Re: [asterisk-users] Need ISDN call generator

[Storer, Darren]

Keep an eye out for older model INET Spectra call generators, with ISDN /
SS7 stacks. These days the old boxes are being sold off very cheaply on
popular auction sites.

Hammer was the other popular call generator hardware that you might find
being sold at a fraction of the original cost.

HTH

Darren

On 28 August 2016 at 10:20, Hooman Fazaeli  wrote:

>
> Hi
>
> To troubleshoot FreeBSD panics triggered by ISDN load on an asterisk
> system,
> we are looking to buy an ISDN call generator/simulator device.
>
> The minimum requirements include:
>
> - Not too expensive
> - PRI support (BRI support is a plus)
> - CCS+CRC4 farming + HDB3 coding
> - EuroISDN (DSS1) support.
> - A minimum of 4 ports (120 channels/concurrent calls)
> - Compatibility with Digium cards.
> - DUT in TE mode.
> - Reliable & stable operation.
>
> I would like to hear your recommendations for and experiences about
> such a device. Recommendations on hand crafted systems using
> Asterisk, DAHDI and PRI cards on any OS which has worked stably for
> someone are also welcome
>
> Thanks.
>
>
> --
> Best regards
> Hooman Fazaeli
>
>
> --
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> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
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Re: [Asterisk-Users] ISDN/SS7 book?

[Storer, Darren]

Hi Roy,

On Wed, 5 Jan 2005 15:56:39 +0100, Roy Sigurd Karlsbakk
[EMAIL PROTECTED] wrote:

 I was told the book ISDN and SS7: Architectures for
 Digital Signaling Networks by Uyless Black (ISBN 0132591936) was a
 good choice, but this seems sold out. Does anyone know about another
 book about the subject?

For a book on SS7/VoIP only, Signaling System #7 by Travis Russell
is good (ISBN 0071361197) but for a good overview of the whole TDM
telephone network including SS7, R2 and ISDN, I recommend Signaling
in Telecommunication Networks by John G. van Bosse (ISBN
0-471-57377-9).

The Travis Russell book can be found very cheaply on eBay:

http://tinyurl.com/3zd87

HTH

Darren
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Re: [Asterisk-Users] TE405P E1 coax cables with balun

[Storer, Darren]

Hi Ciro,

On Tue, 21 Dec 2004 16:54:47 +0100, Ciro La Ferrara
[EMAIL PROTECTED] wrote:
 Hi,
 I am new with asterisk. I am setting a Wildcard TE405P. E1s in Italy come in
 on a pair of RG-59 coax cables with BNC connectors. So I need an
 adapter/balun http://www.allcomtlc.com/al_g703n3.htm . I have It but I am
 not sure that It works. I have configured my asterisk in this way:

[snip]

 
 With cable plugged in, the led are turned off.
 What's wrong?

We have many customers using Asterisk servers with E100P/TE4XXP and 75
Ohm baluns here in the UK. A couple of questions come to mind:

 - Have you tried reversing the TX and RX BNC terminated cables to the balun?
 - Are you confident that you are plugged into span 1 on the TE405P? 
(Try the connector at the opposite end of the card)
 - If you cat /proc/zaptel/1 Does the first line look similar to the following?:
 Span 1: TE4/0/1 TE405P (PCI) Card 0 Span 1 HDB3/CCS/CRC4 ClockSource

HTH

Darren
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Re: [Asterisk-Users] News about SS7?

[Storer, Darren]

On Thu, 9 Dec 2004 10:45:08 +0200, Hadi Jadallah [EMAIL PROTECTED] wrote:
 Hi list,
 
 I have been folowing the SS7 for * thread and it got me wondering about the 
 current status of SS7 for *.
 Anybody knows if ISUP going to be supported?

Hadi,

at this stage ISUP is the only User/Application Part that has been
deployed over SS7 from Asterisk platforms.

The team are more than capable of developing MAP and INAP layers too
but it could be a while before these applications surface.

HTH

Darren
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Re: [Asterisk-Users] skinny error

[Storer, Darren]

Hi Thomas,

On Sun, 14 Nov 2004 10:42:08 +0200, Thomas Andrews [EMAIL PROTECTED] wrote:
 What does this error mean:
 
 Nov 14 10:35:12 WARNING[24733]: Unable to get our IP address, Skinny disabled

I have had problems when the IP address of the Asterisk host has not
been explicitly defined on the line bind = x.x.x.x  in the file
skinny.conf.

In older versions of code bind = 0.0.0.0 was sufficient. I now find
that you must indicate the actual IP address of the LAN card on the
Asterisk server or skinny support will not startup correctly.

HTH

Darren
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Re: [Asterisk-Users] Marconi Sys X/TE410P configuration

[Storer, Darren]

On Wed, 10 Nov 2004 00:19:58 +, Steve Kennedy
[EMAIL PROTECTED] wrote:
  Hi Steve,
 
 Hi, name rings a bell for some reason ?

Hmm, yes, I have the same feeling about your name. Ever been to
borstal? (Just kidding...)

I would have answered sooner but last week turned nasty when someone
delivered 1.2M calls in 10 minutes when they had forecast 30k calls
over 3 weeks; messy, very messy - always carry a spatula!

 Couldn't find any config data on the lists, though did pick out the
 EuroISDN stuff. Hopefully it's just a switch misconfiguration.

Here are some configs from a production server using a TE4XXP card
connected to a Marconi System X switch via ETSI Q.931 PRI  (ISDN110):

http://www.comgate.tv/Marconi_Star/zaptel.conf
http://www.comgate.tv/Marconi_Star/zapata.conf

I hope that these files help. If you still have problems why don't you
let us take a look at the configs you are using?

Regards

Darren
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Re: [Asterisk-Users] Marconi Sys X/TE410P configuration

[Storer, Darren]

Hi Steve,

On Tue, 9 Nov 2004 14:05:18 +, Steve Kennedy
[EMAIL PROTECTED] wrote:
 Has anyone got a working config for a Marconi System X (Q.931) and
 Digium TE410P?
 

Check out the quote that Scott posted from an earlier listing. If you
are sure that you have EuroISDN/ETSI circuits from the System X switch
I can post sample configs for TE410P (although they should already be
somewhere in the mailing list archives...).

TE410P works well with System X; if you give us a clue as to which
carrier you will be connecting via I might be able to offer additional
advice on CLI propagation etc.

HTH

Darren
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RE: [Asterisk-Users] Asterisk testbed for teaching connecting to aPRI-ISDN

[Storer, Darren]

Hi Francesco,

you can easily run two Asterisk systems back to back in the way you
described below.

  --- ---
 |   Asterisk 1 || Asterisk 2   |
 |TE405 |==X=| TE405|
 |Provide Clock |  Cross Over Cable  | Take Clock   |
 |  NET || CPE  |
  --- ---

In the diagram above Asterisk 1 is simulating the PSTN (Telco) and Asterisk
2 is a PBX. To set the ISDN protocol correctly for Asterisk 1 your settings
should include pri_net and for Asterisk 2 you should have an entry such as
pri_cpe. To make sure that timing is generated by 1 and received by 2 you
should use the settings below:

Asterisk 1:  span=1,0,0,ccs,hdb3,crc4
Asterisk 2:  span=1,1,0,ccs,hdb3,crc4

The only hardware that you need to interconnect the two systems is a cross
over RJ45 cable, pin out is shown below:

PRI Cross Over Cable

 2--5
 1--4
 5--2
 4--1

There are plenty of notes on how to achieve your requirements, already
written up by other list contributors; simply check the archive of this
mailing list at: http://lists.digium.com/ or look at the wiki:
http://www.voip-info.org.

Good Luck!

Darren
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TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Francesco
Delfino
Sent: 10 September 2004 09:07
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk testbed for teaching connecting to
aPRI-ISDN


Hi,
I would like to realize a voip testbed that should simulate the scenario
in which two companies have an asterisk PBX connected through a PRI-ISDN
to the Telco operator.
I have no experience of T1/E1 connection but I think that the above
could be relized with 3 asterisk boxes equipped with Digium TE405P cards.
One of the box will represent the Telco, the other two, the two
companies PBX.
I would like to know if it is needed something between the point-point
connections or it is possible to just cross-connect them.
I need the testbed to be representative of the real-world difficulties
in putting on an Asterisk BOX for connecting to a PRI-ISDN: is other
hardware needed (e.g. echo cancellers or failover switches)?

Asterisk BOX (Simulate the Telco)
with Digium TE405P
   |   \
   | E1 \  T1
   | \
[What to put here?]   [What to put here?]
   |   \
   | E1 \ T1
   | \
Asterisk BOX (Company)   Asterisk BOX (Company 2)
with Digium TE405P   with Digium TE405P


Regards,
Francesco Delfino

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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Begumisa
Gerald M
Sent: 10 September 2004 12:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk testbed for teaching connecting
to aPRI-ISDN


  On Fri, 10 Sep 2004, Francesco Delfino wrote:
 [...]One of the box will represent the Telco, the other two, the two
 companies PBX. I would like to know if it is needed something
 between the point-point connections or it is possible to just
 cross-connect them.

