else to use.

I also tried 
signalling = pri_net
but still got the same RED alerts.

Any suggestions?

Thanks

Vieri



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[asterisk-users] ISDN TE/NT Mode

[james kielty]

Hi,am using an atcom isdn pbx its using a colegne HFC chip..It comes in TE 
mode..I want to switch it to NT mode.Its not possible to select NT mode in the 
GUI so I went to config filesmisdn-init.config and edited it there.But 
everytime I do misdn show port 1 its alway in TE mode.Can anyone help me with 
this ISDN stuff?CheersJames-- 
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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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[asterisk-users] ISDN-FAX with Asterisk

[Thorolf Godawa]

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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[asterisk-users] ISDN & SS7

[huu giang]

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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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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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
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



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

[Paulo Santos]

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

[Paulo Santos]

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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[asterisk-users] ISDN BRI call disconnection issue

[Gopalakrishnan A.N]

Hi,

 I have a Netmod ISDN BRI router and from the router I have connected the
analog port in Asterisk via FXO card. Two analog lines I have connected to
asterisk machine. When both the lines are established, after 31 minutes the
call is automatically disconnected.

While checking the log it shows as busy tone is detected because of this
existing call is disconnected.

Did anybody faced this kind of issue. Also some assistance would be much
appreciated.

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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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 Gergomailto:csi...@gmail.com


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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?

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


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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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http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] ISDN -> SIP

[Stefan Dreyer]

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.

My extension conf is:

general]
static=yes
writeprotect=no

[globals]
OUT_PORT=1

[ISDN]
exten => 12345,1,Dial(SIP/012346737...@sipprovider.local)


If i call to the msn 12345, the SIP-call is going out, but after a
second the call is stopped.
What is wrong, with my configuration?

Kernel show
Jun 10 20:48:58 wolf kernel: hdlc_down unknown prim(280)
Jun 10 20:49:04 wolf kernel: MDL_ERROR|REQ (tei_l2)

Asterisk shows:


P[ 1] MGMT: SSTATUS: L1_ACTIVATED

P[ 1] handle_frm: frm->addr:42000103 frm->prim:3f082

P[ 1] channel with stid:0 not in use!

P[ 1] handle_frm: frm->addr:42000103 frm->prim:30582

P[ 1] set_channel: bc->channel:0 channel:1

P[ 1] I IND :NEW_CHANNEL oad:xxx dad:12345 pid:2 state:none

P[ 1]  --> channel:1 mode:TE cause:16 ocause:16 rad: cad:

P[ 1]  --> info_dad: onumplan:2 dnumplan:4 rnumplan:  cpnnumplan:0

P[ 1]  --> caps:Speech pi:0 keypad: sending_complete:1

P[ 1]  --> screen:0 --> pres:0

P[ 1]  --> addr:0 l3id:20007 b_stid:0 layer_id:0

P[ 1]  --> facility:Fac_None out_facility:Fac_None

P[ 1]  --> bc_state:BCHAN_CLEANED

P[ 1] Chan not existing at the moment bc->l3id:20007 bc:0x8721e9c
event:NEW_CHANNEL port:1 channel:1
P[ 1] NO USERUESRINFO

P[ 1]  --> found chan (preselected): 1

P[ 1] set_chan_in_stack: 1

P[ 1] setup_bc: with dsp

P[ 1]  --> Channel is 1

P[ 1]  --> TRANSPARENT Mode

P[ 1] I IND :SETUP oad:xxx dad:12345 pid:2 state:none

P[ 1]  --> channel:1 mode:TE cause:16 ocause:16 rad: cad:

P[ 1]  --> info_dad: onumplan:2 dnumplan:4 rnumplan:  cpnnumplan:0

P[ 1]  --> caps:Speech pi:0 keypad: sending_complete:1

P[ 1]  --> screen:0 --> pres:0

P[ 1]  --> addr:50010102 l3id:20007 b_stid:10010100 layer_id:50010180

P[ 1]  --> facility:Fac_None out_facility:Fac_None

P[ 1]  --> bc_state:BCHAN_ACTIVATED

P[ 1]  --> Bearer: Speech

P[ 1]  --> Codec: Alaw

P[ 0]  --> * NEW CHANNEL dad:12345 oad:xxx

P[ 1] read_config: Getting Config

P[ 1]  --> CTON: Unknown

P[ 1]  --> EXPORT_PID: pid:2

P[ 1]  --> PRES: Allowed (0)

P[ 1]  --> SCREEN: Unscreened (0)

