asterisk Gigaset T500 doesn't receive incoming calls

[Graham J]

New ISDN2e line installed 11 April 2013.

Connected to Gigaset T500 PBX with Sirrix IDSN card.

We can make internal calls from any extension to any other.

Internal extensions can make outgoing calls to the PSTN.

External callers dialling in hear silence followed by the "number
unobtainable" tone. Log entries suggest that the PBX sees some activity
when there is an incoming call, but doesn't know what to do with it.

Cannot determine whether fault is in misconfiguration of the PBX or
inappropriate provisioning of the ISDN2e service.

Any ideas on how to proceed?

I can show extracts from the logs ...

TIA

--
Graham J

[R. Mark Clayton]

"Graham J" <graham@invalid> wrote in message
news:516d24fb$0$1104$5b6a...@news.zen.co.uk...

You have asked this elsewhere.

point to point or point to multi-point that is the question.

[Graham J]

R. Mark Clayton wrote:
[snip]

>
> You have asked this elsewhere.
>
> point to point or point to multi-point that is the question.
>
>

And I have explained elsewhere that we have tried configuring the
Gigaset with both options in turn, rebooting each time. They are called
"Device Connection" or "Multi-Device Connection". Neither setting works!

--
Graham J

[Jono]

R. Mark Clayton explained on 16/04/2013 :

I directed him here...hoping someone like Gordon Henderson might drop
in...as, although I can tinker with Asterisk, I struggle interpretting
logs, verbose or otherwise.

I /think/ his line is point-to-multipoint, however, try as I might, I
can't tell if the Gigaset can handle P-MP...at least there's no direct
mention of it in the manual.

Also, I'm fairly sure that BT haven't closed all the orders off
properly after the install. I have seen this cause issues with routing
in the past (it could be a task that involves creating/updating the
exchange data that's outstanding, for all we know)

[Graham J]

The order form asked "Digital System / Digital Standard"; we ordered
"Standard". Jono says that means P-MP.

The Gigaset says "Device Connection" or "Multi-Device Connection" in the
context of P-P or P-MP. Neither setting works!

--
Graham J

[alexd]

Graham J (for it is he) wrote:

> Any ideas on how to proceed?

Other than 'core set verbose', debugging in Asterisk is specific to the
thing you're trying to debug, eg for a Sirrix ISDN card, somewhere in this
jumble you will find what you need to either interrogate the card directly
or turn up the logging:

http://www.voip-info.org/wiki/view/Asterisk+Sirrix+ISDN+channels

--
<http://ale.cx/> (AIM:troffasky) (UnSoEs...@ale.cx)
21:31:28 up 32 days, 12:27, 6 users, load average: 0.19, 0.20, 0.16
Qua illic est reprehendit, illic est a vindicatum

[Jono]

Graham J submitted this idea :

> Jono wrote:

>>
>> I /think/ his line is point-to-multipoint, however, try as I might, I
>> can't tell if the Gigaset can handle P-MP...at least there's no direct
>> mention of it in the manual.

>>
>>

> The order form asked "Digital System / Digital Standard"; we ordered
> "Standard". Jono says that means P-MP.
>

From the ISDN2 Product Description:

• T reference (aka point-to-point working) is the default System Access
configuration for customers wishing to take advantage of DDI. It’s used
when the customer has an Integrated Services Private Branch Exchange
(ISPBX) connected to their service. Instead of the equipment deciding
whether the call is meant for it or not, the ISPBX decides and routes
the call accordingly

• S/T reference (aka point-to-multipoint working) is the standard
access setting for ETSI single lines with and without Multi Subscriber
Numbers (MSNs). MSNs are assigned to allow each piece of equipment to
have its own unique dial in number. In this configuration, the network
has a direct signalling relationship with each piece of equipment.

Standard Access permits a maximum of two channels, one main and one
auxiliary. A maximum of eight devices can be connected directly to the
line, with a maximum of 10 directory numbers.

Standard Access supports MSNs on point-to-multipoint configuration.
However, Point-to-point configuration is not supported on standard
access.

Line characteristics:

• Point to multipoint
• Only supports 2B and 1D Channel
• 1 main number plus a maximum of 9 additional MSNs
• Cannot have DDIs.

[Graham J]

alexd wrote:
> Graham J (for it is he) wrote:
>
>> Any ideas on how to proceed?
>
> Other than 'core set verbose', debugging in Asterisk is specific to the
> thing you're trying to debug, eg for a Sirrix ISDN card, somewhere in this
> jumble you will find what you need to either interrogate the card directly
> or turn up the logging:
>
> http://www.voip-info.org/wiki/view/Asterisk+Sirrix+ISDN+channels

OK we need use SSH and log in as root. Any idea as to the default
password, please? No evidence of it that we can find on the device or
packaging.

Tech support from the supplier has not yet been able to answer this.

TIA

--
Graham

[Jono]

Graham J pretended :

It beggars belief that they would suggest using a root login via SSH in
their Wiki...and not say how..