As more experienced people prepare to reply, I'd like to give my [highly
theoretical] opinion (I'm still waiting for hardware I ordered):  I think
it is possible to just cross connect them, as long as you get the
signaling right.  In my opinion, the Box simulating the telco should
signal as the network side and the one representing the company should
signal as the customer side...

Hope that makes sense.


Cheers,
Gerald.
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RE: [Asterisk-Users] Problem with EuroISDN E1

[Storer, Darren]

Hi Claus,

CF Could someone here please explain what these codes mean..
CF
CF Aug 10 10:22:22 NOTICE[1146227632]: PRI got event: 6 on Primary
D-channel
CF of span 2
CF Aug 10 10:25:22 NOTICE[1146227632]: PRI got event: 8 on Primary
D-channel
CF of span 2

Event 6: Abort HDLC Frame
Event 8: Bad HCS

I wonder if you have the correct timing set on your E1 spans? (Sounds like
clock slippage) Please post the contents of your /etc/zaptel.conf file to
the list. It would also be useful to know if one span is to the public
network (PSTN) and the other span is to a PBX or are both spans connected to
a PBX etc.?

Regards

Darren
--
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Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Claus Futtrup
Sent: 10 August 2004 09:29
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Problem with EuroISDN E1


Hi there..

Could someone here please explain what these codes mean..

Aug 10 10:22:22 NOTICE[1146227632]: PRI got event: 6 on Primary D-channel of
span 2
Aug 10 10:25:22 NOTICE[1146227632]: PRI got event: 8 on Primary D-channel of
span 2

I have two E100P installed in the machine, but the problem only seem to
affect span 2. Users have been complaining about being unable to make calls,
but Im not sure if this has anything to do with that..
Please help.

Kind Regards

Claus Futtrup



---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.735 / Virus Database: 489 - Release Date: 06-08-2004

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RE: [Asterisk-Users] E1 config help and guidance

[Storer, Darren]

Hi Julian,

J I want to put asterisk in the middle of our current
J pbx (Meridian Option11)

Something like this?:

  -
 | |
 PSTN ---span1--| CPE  Asterisk   NET |--span2--- Nortel
 | |
 | |
  -


Assuming that you connect your incoming Telco PRI (PSTN) to span1 and the
Nortel PBX to span2 (as depicted above) the lines below should help:

Extract from zaptel.conf


span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47

Extract from zapata.conf


pridialplan=local
switchtype = euroisdn
signalling = pri_cpe

group = 1
channel = 1-10
switchtype = euroisdn
signalling = pri_cpe

group = 2
channel = 32-41
switchtype = euroisdn
signalling = pri_net

In the config lines above, span1 is set to take timing from the PSTN whilst
span2 is configured to give timing to the Nortel. Span1 will behave like a
piece of CPE (PBX) and span2 will behave like the NETwork.
NB. The channels in group 1 and 2 are depleted as you only have 10 channels
enabled on your PRI.

After you have implemented the changes above (or any subsequent changes to
the low level PRI config) you should, at the very least, remember to restart
the Asterisk system or, as Critch advises, power down and up again.

HTH

Darren
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TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of asterisk
Sent: 09 July 2004 19:00
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] E1 config help and guidance


I've googled / voip-info'd / searched until my eyes are blurry, but couldn't
see the info I was looking for. I've turned here for help!

Asterisk CVS head (9/7/04)
Fedora Core 2 (updated to 2.6.6 kernel)
DE405P (jumpers set to E1)

I want to put asterisk in the middle of our current pbx (Meridian Option11)

Currently the meridian has a 2MB pri EuroISDN card linked via a rj-45 into a
euroISDN bearer. This bearer only has 10 channels activated (out of the 30).
Obviously, this works - handsets make external calls.

What I wanted to do was to add * to the mix, in the middle so that it can
intercept inbound / outbound calls and do what it needs to do, as well as
providing all the extra functionality that this wonderful product provides.

In order to achieve this, I assumed that I needed to take rj45 from the
bearer box and plug that into span 2, and take a cable from span 1 into the
bearer box.

My problem (and blurry eyes) come from not understanding the various
protocols to assign to each span. I want the meridian to think that it's
still plugged into the EuroISDN bearer. So span 2 should be set up as a
EuroISDN link ? What should span 1 be set up as ? What channels should be
configured ?

Any guidance (I'm not looking for the solution (would be nice!) but for
pointers in the right direction).

I have previously been able to set up asterisk using the x100p and graduated
to BRI isdn. I just got the 405 today and wanted to play!

Thanks in advance.

Julian.

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RE: [Asterisk-Users] E100P

[Storer, Darren]

JM If people would read the included documentation from Digium
JM they would have known this little fact.

What documents? What do the documents say? Can we get one scanned and posted
in the wiki? (Please).

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: 08 July 2004 09:07
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] E100P


Andres wrote:
 Ing. Angel Gomez wrote:

 Hi, i just received an E100P, this is the first one I have ever seen,
 and notice that the board reads T100P. Is this right ?


 I think this was asked just a few days ago...the answer is YES.



If people would read the included documentation from Digium they would
have known this little fact.



Jeremy McNamara
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RE: [Asterisk-Users] Sending SABME continuosly. Urgent help needed!

[Storer, Darren]

Hi David,

which Telco/Carrier (type of public switch etc.) are you trying to connect
Asterisk to? Also, is this a new circuit or an existing one that was
previously working with other telephony hardware?

Regards

Darren
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TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David Morillo
Sent: 12 June 2004 16:57
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sending SABME continuosly. Urgent help needed!


Hi, I'm trying to install an E1 PRI, and I need it working by
Monday, but although everything seems ok, I get no response to calls.
When I make a pri extense debug on span 1, I repeatedly get the
following:
Sending Set Asynchronous Balanced Mode Extended
[ 00 01 7f ]
Unnumbered frame:
SAPI: 00 C/R: 0 EA: 0
 TEI: 999   EA: 1
  M3: 3  P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced
mode extended]
0 bytes of data.

And nothing else.

When making a call to that E1, I see the message D-Channel on
span 1 up 4 times, and then a Informational frame, with
TEI:000 EA:1 and anything else with zero (13 bytes of data).
Stopping T_203 timer
Starting T_200 timer
Protocol Discriminator: Q.931 (8) len=13
Call Ref: len= 2 (reference 0/0x0) (Originator)
Message type: RESTART (70)
Channel ID (len= 5) [Ext: 1  IntID: Implicit, PRI Spare: 0,
Exclusive Dchan:0
Chan Sel Reserved
Ext: 1  Coding:0  Number Specified   Channel Type: 3
Ext: 1  Spare: o  Resetting Inidicated Channel (0) ]

Then D-Channel on span 1 downn, and finally, after a while:
(...) Warning[11276]: chan_zap.c:5993 zt_pri_error: PRI: Read on
46 failed: Unknown error 500
(...) Notice[11276]: chan_zap.c:6708 pri_dchannel: PRI got
event: 8 on span 1
I think I have Asterisk stable version 1.0, CVS updated today
Can anyone help me? Please! :S

Zaptel.conf
--
span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31
loadzone = es
defaultzone=es

Zapata.conf:
--
 [channels]
language=es
context=default
switchtype=euroisdn
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
jitterbuffers=4
group = 1
channel = 1-15,17-31

I have also tried with
span=1,0,0,ccs,hdb3
immediate = yes

The line has not CRC activated (I have asked)
Thanks!


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RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID

[Storer, Darren]

Hi 
Steve,

SH Is anybody 
in the UK using Telewest as a PRI Telco 
provider?
SH Are you sending them 
caller ID?

Just a quick point of clarification 
before commenting further, do you wish to make calls via Telewest's network and 
send the CLI of your own DDI number range or do you wish to send "other numbers" 
as your CLI? If you are seeking toachieve the latter, what sort of numbers 
do you wish to propagate asthe CLI for your 
calls?

Regards
Darren
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  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Steve 
  HanselmanSent: 30 June 2004 18:57To: 
  '[EMAIL PROTECTED]'Subject: [Asterisk-Users] 
  Providing Telewest in the UK with per extension outbound 
  callerID
  
  Hi,
  
  Is anybody in the 
  UK using Telewest as a 
  PRI Telco provider?
  
  Are you sending them caller 
  ID?
  
  I've been told by Telewest 
  that:-
  
  
Oftel doesn't allow them to 
accept caller ID (this is rubbish, and I replied pointing out where in the 
link to Oftel that they sent me it was stated. We need Type 2 caller 
ID) 
Telewest can't do this. (this is 
rubbish, I'm certain that some of our customers use Telewest and they 
provide them with caller ID) 
  
  So, does anybody do this, and if 
  so, what did you have to request from them in order to enable it, and what do 
  you provide to them (how many digits and in what format).
  