P[ 1] * Queuing chan 0x89e5410

P[ 1] I SEND:RELEASE oad:xxx dad:12345 pid:2

P[ 1]  --> bc_state:BCHAN_ACTIVATED

P[ 1]  --> channel:1 mode:TE cause:16 ocause:1 rad: cad:

P[ 1]  --> info_dad: onumplan:2 dnumplan:4 rnumplan:  cpnnumplan:0

P[ 1]  --> caps:Speech pi:0 keypad: sending_complete:1

P[ 1]  --> screen:0 --> pres:0

P[ 1]  --> addr:50010102 l3id:20007 b_stid:10010100 layer_id:50010180

P[ 1]  --> facility:Fac_None out_facility:Fac_None

P[ 1] GOT SETUP OK

P[ 1] Sending msg, prim:34d80 addr:41000104 dinfo:20007

P[ 1] BCHAN: bchan ACT Confirm pid:2

P[ 1] handle_frm: frm->addr:42000103 frm->prim:3f182

P[ 1]  --> lib: RELEASE_CR Ind with l3id:20007
P[ 1]  --> lib: CLEANING UP l3id: 20007
P[ 1]  --> hangup
P[ 1] * IND : HANGUPpid:2 ctx:ISDN dad:12345 oad: State:EXTCANTMATCH
P[ 1]  --> l3id:20007
P[ 1]  --> cause:16
P[ 1]  --> out_cause:16
P[ 1]  --> Channel: mISDN/1-u0 hungup new state:CLEANING
P[ 1] $$$ CLEANUP CALLED pid:2
P[ 1] $$$ Cleaning up bc with stid :10010100 pid:2
P[ 1]  --> ec_disable
P[ 1] Sending Control ECHOCAN_OFF
P[ 1] ph_control: c1:2319 c2:0
P[ 1] empty_chan_in_stack: 1
P[ 0] handle_bchan: BC not found for prim:f2481 with addr:55010180 dinfo:0
P[ 0] received 1k Unhandled Bchannel Messages: prim f2481 len 0 from
addr 55010180, dinfo 0 on this port.
P[ 1] MGMT: SSTATUS: L1_DEACTIVATED



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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.

-- 
Tilghman Lesher
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.

-- 
Tilghman Lesher
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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[asterisk-users] ISDN config: LBO values

[Jaap Winius]

Hi all,

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?

Thanks,

Jaap

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[asterisk-users] ISDN Options

[Razza]

All,
I have not found/seen a resolution to the issue where my TDM400P seems
to cause problems, as outlined in the "mISDN (HFC-S) and TDM400P -
mISDN: ISAC XDU no TX_BUSY" thread.
I have also not found/seen a simple 'how to' on patching DAHDi with
ZAPHFC as outlined in the "HFC-S card" thread.

Do I have anyother options with ISDN?

Thanks in advance!

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[asterisk-users] ISDN phone not ringing. ISDN PBX not answering?!

[René Rössler]

Hi,

I've set up an Asterisk as voip gatway:

VOIP <-> Asterisk <-> hfc-s card <-> NTBA <-> Siemens Gigaset Dect ISDN pbx.

Outgoing calls from dect handset to the world are working. Incoming calls don't 
even ring the handset.

I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is in 
nt mode.
The msn is set at the dect phone/base station for outgoing and incoming calls.

Asterisk version: 1.6.2.0~dfsg~beta4-0.7501

/etc/dahdi/system.conf:

# Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [NT] layer 1 ACTIVATED (G3)" (MASTER) 
AMI/CCS 
span=2,0,0,ccs,ami
# termtype: nt
bchan=1-2
dchan=3
echocanceller=oslec,1-2
alaw=1-3

# Global data

loadzone= de
defaultzone = de

EOF

/etc/asterisk/chan_dahdi.conf:

[channels]
language=de
switchtype=euroisdn
pridialplan=local
prilocaldialplan=dynamic
internationalprefix = 00
nationalprefix = 0
localprefix = 0711
privateprefix = 0711
unknownprefix =
signalling=bri_net_ptmp
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=100
mohinterpret=default
mohsuggest=default
callerid = asreceived
immediate=no
overlapdial=yes
facilityenable=yes
callprogress=yes

group=1
context=isdn1
channel => 1-2

EOF

/etc/asterisk/extensions.conf:

[default]
exten => _X.,1,NoOp(${EXTEN})

[isdn1]
exten => _X.,1,Dial(SIP/${ext...@sipgate,30,trg)
exten => _X.,n,Hangup 

[from-sipgate]