<https://gigaset.atlassian.net/wiki/dosearchsite.action?queryString=ssh>

[Graham J]

For future reference: on the base of the unit there is a label with
numbers and letters, like this:

1234 1234 1234 abcdef

It is the password, but it needs the white space replaced with hyphens,
thus:

1234-1234-1234-abcdef

So we are now getting debug output. It clearls shows the incoming call
being received then immediately disconnected.

This of course is debug from the ISDN card, so presumably the rest of
the PBX does not know what to do with what the ISDN card tells it.

I will post the debug to follow on from the start of the thread.

--
Graham J

[Graham J]

Sample log from the ISDN card follows (with acknowledgements for help
from Jono):

2013/04/17 10:35:37.398 : NT -> TE : '(0x2003) HEX: 02 FF 03 08 01 51 05
A1 04 03 90 90 A3 18 01 89 1E 02 84 83 '
tei = 127
Layer 3 Data:
PD = 0x08 = DSS1
CR = 0x51 = 81
MT = 0x05 = SETUP
IE_IDENTIFIER = 0x21 = SENDING COMPLETE
IE_IDENTIFIER = 0x04 = BEARER CAPABILITY
CODING STANDARD = 0x00 = CCITT
INF. TRANSF. CAP. = 0x10 = 3.1 kHz Audio
TRANSFER MODE = 0x00 = circuit mode
INF. TRANSF. RATE = 0x10 = 64 kbit/s
USER INF L1 PROT = 0x03 = G.711 A-law
IE_IDENTIFIER = 0x18 = CHANNEL ID
IF IDENT PRESENT = 0
INTERFACE TYPE = 0 = BRI
PREFERRED / EXCL. = 1 = exclusive
D CHANNEL INDICAT = 0 = is not D channel
INF. CHAN. SELECT = 1 = B1
IE_IDENTIFIER = 0x1E = PROGRESS INDICATOR
CODING = 0 = CCITT (ETS)
LOCATION = 4 = remote public net
PROGRESS DESCR = 3 = calling number is not ISDN

2013/04/17 10:35:37.412 : TE -> NT : '(0x2003) ECHO: 00 97 00 00 08 01
D1 02 18 01 89 '
tei = 75
Layer 3 Data:
PD = 0x08 = DSS1
CR = 0x51 = 81
MT = 0x02 = CALL_PROCEEDING
IE_IDENTIFIER = 0x18 = CHANNEL ID
IF IDENT PRESENT = 0
INTERFACE TYPE = 0 = BRI
PREFERRED / EXCL. = 1 = exclusive
D CHANNEL INDICAT = 0 = is not D channel
INF. CHAN. SELECT = 1 = B1

2013/04/17 10:35:37.430 : TE -> NT : '(0x2003) ECHO: 00 97 02 00 08 01
D1 45 08 02 80 9F '
tei = 75
Layer 3 Data:
PD = 0x08 = DSS1
CR = 0x51 = 81
MT = 0x45 = DISCONNECT
IE_IDENTIFIER = 0x08 = CAUSE
CODING = 0 = CCITT (ETS)
LOCATION = 0 = user
CAUSE = 0x1F = Normal, unspecified.

2013/04/17 10:35:37.491 : NT -> TE : '(0x2003) HEX: 02 97 00 04 08 01 51
4D '
tei = 75
Layer 3 Data:
PD = 0x08 = DSS1
CR = 0x51 = 81
MT = 0x4D = RELEASE

2013/04/17 10:35:37.492 : TE -> NT : '(0x2003) ECHO: 00 97 04 02 08 01
D1 5A '
tei = 75
Layer 3 Data:
PD = 0x08 = DSS1
CR = 0x51 = 81
MT = 0x5A = RELEASE_COMPLETE

Any ideas, please?

--
Graham J

[Jono]

Graham J has brought this to us :

>
> Any ideas, please?

Are you using something like putty to SSH in?

I'd be interested to see the output from the Asterisk CLI itself.

Once logged in, try issuing "asterisk -vvvvr" without the quotes...then
call the ISDN number from your mobile.

Alternatively/also, let's have a butchers at the Asterisk log:

<https://gigaset.atlassian.net/wiki/pages/viewpage.action?pageId=17596951>

[Dave Higton]

In message <516e792a$0$1137$5b6a...@news.zen.co.uk>

It's some years since I was doing this, but... the Information
Transfer Capability says 3.1 kHz audio, not speech. That suggests
the call should be directed to a modem or fax machine and not a
phone. If that's the case, your Asterisk switch would be correct
to refuse the call.

I reserve the right to be wrong!

Dave

[alexd]

Graham J (for it is he) wrote:

> Jono wrote:

>> It beggars belief that they would suggest using a root login via SSH in
>> their Wiki...and not say how..
>>
>> <https://gigaset.atlassian.net/wiki/dosearchsite.action?queryString=ssh>

There's a quite a bit that beggars belief on that wiki - I get the feeling
here that Graham J is a paying beta tester...

> This of course is debug from the ISDN card, so presumably the rest of
> the PBX does not know what to do with what the ISDN card tells it.

OK so you need to look at the Asterisk console while trying an inbound call.

--
<http://ale.cx/> (AIM:troffasky) (UnSoEs...@ale.cx)

21:33:44 up 33 days, 12:29, 6 users, load average: 0.19, 0.15, 0.20