  Regards
  
  Steve

RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID

[Storer, Darren]

Hi 
Steve,

Telewest should already allow the CLI transmission of your DDI range, 
without further datafill changes. If it doesn't work you should check that you 
are sending the appropriate number of digits.

Try 
sending:

-3 digit CLI
-the whole number (minus the leading 
zero)

If the 
comments above don't help please post a trace of an outgoing call and detail the 
number, if any, that is presented to theCalled Party.

HTH

Darren
-- 

Comgate
TelcoInternetBroadcast


  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Steve 
  HanselmanSent: 01 July 2004 09:57To: 
  '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] 
  Providing Telewest in the UK with per extension outbound 
  callerID
  
  Would be nice to do 
  both (type 2 and 3 I believe in Oftel terms), but I'd accept just our DDI if 
  that was all I could get.
  
  Steve
  
  
  -Original 
  Message-From: Storer, 
  Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 09:35To: 
  [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Providing 
  Telewest in the UK with per extension outbound callerID
  
  
  Hi 
  Steve,
  
  
  
  SH 
  Is anybody in the UK using Telewest as a PRI Telco 
  provider?
  SH 
  Are you sending them caller ID?
  
  Just a 
  quick point of clarification before commenting further, do you wish to make 
  calls via Telewest's network and send the CLI of your own DDI number range or 
  do you wish to send "other numbers" as your CLI? If you are seeking 
  toachieve the latter, what sort of numbers do you wish to propagate 
  asthe CLI for your calls?
  
  Regards
  Darren
  -- 
  
  Comgate
  TelcoInternetBroadcast
  
-Original 
Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]On Behalf Of Steve 
HanselmanSent: 30 June 
2004 18:57To: 
'[EMAIL PROTECTED]'Subject: [Asterisk-Users] Providing 
Telewest in the UK with per extension outbound callerID
Hi,

Is anybody in the UK using 
Telewest as a PRI Telco provider?

Are you sending them caller 
ID?

I've been told by Telewest 
that:-

1. 
Oftel doesn't allow them to 
accept caller ID (this is rubbish, and I replied pointing out where in the 
link to Oftel that they sent me it was stated. We need Type 2 caller 
ID) 
2. 
Telewest can't do this. (this is 
rubbish, I'm certain that some of our customers use Telewest and they 
provide them with caller ID) 

So, does anybody do this, and if 
so, what did you have to request from them in order to enable it, and what 
do you provide to them (how many digits and in what 
format).

Regards

Steve
  The information contained in this 
  email is intended for the personal and confidential useof the addressee 
  only. It may also be privileged information. If you are not the 
  intendedrecipient then you are hereby notified that you have received this 
  document in error andthat any review, distribution or copying of this 
  document is strictly prohibited. If you have received this communication 
  in error, please notify Brendata immediately on: +44 (0)1268 466100, 
  or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon 
  Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as 
  above. Registered in England No. 2764339See our current vacancies at 
  www.brendata.co.uk

RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID

[Storer, Darren]

SHthe contact at Telewest 
believes it's somewhere between
SHillegal and impossible to 
provide DDI numbers to the outside world.

Complete nonsense, ask to speak with someone from the 
Datafill Department.

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Steve 
  HanselmanSent: 01 July 2004 10:16To: 
  '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] 
  Providing Telewest in the UK with per extension outbound 
  callerID
  
  When the original PBX 
  was installed we asked them to override the CLI and provide a single number as 
  the PBX couldn't provide the DDI number, now the contact at Telewest believes 
  it's somewhere between illegal and impossible to provide DDI numbers to the 
  outside world.
  
  
  -Original 
  Message-From: Storer, 
  Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 10:13To: 
  [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Providing 
  Telewest in the UK with per extension outbound callerID
  
  
  Hi 
  Steve,
  
  
  
  Telewest 
  should already allow the CLI transmission of your DDI range, without further 
  datafill changes. If it doesn't work you should check that you are sending the 
  appropriate number of digits.
  
  
  
  Try 
  sending:
  
  
  
  -3 digit 
  CLI
  
  -the 
  whole number (minus the leading zero)
  
  
  
  If the 
  comments above don't help please post a trace of an outgoing call and detail 
  the number, if any, that is presented to theCalled 
  Party.
  
  
  
  HTH
  
  
  
  Darren
  
  -- 
  
  
  Comgate
  
  TelcoInternetBroadcast
  
  
  
-Original 
Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]On Behalf Of Steve 
HanselmanSent: 01 July 
2004 09:57To: 
'[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Providing 
Telewest in the UK with per extension outbound callerID
Would 
be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept 
just our DDI if that was all I could get.

Steve


-Original 
Message-From: Storer, 
Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 09:35To: 
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Providing 
Telewest in the UK with per extension outbound callerID


Hi 
Steve,



SH 
Is anybody in the UK using Telewest as a PRI Telco 
provider?
SH 
Are you sending them caller ID?

Just a 
quick point of clarification before commenting further, do you wish to make 
calls via Telewest's network and send the CLI of your own DDI number range 
or do you wish to send "other numbers" as your CLI? If you are seeking 
toachieve the latter, what sort of numbers do you wish to propagate 
asthe CLI for your calls?

Regards
Darren
-- 

Comgate
TelcoInternetBroadcast
-Original 
  Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Steve 
  HanselmanSent: 30 June 
  2004 18:57To: 
  '[EMAIL PROTECTED]'Subject: [Asterisk-Users] Providing 
  Telewest in the UK with per extension outbound callerID
  Hi,
  
  Is anybody in the UK using 
  Telewest as a PRI Telco provider?
  
  Are you sending them caller 
  ID?
  
  I've been told by Telewest 
  that:-
  
  1. Oftel 
  doesn't allow them to accept caller ID (this is rubbish, and I replied 
  pointing out where in the link to Oftel that they sent me it was 
  stated. We need Type 2 caller ID) 
  2. Telewest can't do this. (this 
  is rubbish, I'm certain that some of our customers use Telewest and they 
  provide them with caller ID) 
  
  So, does anybody do this, and 
  if so, what did you have to request from them in order to enable it, and 
  what do you provide to them (how many digits and in what 
  format).
  
  Regards
  
  Steve
The information 
contained in this email is intended for the personal and confidential 
useof the addressee only. It may also be privileged information. If you 
are not the intendedrecipient then you are hereby notified that you have 
received this document in error andthat any review, distribution or 
copying of this document is strictly prohibited. If you have received 
this communication in error, please notify Brendata immediately on: 
+44 (0)1268 466100, or email '[EMAIL PROTECTED]' 
Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. 
SS13 1BX UKRegistered Office as above. Registered in England No. 
2764339See our current vacancies at 
www.brendata.co.uk
  The information contained in this 
  email is intended for the personal and confidential useof the addressee 
  only. It may also be privileged information. If you are not the 
  intendedrecipient then you are hereby notified that you have received this 
  documen

RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID

[Storer, Darren]

Hi 
Steve,

Try 
Telewest Provisioning Dept. on: 01483 582 966

HTH

Darren
-- 

Comgate
TelcoInternetBroadcast


  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Steve 
  HanselmanSent: 01 July 2004 10:16To: 
  '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] 
  Providing Telewest in the UK with per extension outbound 
  callerID
  
  When the original PBX 
  was installed we asked them to override the CLI and provide a single number as 
  the PBX couldn't provide the DDI number, now the contact at Telewest believes 
  it's somewhere between illegal and impossible to provide DDI numbers to the 
  outside world.
  
  
  -Original 
  Message-From: Storer, 
  Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 10:13To: 
  [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Providing 
  Telewest in the UK with per extension outbound callerID
  
  
  Hi 
  Steve,
  
  
  
  Telewest 
  should already allow the CLI transmission of your DDI range, without further 
  datafill changes. If it doesn't work you should check that you are sending the 
  appropriate number of digits.
  
  
  
  Try 
  sending:
  
  
  
  -3 digit 
  CLI
  
  -the 
  whole number (minus the leading zero)
  
  
  
  If the 
  comments above don't help please post a trace of an outgoing call and detail 
  the number, if any, that is presented to theCalled 
  Party.
  
  
  
  HTH
  
  
  
  Darren
  
  -- 
  
  
  Comgate
  
  TelcoInternetBroadcast
  
  
  
-Original 
Message-From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]On Behalf Of Steve 
HanselmanSent: 01 July 
2004 09:57To: 
'[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Providing 
Telewest in the UK with per extension outbound callerID
Would 
be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept 
just our DDI if that was all I could get.

Steve


-Original 
Message-From: Storer, 
Darren [mailto:[EMAIL PROTECTED] Sent: 01 July 2004 09:35To: 
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Providing 
Telewest in the UK with per extension outbound callerID


Hi 
Steve,



SH 
Is anybody in the UK using Telewest as a PRI Telco 
provider?
SH 
Are you sending them caller ID?