;Skype issues
exten => _X.,1,GotoIf($["${CALLERID(num)}" != "anonymous"]?notanonymous)
exten => _X.,n,NoOp(Changing Caller ID number from ${CALLERID(num)} to 
99})
exten => _X.,n,Set(CALLERID(num)=99)
exten => _X.,n(nowanonymous),NoOp(The number shown in the CALLERID NUMBER field 
is ${CALLERID(num)})

;Call Handset
exten => _X.,n,Dial(DAHDI/g1/${EXTEN})
exten => _X.,n,Congestion
exten => _X.,n,Busy
exten => _X.,n,Hangup

EOF

Output with verbose 3 and debug 3, call from skype out:

  == Using SIP RTP CoS mark 5
-- Executing [...@from-sipgate:1] GotoIf("SIP/sipgate-XXX", 
"0?notanonymous") in new stack
-- Executing [...@from-sipgate:2] NoOp("SIP/sipgate-XXX", "Changing 
Caller ID number from anonymous to 99}") in new stack
-- Executing [...@from-sipgate:3] Set("SIP/sipgate-XXX", 
"CALLERID(num)=99") in new stack
-- Executing [...@from-sipgate:4] NoOp("SIP/sipgate-XXX", "The 
number shown in the CALLERID NUMBER field is 99") in new stack
-- Executing [...@from-sipgate:5] Dial("SIP/sipgate-XXX", 
"DAHDI/g1/XXX") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/XXX
-- Hungup 'DAHDI/1-1'
  == Spawn extension (from-sipgate, XXX, 5) exited non-zero on 
'SIP/sipgate-XXX'

EOF

Same with pri intense:

  == Using SIP RTP CoS mark 5
-- Executing [...@from-sipgate:1] GotoIf("SIP/sipgate-XXX", 
"0?notanonymous") in new stack
-- Executing [...@from-sipgate:2] NoOp("SIP/sipgate-XXX", "Changing 
Caller ID number from anonymous to 99}") in new stack
-- Executing [...@from-sipgate:3] Set("SIP/sipgate-XXX", 
"CALLERID(num)=99") in new stack
-- Executing [...@from-sipgate:4] NoOp("SIP/sipgate-XXX", "The 
number shown in the CALLERID NUMBER field is 99") in new stack
-- Executing [...@from-sipgate:5] Dial("SIP/sipgate-XXX", 
"DAHDI/g1/XXX") in new stack
2 -- Making new call for cr 32773
-- Requested transfer capability: 0x00 - SPEECH
2 > Protocol Discriminator: Q.931 (8)  len=47
2 > Call Ref: len= 1 (reference 5/0x5) (Originator)
2 > Message type: SETUP (5)
2 > [04 03 80 90 a3]
2 > Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
2 >  Ext: 1  Trans mode/rate: 64kbps, circuit-mode 
(16)
2 >User information layer 1: A-Law (35)
2 > [18 01 89]
2 > Channel ID (len= 3) [ Ext: 1  IntID: Implicit  Other  Spare: 0  Exclusive  
Dchan: 0
2 >ChanSel: B1 channel
2  ]
2 > [28 09 61 6e 6f 6e 79 6d 6f 75 73]
2 > Display (len= 9) [ anonymous ]
2 > [6c 0c 41 80 39 39 39 39 39 39 39 39 39 39]
2 > Calling Number (len=14) [ Ext: 0  TON: Subscriber Number (4)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
2 >   Presentation: Presentation permitted, user number 
not screened (0)  '99' ]
2 > [70 08 c1 35 38 34 38 34 30 36]
2 > Called Number (len=10) [ Ext: 1  TON: Subscriber Number (4)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1)  'XXX' ]
2 q931.c:3134 q931_setup: call 32773 on channel 1 enters state 1 (Call 
Initiated)
-- Called g1/XXX
T203 counter expired, sending RR and scheduling T203 again
Sending Receiver Ready (0)
2 
> [ 02 81 01 01 ]
2 
> Supervisory frame:
2 > SAPI: 00  C/R: 1 EA: 0
>  TEI: 064EA:

[asterisk-users] ISDN users: 1.6.x users, I need some testing done please, regarding Overlap Receiving

[Alec Davis]

Overlap receiving timeout, plus dialplan latency, causes network to retry
SETUP 
https://issues.asterisk.org/view.php?id=16789
 
This patch removes the requirement that some may have found that you need to
insert a Proceeding() statement very early in your dialplan, otherwise an
inbound overlap call may retry and fail.
 
Our experience was from a PRI connected PABX, if we took too long doing
database lookups, the PABX would resend the SETUP, asterisk doesn't handle
these yet, treats as not newcall and rejects it.
 