Just a 
quick point of clarification before commenting further, do you wish to make 
calls via Telewest's network and send the CLI of your own DDI number range 
or do you wish to send "other numbers" as your CLI? If you are seeking 
toachieve the latter, what sort of numbers do you wish to propagate 
asthe CLI for your calls?

Regards
Darren
-- 

Comgate
TelcoInternetBroadcast
-Original 
  Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Steve 
  HanselmanSent: 30 June 
  2004 18:57To: 
  '[EMAIL PROTECTED]'Subject: [Asterisk-Users] Providing 
  Telewest in the UK with per extension outbound callerID
  Hi,
  
  Is anybody in the UK using 
  Telewest as a PRI Telco provider?
  
  Are you sending them caller 
  ID?
  
  I've been told by Telewest 
  that:-
  
  1. Oftel 
  doesn't allow them to accept caller ID (this is rubbish, and I replied 
  pointing out where in the link to Oftel that they sent me it was 
  stated. We need Type 2 caller ID) 
  2. Telewest can't do this. (this 
  is rubbish, I'm certain that some of our customers use Telewest and they 
  provide them with caller ID) 
  
  So, does anybody do this, and 
  if so, what did you have to request from them in order to enable it, and 
  what do you provide to them (how many digits and in what 
  format).
  
  Regards
  
  Steve
The information 
contained in this email is intended for the personal and confidential 
useof the addressee only. It may also be privileged information. If you 
are not the intendedrecipient then you are hereby notified that you have 
received this document in error andthat any review, distribution or 
copying of this document is strictly prohibited. If you have received 
this communication in error, please notify Brendata immediately on: 
+44 (0)1268 466100, or email '[EMAIL PROTECTED]' 
Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. 
SS13 1BX UKRegistered Office as above. Registered in England No. 
2764339See our current vacancies at 
www.brendata.co.uk
  The information contained in this 
  email is intended for the personal and confidential useof the addressee 
  only. It may also be privileged information. If you are not the 
  intendedrecipient then you are hereby notified that you have received this 
  document in error andthat any review, distribution or copying of this 
  document is strictly prohibit

RE: [Asterisk-Users] T100P-E100P circuit board differences

[Storer, Darren]

Hi Scott,

SS Can someone please confirm that their E100P says T100P
SS on the artwork?

Yes.

HTH

Darren
-- 
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel
Sent: 29 June 2004 17:57
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] T100P-E100P circuit board differences


Hi-
 
Perhaps someone with an E100P in hand can answer this:

 
I just received an E100P from Digium (I normally buy quad boards)

I noticed that the circuit board says T100P on it, and I assume that the
T100P and E100P both use the same circuit board.

Can someone please confirm that their E100P says T100P on the artwork?

Thanks
Scott
 
 

Scott M. Stingel 
Emerging Voice Technology Inc.
Palo Alto, California and London, England

Email:  [EMAIL PROTECTED]  
URL:www.evtmedia.com   


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RE: [Asterisk-Users] SS7 to Pri

[Storer, Darren]

Hi Joseph,

J Does anyone know of a device that will take an SS7 link
J and convert it to a PRI?

Telesoft Technologies make the Okeford range of protocol converters and baby
switches that I have used for this purpose. Have a look at:

http://tinyurl.com/3drjp

If you are converting a number of SS7/PRI circuits at the same time the cost
per conversion comes down dramatically.


HTH

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Joseph
Sent: 25 June 2004 15:25
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SS7 to Pri


Does anyone know of a device that will take an SS7 link and convert it
to a PRI?


--
respectfully, Joseph - (606) 477-2355 x140
   --=

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RE: [Asterisk-Users] Call generator

[Storer, Darren]

http://lists.digium.com/pipermail/asterisk-users/2004-May/048245.html

--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of GIBERT
Frédéric
Sent: 23 June 2004 09:46
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Call generator


Hello,

Has someone know a good call generator for asterisk including SIP protocol
(freeware if possible)?
I need to stress a plateform and I don't find any.

Thanks by advance.



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RE: [Asterisk-Users] Outgoing CLI

[Storer, Darren]

Hi Simon,

the bad news is that you cannot change pridialplan on a per call basis (or
if you can I don't know how it's done). So even if setting
pridialplan=national works for your 0845 presentation number calls it's
unlikely to work for ordinary calls that present your geographic PSTN (0207)
DDI range.

I have just setup a test using a PRI into your carrier (using
pridialplan=national instead of local) and here's the result from a normal
call to a UK geographic PSTN number with a geographic PSTN DDI range CLI
being presented:

-- Making new call for cr 32775
 Protocol Discriminator: Q.931 (8)  len=44
 Call Ref: len= 2 (reference 7/0x7) (Originator)
 Message type: SETUP (5)
 Bearer Capability (len= 3) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
  Ext: 1  Trans mode/rate: 64kbps, circuit-mode
(16)
  Ext: 1  User information layer 1: A-Law (35)
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel Type:
3
   Ext: 1  Channel: 1 ]
 Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
 Called Number (len=14) [ Ext: 1  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0161XXX' ]
 Sending Complete (len= 0)
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 32775/0x8007) (Terminator)
 Message type: RELEASE COMPLETE (90)
 Cause (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location:
Public network serving the local user (2)
  Ext: 1  Cause: Invalid number format (28), class = Normal
Event (1) ]

The good news is that the NPI is exactly as you need (E.164) but as you can
see from the line above, when pridialplan=local is NOT used, the call is
rejected with release cause 28 (Invalid number format). Some switches accept
a National TON and others don't, it varies from carrier to carrier. The
strange part is that the geographic DDI number that I used for the Calling
Number does not appear on the trace above; does anyone know if this is a bug
with * ?
NB. I tried the Calling Number both with and without a leading 0.

Maybe your datafill for this PRI is now different (after your request for an
08XX presentation number to the carrier) and it will accept TON: National
Number instead of TON: Subscriber Number for all calls. Please drop a
note to the list and let us know what happens when you make the changes. Use
'pri debug span 1' to gather some trace information if you need to post more
detail back with your next e-mail.

HTH

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Thilo Salmon
Sent: 23 June 2004 14:49
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Outgoing CLI


 User must provide - TON = national or international

Add pridialplan=national before just above your channel =... line in
zapata.conf to set TON to national for outgoing calls. NPI will be
always be set to E.164 afaik. Now set callerid to 845 or 870 without the
leading 0.

Thilo

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RE: [Asterisk-Users] Testing UK emergency dialing and LCR.

[Storer, Darren]

Hi Adam,

AG (BTW, why is it that people used 000, 999, 911, etc for EMERGENCY
AG calls (every second counts) when we used to dial from rotary dial
AG phones, where dialling a 0 or 9 takes a long time compared to
AG dialling a 1Why didn't we all use 111, or something similar?)

In the UK (c.1935) 111 was rejected because faulty rotary phones could
easily dial that number by accident. Ultimately 999 was chosen because it
wasn't hard to remember and it was easy/cheap to modify coin operated public
phones not to charge for 999 calls.

Regards

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adam
Goryachev
Sent: 21 June 2004 09:59
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Testing UK emergency dialing and LCR.


On Sat, 2004-06-19 at 19:27, Storer, Darren wrote:
 Hi Kevin,

 KW By the way, it's useful to map 911 and 112 onto your 999
 KW route for the benefit of foreigners who don't know any better.


Well, while you are at it, you might as well add-in 000, because that is
what we use.

(BTW, why is it that people used 000, 999, 911, etc for EMERGENCY calls
(every second counts) when we used to dial from rotary dial phones,
where dialling a 0 or 9 takes a long time compared to dialling a 1
Why didn't we all use 111, or something similar?)

Regards,
Adam

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RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk

[Storer, Darren]

Steve,

your config description (timing) does sound odd. Could you re-post your
revised config files?

Thanks

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 20 June 2004 16:18
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk


I've changed the zaptel.conf to set both as internal, and it now seems to
work, which is backwards to the config I thought it should have been, I
would have thought that the Telewest PRI would have been 1 and the GDK 0?

Can somebody confirm that this is the correct definition for timing, if it's
a +ve number then it's external clocking with the lowest 1 being the highest
priority.

All spans are clocked relative to the external source and the external
source selected is the lowest priority number that is currently being
clocked?

I'll experiment some more.



-Original Message-
From: Yifang Dai [mailto:[EMAIL PROTECTED]
Sent: 19 June 2004 03:22
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problems with faxing via TE405P/Asterisk

Let's try again, missed a line in the last reply...

On Fri, Jun 18, 2004 at 09:55:24PM -0400, Yifang Dai wrote:
 On Fri, Jun 18, 2004 at 09:27:52PM +0100, Steve Hanselman wrote:
 
 LG GDK-186 PBX --PRI---  TE405P/Asterisk  ---PRI--- Telewest (Telco
 provider)
 

 --snip---

 Any ideas on where to start?


This is most likely to be a timing issue. You need to make sure your
asterisk is get timing from your telco, and provide timing for you gdk
pbx. /etc/asterisk/zaptel.conf is the place to look.

--
Yifang Dai
Senior System Administrator
Yarde Metals Inc
45 Newell St, Southington, CT 06010
(Phone) 860-406-6107; (FAX) 860-406-4060
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The information contained in this email is intended for the personal and
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of the addressee only. It may also be privileged information. If you are not
the intended
recipient then you are hereby notified that you have received this document
in error and
that any review, distribution or copying of this document is strictly
prohibited. If you have
received  this communication in error, please notify Brendata immediately
on:

+44 (0)1268 466100, or email '[EMAIL PROTECTED]'

Brendata (UK) Ltd
Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UK
Registered Office as above. Registered in England No. 2764339

See our current vacancies at www.brendata.co.uk
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RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk

[Storer, Darren]

Hi Steve,

How bizarre, your config doesn't look like it should work too well and
certainly doesn't look like it should improve your fax problem!

I assume that pri_cpe is set for span1 and pri_net for span2 ?

Maybe, just maybe, Telewest reconfigured your PRI to look for clocking from
your CPE but I've not encountered that configuration before. Try and leave
the current config up for as long as you can before you return it to
production mode and watch the CLI/logs to see if you get any sporadic clock
slips (within a couple of hours I'd expect at least one episode of
messages).

One last thought, did you bounce the system after you made the changes to
zaptel.conf or did you just reload * ?

HTH

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 20 June 2004 16:48
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk


They look odd to me for sure, I'm certain (99.9%) that Telewest would not
clock off of us, but as far as I can see, the current config (which allows
the GDK to send and receive faxes) has no external clocking???

Here's the current config:

span=1,0,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4

Here's the original config which I took to mean that Telewest provided clock
and span2 clocked off span1?

span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4


(Span1 goes to Telewest - our Telco provider, span2 goes to our current PBX,
a GDK-186)


Steve

-Original Message-
From: Storer, Darren [mailto:[EMAIL PROTECTED]
Sent: 20 June 2004 16:34
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk

Steve,

your config description (timing) does sound odd. Could you re-post your
revised config files?

Thanks

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 20 June 2004 16:18
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk


I've changed the zaptel.conf to set both as internal, and it now seems to
work, which is backwards to the config I thought it should have been, I
would have thought that the Telewest PRI would have been 1 and the GDK 0?

Can somebody confirm that this is the correct definition for timing, if it's
a +ve number then it's external clocking with the lowest 1 being the highest
priority.

All spans are clocked relative to the external source and the external
source selected is the lowest priority number that is currently being
clocked?

I'll experiment some more.



-Original Message-
From: Yifang Dai [mailto:[EMAIL PROTECTED]
Sent: 19 June 2004 03:22
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problems with faxing via TE405P/Asterisk

Let's try again, missed a line in the last reply...

On Fri, Jun 18, 2004 at 09:55:24PM -0400, Yifang Dai wrote:
 On Fri, Jun 18, 2004 at 09:27:52PM +0100, Steve Hanselman wrote:
 
 LG GDK-186 PBX --PRI---  TE405P/Asterisk  ---PRI--- Telewest (Telco
 provider)
 

 --snip---

 Any ideas on where to start?


This is most likely to be a timing issue. You need to make sure your
asterisk is get timing from your telco, and provide timing for you gdk
pbx. /etc/asterisk/zaptel.conf is the place to look.

--
Yifang Dai
Senior System Administrator
Yarde Metals Inc
45 Newell St, Southington, CT 06010
(Phone) 860-406-6107; (FAX) 860-406-4060
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RE: [Asterisk-Users] Testing UK emergency dialing and LCR.

[Storer, Darren]

Hi Kevin,

KW By the way, it's useful to map 911 and 112 onto your 999
KW route for the benefit of foreigners who don't know any better.

Your point about 112 is very useful but slightly misguided; although the UK
has used 999, nationally since 1938, (the world's first single number access
for emergency services) 112 was mandated for pan European use from 1992
onwards. 112 is *not* for foreigners who don't know any better, it's for
everyone in the EU to learn so that when you are anywhere in the EU you
stand a fighting chance of getting hold of emergency help at the first
attempt. 999 will continue to run in parallel with 112 for many years to
come but 112 should be taught to children and adults alike as the universal
number for emergency services. Some UK Telcos also provided support for 911
for a little while but I believe that this was officially frowned upon; I'm
not sure what the policy is now.

W As another thing, what is the correct method when using least cost
W routing... If you have a branch office that has no outside line
W connectivity directly routing its calls over IP to HQ the other end of
W the country when you dial 999 it gets handled by the local call center
W to your HQ rather than the branch office.

It became apparent, back in 1999, when I was part of a team providing
consultancy to a UK Telco for VoIP VPN launch, that a POTS line would be
required locally at each branch office for power-fail compliance and to
ensure that the OACs (Operator Assistance Centres) did not get confused
about which location the emergency call was originated from. We discussed
spoofing the branch office CLI in network at an SS7 level but that idea
was shelved as there would have to have been an associated POTS line entry
in the OAC database in the first place. At that time Cisco CPE had no way of
utilising the power-fail POTS lines so a red 'phone was provided for use on
each floor of the branch offices that only had VoIP VPN telephony.

HTH

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kevin Walsh
Sent: 19 June 2004 02:56
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Testing UK emergency dialing and LCR.


Wayne [EMAIL PROTECTED] wrote:
 Just wondering how people test your emergency dialing in the UK.
 Obviously you need to dial the 999 for emergency services, but am a bit
 unsure if this would go down too well with the operator with a 'sorry
 just testing' call. (you do all /test/ your emergency dialing dont
 you!?:-) )

I tend to test by unplugging the phone line and dialling 999.
You can watch the log and see that the call attempted to route
to the POTS line.  You can then dial a real POTS number and
watch the same route succeed.

The emergency services get very upset if you call them to test,
unless you've arranged to do so in advance and have an allotted
time slot.

You're right though; you can't be absolutely sure that the 999
route will work until you test it with a real call.  Just start
a fire before you call.  That'll probably work. :-)


 As another thing, what is the correct method when using least cost
 routing... If you have a branch office that has no outside line
 connectivity directly routing its calls over IP to HQ the other end of
 the country when you dial 999 it gets handled by the local call center
 to your HQ rather than the branch office.

If you need emergency services access in your branch office then
you should get a single line into that office.  The emergency
services tend to rush to the destination they know is correct for
that phone number.

By the way, it's useful to map 911 and 112 onto your 999 route
for the benefit of foreigners who don't know any better.

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RE: [Asterisk-Users] PRI problems (telewest - * - LG GDK 186)

[Storer, Darren]

Hi Steve,

please could you post your zapata.conf and zaptel.conf files?

Regards

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 15 June 2004 10:28
To: '[EMAIL PROTECTED]'
Subject: [Asterisk-Users] PRI problems (telewest - * - LG GDK 186)


Hi,
 
I'm trying to figure out what the issue is splicing Asterisk between our
Telewest PRI and a GDK-186 with a PRI card.
 
We're using the Digium TE405P
 
Our telco provider is Telewest, and Telco directly into switch is fine.
 
When I splice Asterisk in, I can make and receive calls from Asterisk
extensions, I can make outbound calls from the GDK, but inbound calls do not
seem to pass over the called number, therefore in the GDK they go to a
default extension, and not to the DDI required.
 
The presentation is 3 digit, I've made an entry in extensions.conf to allow
the looping of a call back from the GDK.
 
Here's the relevant extension info from extensions.conf:
 
[gdk]
;
; This context is used for all calls coming in from the GDK, they are meant
to go directly out of the telewest PRI
;
 
exten = s,1,Answer ; Answer the line
exten = s,2,DigitTimeout,2 ; Set Digit Timeout to 5 seconds
exten = s,3,ResponseTimeout,2  ; Set Response Timeout to 10 seconds
 
exten = _X.,1,Dial(${TELEWEST}/${EXTEN})
exten = _X.,2,Congestion
 
exten = _4661XX,1,dial(${GDK}/${EXTEN:3})
exten = _4661XX,2,Congestion
 
So from the GDK I can dial 9466110 and I should be looped back to the GDK
and into the support queue, the loop works, but the called number IE is not
understood by the GDK?
 
Looking in the technical book for the GDK it states that the requirement is
that the information element Called Number contains the required digits,
and it does, I'm stumped?
 