Please feedback to the bug.
 
Currently the patch is for Asterisk 1.6.1, but the issue applies to all.
 
Alec Davis  
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[asterisk-users] ISDN Cause codes for unanswered calls

[Steve Moran]

I am wanting to use the ISDN cause code on an Asterisk 1.6 server to
determine the status of a call attempt, where the call might not actually
connect. Reason is I am checking for valid telephone numbers from a list of
numbers, and I would like to know if the call has answered and cleared which
I can by writing the hangupcause variable, but where I get an out of order
network message, or number doesn't exist, I want to capture these ISDN cause
codes where the call might not have connected and started the dialplan
extension.

Is there any way to capture cause codes from calls that didn't connect?

Thanks

Steve Moran
Sydney, Australia
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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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http://gocept.com · tel +49 345 1229889 0 · fax +49 345 1229889 1
Zope and Plone consulting and development


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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http://gocept.com · tel +49 345 1229889 0 · fax +49 345 1229889 1
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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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[asterisk-users] ISDN: Inband DTMF doesn't trigger transfer feature

[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) 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)

-- 
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http://gocept.com · tel +49 345 1229889 0 · fax +49 345 1229889 1
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[asterisk-users] ISDN Remote HOLD

[Andrea Cristofanini]

Hi there,
i'm using dahdi to manage a B400P openvox BRI card.
All works as expected, i would like to know if there ia a way to put the
call in REMOTE HOLD, like pressing R  button on ISDN phone.

This can be done by CAPI  using the proper application ,
It is implemented on DAHDI ?
Regards Andrea

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[asterisk-users] ISDN Calling Sub Address and Called Sub Address for the branches

[Alec Davis]

Since 2004 asterisk/libpri have been able to receive the Calling Sub Address
in the ISDN setup message, and the dialplan was able to use it if required.
It's support is limited to only NSAP, not BCD or user formatted.
 
At the time 25/06/04 the questioned was asked, wouldn't it be a good idea to
be able to transmit it as well, but that never got implemented, as it wasn't
required at the time.
 
Further to this, there is also Called Sub Address that allows you to dial a
particular terminal device at an ISDN number, these days isn't a terminal
device a users extension.
 
Finishing off the limited support for SUBADDR, if your keen is at

https://issues.asterisk.org/view.php?id=15604, this adds CallingSubAaddress
(Transmit) and CalledSubAddress (Transmit and Receive), still only in NSAP,
not User formatted.
 
This code may not ever make it into trunk, but if you find this code useful
please leave a comment on the mantis bug. This has been tested with the
exisiting 1.4 -1.6.2 branches, and is in use at 3 PRI/BRI sites with
asterisk 1.6.1.
 
The Digium team have other good ideas for 1.6.3 which will as I understand
it support SUBADDR over any transport, but it will be a while before most of
us are happy using the latest offering in production.
 
Alec Davis
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[asterisk-users] ISDN Error Code 42

[Nitesh Divecha]

Hello All,

Just got one general question on ISDN error code 42. As per Cisco docs 
and Wiki 
(http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause) it 
says "Switch equipment Congestion" with explanation "The destination 
cannot be reached because the network switching equipment is temporarily 
overloaded"...

My question is are the calls treated as "Failed" and I will be pulled 
away from the routing? Is this similar to ISDN error code 34's?

Please help...

Cheers,
Nitesh


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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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[asterisk-users] ISDN from Macau CTM

[Si Tai Fan]

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

[asterisk-users] ISDN Timer T309

[Afonso Zimmermann]

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
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 18: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 18: Invalid argument
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 19: Red Alarm
[Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disab

[asterisk-users] ISDN and routers...

[Julien Claassen]

Hello!
   Sorry for not being able to phrase the problem in one line. My phone 
situation is this:
The calls go over analog line (or NGN/vip) I don't really get to see it. I 
have got a router with a lot of jacks. One or two of them are for ISDN phones 
or other ISDN capable devices. Can I use chan_misdn and my good old ISDN card 
with this setup. Or do I have to get a card, that can handle analog lines.
   The telephone now connected to the router is analog and quite old. the 
router is by Samsung, but I couldn't find out, which device it is exactly. 
will have to wait till I get braille-support for the stupid win-notebook. Too 
much javascript in the webinterface... :-(
   Thanks for any good hints on this!
   Kindest regards
 Julien


Music was my first love and it will be my last (John Miles)

 FIND MY WEB-PROJECT AT: 
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL PAGES AT: ===
http://www.juliencoder.de

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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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