 
Here's the debug of the session - slightly edited to reduce the size(!):
 
 
Linux3*CLI pri debug span 2
Enabled debugging on span 2
 Protocol Discriminator: Q.931 (8)  len=22
 Call Ref: len= 2 (reference 32527/0x7F0F) (Originator)
 Message type: SETUP (5)
 [04 03 80 90 a3]
 Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
  Ext: 1  Trans mode/rate: 64kbps, circuit-mode
(16)
  Ext: 1  User information layer 1: A-Law (35)
 [18 03 a9 83 8f]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
    ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel Type:
3
   Ext: 1  Channel: 15 ]
 [6d 05 80 50 31 31 31]
 Calling Sub-Address (len= 7) [ Ext: 1  Type: NSAP (X.213/ISO 8348 AD2) (0)
O: 0 '111' ]
-- Making new call for cr 32527
-- Processing Q.931 Call Setup
-- Processing IE 4 (Bearer Capability)
-- Processing IE 24 (Channel Identification)
-- Processing IE 109 (Calling Party Subaddress)
 Protocol Discriminator: Q.931 (8)  len=14
 Call Ref: len= 2 (reference 65295/0xFF0F) (Terminator)
 Message type: SETUP ACKNOWLEDGE (13)
 [18 03 a9 83 8f]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
    ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel Type:
3
   Ext: 1  Channel: 15 ]
 [ [1e [1e 02 [1e 02 81 [1e 02 81 82 [1e 02 81 82]
 Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0:
0   Location: Private network serving the local user (1)
   Ext: 1  Progress Description: Called
equipment is non-ISDN. (2) ]
    -- Accepting overlap call from '' to 'unspecified' on channel 0/15,
span 2
    -- Starting simple switch on 'Zap/46-1'
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 32527/0x7F0F) (Originator)
 Message type: INFORMATION (123)
 [70 02 80 34]
 Called Number (len= 4) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
Unknown Number Plan (0) '4' ]
-- Processing IE 112 (Called Party Number)
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 32527/0x7F0F) (Originator)
 Message type: INFORMATION (123)
 [70 02 80 36]
 Called Number (len= 4) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
Unknown Number Plan (0) '6' ]
-- Processing IE 112 (Called Party Number)
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 32527/0x7F0F) (Originator)
 Message type: INFORMATION (123)
 [70 02 80 36]
 Called Number (len= 4) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
Unknown Number Plan (0) '6' ]
-- Processing IE 112 (Called Party Number)
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 32527/0x7F0F) (Originator)
 Message type: INFORMATION (123)
 [70 02 80 31]
 Called Number (len= 4) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
Unknown Number Plan (0) '1' ]
-- Processing IE 112 (Called Party Number)
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 32527/0x7F0F) (Originator)
 

RE: [Asterisk-Users] PRI problems (telewest - * - LG GDK 186)

[Storer, Darren]

Hi Steve,

SH The presentation is 3 digit, I've made an entry in extensions.conf
SH to allow the looping of a call back from the GDK.

3 Digits seems a bit short from Telewest, I would have expected the last six
digits to have been sent for inbound PSTN calls (as per the BT standard). If
Telewest are sending six digits then it would seem best use the same
standard from * to the GDK as well, rather than 3. From your trace *
indicates that it's calling 110 which seems to match your expectation. Can
you make the following changes in the * config files, change the digit
length to match Telewest and try again?

pridialplan=local /* all entries
switchtype=euroisdn   /* all entries

In my working system configs there is always a switchtype declaration for
each span; this may be overkill but it works and certainly won't cause any
harm.

Is there a reason why you have only provisioned half your B channels on each
PRI span?; if that's all you need then that's fair enough.

HTH

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 15 June 2004 12:59
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] PRI problems (telewest - * - LG GDK 186)


Certainly, here they are (I've stripped the commented bits away):

Zapata.conf

[trunkgroups]
[channels]
language=en
context=default
switchtype=national
overlapdial=yes
signalling=fxo_ls
rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

;
; For our config, port 1 goes to Telewest, port 2 goes to the GDK
;
pridialplan=unknown
switchtype = euroisdn
signalling = pri_cpe
context = telewest
group = 1
overlapdial=no
callerid=asreceived
channel = 1-15

pridialplan=national
signalling = pri_net
context = gdk
group = 2
overlapdial=yes
callerid=asreceived
channel = 32-46


Here's zaptel.conf (timing only on span 1 as that's the one from Telewest):

span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=32-46
dchan=47
defaultzone=uk

-Original Message-
From: Storer, Darren [mailto:[EMAIL PROTECTED]
Sent: 15 June 2004 12:42
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] PRI problems (telewest - * - LG GDK 186)

Hi Steve,

please could you post your zapata.conf and zaptel.conf files?

Regards

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 15 June 2004 10:28
To: '[EMAIL PROTECTED]'
Subject: [Asterisk-Users] PRI problems (telewest - * - LG GDK 186)


Hi,
 
I'm trying to figure out what the issue is splicing Asterisk between our
Telewest PRI and a GDK-186 with a PRI card.
 
We're using the Digium TE405P
 
Our telco provider is Telewest, and Telco directly into switch is fine.
 
When I splice Asterisk in, I can make and receive calls from Asterisk
extensions, I can make outbound calls from the GDK, but inbound calls do not
seem to pass over the called number, therefore in the GDK they go to a
default extension, and not to the DDI required.
 
The presentation is 3 digit, I've made an entry in extensions.conf to allow
the looping of a call back from the GDK.
 
Here's the relevant extension info from extensions.conf:
 
[gdk]
;
; This context is used for all calls coming in from the GDK, they are meant
to go directly out of the telewest PRI
;
 
exten = s,1,Answer ; Answer the line
exten = s,2,DigitTimeout,2 ; Set Digit Timeout to 5 seconds
exten = s,3,ResponseTimeout,2  ; Set Response Timeout to 10 seconds
 
exten = _X.,1,Dial(${TELEWEST}/${EXTEN})
exten = _X.,2,Congestion
 
exten = _4661XX,1,dial(${GDK}/${EXTEN:3})
exten = _4661XX,2,Congestion
 
So from the GDK I can dial 9466110 and I should be looped back to the GDK
and into the support queue, the loop works, but the called number IE is not
understood by the GDK?
 
Looking in the technical book for the GDK it states that the requirement is
that the information element Called Number contains the required digits,
and it does, I'm stumped?
 
 
Here's the debug of the session - slightly edited to reduce the size(!):
 
 
Linux3*CLI pri debug span 2
Enabled debugging on span 2
 Protocol Discriminator: Q.931 (8)  len=22
 Call Ref: len= 2 (reference 32527/0x7F0F) (Originator)
 Message type: SETUP (5)
 [04 03 80 90 a3]
 Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
  Ext: 1  Trans mode/rate: 64kbps, circuit-mode
(16)
  Ext: 1  User information layer 1: A-Law (35)
 [18 03 a9 83 8f]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0

RE: [Asterisk-Users] GSM to ISDN or TAPI

[Storer, Darren]

Hi Chris,

from what you've described below it sounds like a Nokia device with a serial
connection to a host will provide the CC features you need. Much as I like
IVR applications the ubiquitous GSM handset makes portable SMS access easy,
secure and the output can be stored for future reference.

With a little forethought you can design a simple protocol that can use
either a central GUI front-end or direct SMS input from an engineer's own
handset; authentication can still be performed against CLI and a PIN
contained within the SMS.

The results should be less prone to mis-keying as commands can be reviewed
before execution/transmission and the results can be sent back in writing
for subsequent reference. Another benefit with SMS is remote alarm handling
which is easier to guarantee because the messages are sent via a store and
forward system, so the recipient's handset does not have to be powered on
24x7 in order to ensure receipt of all alarms.

Both Nokia Premicell (22) and Ericsson equivalents are currently being sold
cheaply on UK eBay.

Good Luck.

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Lee
Sent: 10 June 2004 22:54
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] GSM to ISDN or TAPI


Storer, Darren wrote:
 Hi Chris,

 CL Does the incoming DTMF and voice work over the serial
 CL interface with the 22?

 I can't help but feel that you are going about this all the wrong way
(based
 upon the limited information you have chosen to share with us). If you
need
 to pass control information from one node to another and you have a pair
of
 Nokia 22s then why not simply send an SMS message? Maybe this is not a
good
 solution for you but until you fill in the gaps it's the best I can come
 up with.
 IE. Tell us a bit more about what you are trying to achieve.

I am going to have some remote machines which need to have adjustments
made to their settings on occasion, the most cost effective and user
friendly way I can come up with is a simple IVR system that says press 1
to set limits on flow, press two for flow status report etc.
It will use a combination of CLID and pin number to authenticate the
engineer doing the config.
So what I need is something that can take a pay as you go sim (least
cost for line rental) and accepts calls without too many problems.
As I said earlier I had a Nokia 32;
I plugged it into a windows box with USR voice modems and it would not
work at all, it only provides a dial tone when connected to select
hardware, certain phones and one or two winmodems, I could not justify
the cost of the X100p for testing based on the mixed results.
So now I am looking at finding some other method of linking the GSM to
the PBX, I like the concept of digital all the way and RS-232 looks
like voice modem would be great.

Hope that clears up my requirements.

Thanks for the help

Chris.
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RE: [Asterisk-Users] GSM to ISDN or TAPI

[Storer, Darren]

Hi Philipp,

PK BUT: There is no guarantee that a SMS message reaches
PK the destination; that's how the SMS network is designed.

That's why we implement higher level protocols that send an ack when alarm
delivery is vital. If the ack fails to arrive before timer expiry, we
resend the alarm; it's not rocket science! If the alarm is not service
affecting (Eg. information alarm) no ack is usually required. You pays
your money and makes your choice. Anyone with limited experience of
protocol design, whether the transport is IP, V.24 etc., should be familiar
with these basic principles.

Other techniques that can be used to implement acknowledged SMS
transmissions between IT systems include the use of delivery reports (where
supported). As with Asterisk, there are many ways to achieve the same goal,
it's more a question of finding the combination that best suits your needs.

Just my 2ds

Darren
--
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Philipp von
Klitzing
Sent: 11 June 2004 16:12
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] GSM to ISDN or TAPI


Hi!

 for subsequent reference. Another benefit with SMS is remote alarm
handling
 which is easier to guarantee because the messages are sent via a store and
 forward system

BUT: There is no guarantee that a SMS message reaches the destination;
that's how the SMS network is designed.

Cheers, Philipp

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RE: [Asterisk-Users] GSM to ISDN or TAPI

[Storer, Darren]

Hi Chris,

CL All I want is two GSM lines that look like voice modems to
CL the PC and provide full telephony interface, that is DTMF
CL both ways CLI and a few other bits and pieces.

We use the Nokia 22:

http://www.nokia.com/nokia/0,,56024,00.html

They have worked well providing both telephony applications on remote sites
and SMS support for Broadcast work in the UK (serial AT command interface).

If you don't mind single band (900 or 1800 MHz GSM) operation there is an
older device (Nokia Premicell) that can be sourced cheaply from eBay:

http://www.nokia.com/cda1/0,1080,2700,00.html

HTH

Darren
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Lee
Sent: 10 June 2004 15:50
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] GSM to ISDN or TAPI


Hi
I am in the UK and am looking for a device that will allow me to connect
two sim cards (read wireless lines) to either the port on the back of my
fritz card or any other connection direct to the PC that provides a
usable telephony interface.
I will even plug two devices into a windows box and have that do ISDN to
ISDN if required.
All I want is two GSM lines that look like voice modems to the PC and
provide full telephony interface, that is DTMF both ways CLI and a few
other bits and pieces.
I am looking to using asterisk as a remote IVR for looking after some
equipment, but land lines are a problem.
Any help is much appreciated
Regards

Chris.
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RE: [Asterisk-Users] Automating calls

[Storer, Darren]

Hi Simon,

SG I have heard that i can put a file in a certain directory
SG to get * to initiate a call. Is this true ? if so where
SG would i look ?

It *really* is time that you got to grips with voip-info.org. There are many
gems in there; I typed in auto dial out and pressed the search button,
have a look at what came back:

http://www.voip-info.org/wiki-Asterisk+auto-dial+out

;-)

HTH

Darren
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Simon
Sent: 10 June 2004 16:28
To: Asterisk-Users
Subject: [Asterisk-Users] Automating calls


Hello

I have heard that i can put a file in a certain directory to get * to
initiate a call.

Is this true ? if so where would i look ?

Best Regards
Simon Garvey



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RE: [Asterisk-Users] GSM to ISDN or TAPI

[Storer, Darren]

Hi Chris,

CL Does the incoming DTMF and voice work over the serial
CL interface with the 22?

I can't help but feel that you are going about this all the wrong way (based
upon the limited information you have chosen to share with us). If you need
to pass control information from one node to another and you have a pair of
Nokia 22s then why not simply send an SMS message? Maybe this is not a good
solution for you but until you fill in the gaps it's the best I can come
up with.
IE. Tell us a bit more about what you are trying to achieve.

HTH

Darren
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Lee
Sent: 10 June 2004 17:13
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] GSM to ISDN or TAPI


Storer, Darren wrote:
 Hi Chris,

 CL All I want is two GSM lines that look like voice modems to
 CL the PC and provide full telephony interface, that is DTMF
 CL both ways CLI and a few other bits and pieces.

 We use the Nokia 22:

 http://www.nokia.com/nokia/0,,56024,00.html

 They have worked well providing both telephony applications on remote
sites
 and SMS support for Broadcast work in the UK (serial AT command
interface).

 If you don't mind single band (900 or 1800 MHz GSM) operation there is an
 older device (Nokia Premicell) that can be sourced cheaply from eBay:

 http://www.nokia.com/cda1/0,1080,2700,00.html


Does the incoming DTMF and voice work over the serial interface with the 22?
I had a Nokia 32 for test and could not get it to return DTMF, it has AT
commands to generate DTMF and to receive CLI but I could not get it into
voice mode or get DTMF out of it.

Thanks for your help

Chris.
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RE: [Asterisk-Users] Configuring cisco 7940

[Storer, Darren]

Hi Tony,

TH First thing I think I need to do is work out how to set the
TH TFTP server IP as it's using the wrong one (it's ignoring
TH the setting in the DHCP server).

http://tinyurl.com/37fe4

HTH

Darren
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tony Hoyle
Sent: 05 June 2004 19:22
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Configuring cisco 7940


I've just managed to get hold of a cisco 7940, which looks nice but I'm
unable to make it actually do anthing...!

All the online manuals say things like see your network administrator
which isn't a whole lot of use.

First thing I think I need to do is work out how to set the TFTP server
IP as it's using the wrong one (it's ignoring the setting in the DHCP
server).  When you point a browser at the phone it gives you the
settings but no opportunity to set them.

Also, what is the code of the $8 support option and who sells it (it
seems cisco don't sell direct to end users)?  The cheapest I've seen is
$100 and if it's that kind of price I'll just see how far I can get with 
the default firmware.

Tony



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RE: [Asterisk-Users] Configuring cisco 7940

[Storer, Darren]

TH Unfortunately those instructions don't seem to relate
TH to my phone (eg. there's no option 6 on the 'Settings'
TH menu).

Sorry Tony, those instructions work well for 12SP and VIP30 phones (although
you have to know to use 1 to activate your changes as you exit at the end of
the sequence).

I'm sure one of the other list readers will be able to help - good luck!

Darren
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tony Hoyle
Sent: 05 June 2004 21:56
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Configuring cisco 7940


Storer, Darren wrote:

 http://tinyurl.com/37fe4

Unfortunately those instructions don't seem to relate to
my phone (eg. there's no option 6 on the 'Settings' menu).

I've found some other documents which seem to help but am unable to
change any of the settings even in the unlocked state - it all seems to
be hardcoded.

I eventually gave up and installed an extra tftp server so I could get
an XMLDefault file onto it.  Now for some reason it's trying to query
the router for something (which isn't going to get very far as it's just
a Netgear gateway).  I'll try a few packet traces to see if I can fake
the responses... presumably as shipped they assume you're running cisco
routers etc.

Tony


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RE: [Asterisk-Users] TE410P Q.931

[Storer, Darren]

Hi Simon,

the following changes are important:

/etc/asterisk/zapata.conf

switchtype=euroisdn /* (uncomment this line)
pridialplan=local

and

/etc/zaptel.conf

span=1,1,0,ccs,hdb3,crc4
span=2,2,0,ccs,hdb3,crc4
span=3,3,0,ccs,hdb3,crc4
span=4,4,0,ccs,hdb3,crc4

(The lines above assume that all your PRIs will come from a Telco's switch)

After you make the changes don't forget to 'init 6' the box.

HTH

Darren
-- 
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TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Simon
Sent: 03 June 2004 15:43
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] TE410P Q.931


Got this in
/etc/asterisk/zapata.conf
[channels]
context=default
signalling=pri_cpe
;switchtype=euroisdn
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
rxgain=0.0
txgain=0.0
immediate=no
musiconhold=default

pridialplan=international
; Channels inherit configuration above them
; Span 1

group=1
channel = 1-15
channel = 17-31


signalling=pri_cpe
;switchtype=euroisdn

group=2
channel = 32-46
channel = 48-62
signalling=pri_cpe
;switchtype=euroisdn

group=3
channel = 63-77
channel = 79-93
signalling=pri_cpe
;switchtype=euroisdn
group=4
channel = 94-108
channel = 110-124

and this in
etc/zaptel.conf

span=1,0,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
bchan=1-15
bchan=17-31
bchan=32-46
bchan=48-62
bchan=63-77
bchan=79-93
bchan=94-108
bchan=110-124
dchan=16
dchan=47
dchan=78
dchan=109
loadzone = uk
defaultzone=uk

would this look right ?
using zttool i am seeing span 1 OK span 2 YELLOW

only got 1  2 plugged in

Simon

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Apollon
Koutlides
Sent: 03 June 2004 14:17
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] TE410P Q.931


Simon wrote:

Can anyone help i have * running on debian with a te410p , my telco tells
me
i need it to run in Q.931 anyone know how to make this happen ?


That's the Layer 2 protocol, PRI signalling. You would obviously do CPE
signalling (insert a line signalling=pri_cpe in
/etc/asterisk/zaptel.conf) - first you need to get Layer 1 up, of
course, and define which channel to be used for signalling in
/etc/zapata.conf ( dchan=XX ).

Apollon Koutildes
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RE: [Asterisk-Users] Dialogic D/41E

[Storer, Darren]

Hi Philip,

PK Is a D/41E usable w/Asterisk?  If so how does one obtain the drivers?
PK Or is it a better pots adapter for the wastebasket?

If your card is older and does not have a JCT suffix then it will not work
using the Digium Dialogic drivers for Asterisk.

HTH

Darren
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kubat, Philip
Sent: 02 June 2004 18:47
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dialogic D/41E


I have an old Dialogic D/41E card.  I searched the mailing list and it
looks like there was or could be a module for it.  Although the posts never
specified where or how.

Is a D/41E usable w/Asterisk?  If so how does one obtain the drivers?  Or is
it a better pots adapter for the wastebasket?

Thanks
Phil

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RE: [Asterisk-Users] Feature request for integrating an OSS (Operations Support System) and Asterisk

[Storer, Darren]

Hi Nathan,

The killer app feature that's missing (IMHO) is a provisioning interface.

I'd like to see the following features implemented:

 - initial provision
 - move and changes
 - cease

(All with the appropriate links to the billing and customer care modules).

The provisioning interface should ideally have a GUI for internal staff to
use and also an API for external Web logic to interface. The external
provisioning API would be a great benefit to companies that need to
implement self provisioning services for their customers.

Good luck with the project!

Darren
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nathan
Sent: 02 June 2004 13:44
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Feature request for integrating an OSS
(Operations Support System) and Asterisk


Hi,

I work for an ISP/CLEC, and we have developed our own OSS (Operations
Support System), which handles all billing, sales, provisioning, and
support issues. When it was originally being designed, the idea was to
integrate it with Asterisk.

Other than Caller-ID information (so that past trouble tickets, and
billing issues can be brought up for the agent), how else would the
Asterisk community like the OSS we developed and Asterisk to interact
(perhaps transferring calls, etc.)?

More information about the OSS is here:

http://www.vylink.com/oss/

Also, if you have any other suggestions for features that aren't on the
webpage, feel free to email [EMAIL PROTECTED] If there is enough demand for
a feature we don't have or we like the feature enough, we will likely add
it.

Thanks,
Nathan
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RE: dialogic was RE: [Asterisk-Users] Glare condition - How well does asteriskhandle?

[Storer, Darren]

Hi Steve,

SU If you are using CTR4, then I guess they use CTR4. :-)
SU CTR4 ==  Net 5 == various other names == EuroISDN.

Attempt #2: Reasonable logic and _good_ assumption (this time)...

It would appear that Manchester IS the NTL region in the UK that supports
ISDN 110 (ETSI EuroISDN etc.). Maybe this is because Manchester was an old
Mercury/CW operation before it changed hands to NTL?!? I know they have DMS
and Nokia switch equipment so maybe they don't have the Marconi System X
ISDN 85 service for Q.931 deployment...

So, it's good news for Tim, who can now deploy * without risk. However it's
bad news for ISDN 85 customers as we failed to obtain the traces we need to
move the incompatibility problem forward when we set-up * and an analyser
today.

One interesting find was that CRC4 was not enabled on the Q.931 circuit from
NTL (despite the standards) at Tim's site, something to remember if you're
planning an install on an NTL PRI in Manchester.

It's looking like a stint in a switch site is required to gather the traces
unless anyone else in NW UK has an ISDN 85 PRI we could run a quick test
against?

Regards

Darren
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Storer,
Darren
Sent: 28 May 2004 09:32
To: [EMAIL PROTECTED]
Subject: RE: dialogic was RE: [Asterisk-Users] Glare condition - How
well does asteriskhandle?


Hi Steve,

SU If you are using CTR4, then I guess they use CTR4. :-)
SU CTR4 ==  Net 5 == various other names == EuroISDN.

Reasonable logic but bad assumption in this case.

The Dialogic Q.931 stack (D/300, DM3 etc.) is solid and quite tolerant of
ISDN 85 as are most hardware PBXs. Other (PC based) products exhibit exactly
the same fussy behaviour though; the Digi RAS products (
http://tinyurl.com/36e7l ) work well with EuroISDN but won't work with ISDN
85 so the Asterisk stack is not alone in freaking when presented with this
Frankenstein Protocol of the ISDN world. (Thanks a bunch BT/Marconi/GPT et
al who rushed ISDN85 into service because they didn't want 18 months of
effort to delay real Q.931 deployment in the UK, so they bolted a protocol
converter on the end of existing DASS line cards instead of developing a
native solution...ugly stuff!)

I would like to try to help Tim decide which version of PRI he has as I'm
local to him, let's see if he takes me up on the offer to plug a working *
box into his PRI... Even if he has ISDN85 we would still benefit from the
chance to capture the failure (using an MPA) and compare it to some good
(working) * traces from a real EuroISDN circuit. Then the fun starts trying
to find a neat way to patch the stack...

Regards

Darren
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Underwood
Sent: 28 May 2004 01:39
To: [EMAIL PROTECTED]
Subject: Re: dialogic was RE: [Asterisk-Users] Glare condition - How
well does asteriskhandle?


tim panton wrote:

 Steve Underwood wrote:

 Jason Williams wrote:

 At 09:16 27/05/2004 -0500, you wrote:

 Maybe the time and effort would be better spent finding out why the
 Digium card won't work on the NTL's PRI and either fixing it or
 providing the information and testing facility to someone who can.





 NTL's PRI uses ISDN 85  not q931 so a ne protocol stack would need
 to be written.



 I think you means ISDN 85 not EuroISDN.

 Good heavens. I thought ISDN 85 died out in about 89. :-)

 I don't know where you would get the spec these days, but it
 shouldn't be a lot of work to modify libpri to add another variant of
 ISDN.


 I should say that I don't _know_ what NTL are delivering me,
 I haven't (yet) tried it with a digium E1 card.

 What I do know is :
 1) the dialogic card claims to be running CTR4 on an E1 ISDN PRI
 2) other folks on this list have had difficulty getting digium
 cards to talk to NTL.
 3) exactly the same dialogic config works on BT and the Dutch
 PTT's E1 lines.

 So it _may_ not be a problem for me as NTL is a patchwork of smaller
 telcos, my area (Manchester) may be more up to date.

 Anyone know an easy way to tell what I've got ?
 (or will I have to ask NTL -gh)

 T.

If you are using CTR4, then I guess they use CTR4. :-)

CTR4 ==  Net 5 == various other names == EuroISDN.

It sounds like you are OK.

Regards,
Steve

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RE: [Asterisk-Users] Crc4 issues

[Storer, Darren]

Hi Paulo,

PM This is our 2nd E1 client that we try to use crc4 either with the
PM e100p or with the e405p without luck.

PM After some trials, we ask the telco to switch off crc4 on their side
PM and everything works flawlessly.

[span=1,1,0,ccs,hdb3,crc4,yellow] looks good as it uses CRC4 and sets the
timing to be synchronised with the clock coming in from your Telco's switch.
You do not mention what sort of switch you are trying to connect to and what
sort of physical cabling (including length) is used for connection to the
Telco (Coax, baluns, 120 Ohm RJ45 etc.)???

On the occasions where CRC4 has proved to be a major problem from Asterisk
to the Telco's switch, bad cable termination on the frame proved to be the
problem and as soon as the connections were re-made properly CRC4 worked
perfectly.

I would also refer you to a recent comment from Critch who advised that
Asterisk systems should be power cycled when changing CRC4 and timing
settings for PRI. I agree with Critch _completely_; you must 'init 6' the
system when you make PRI changes otherwise you will obtain false results and
waste a lot of time.

If the comments above do not help perhaps you could provide a bit more
background information and then someone on the list will be able to assist.

HTH

Darren
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Paulo
Mannheimer
Sent: 31 May 2004 17:08
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Crc4 issues


Hi All,

This is our 2nd E1 client that we try to use crc4 either with the e100p
or with the e405p without luck.

After some trials, we ask the telco to switch off crc4 on their side and
everything works flawlessly.

Is there anything in the crc4 calculation that may be broken? We took a
look at wct1xxx.c and wct4xx.c but there doesn't seem to be much there
to be fixed (apparently the crc4 calculation is done within the chip
itself).

We also took a look at
http://lists.digium.com/pipermail/asterisk-cvs/2003-September/000126.htm
l but couldn't figure out what bits should we try to set to test other
card options.

Is there any documentation on the card that could help us?

Our zaptel looks like ...

span=1,0,0,ccs,hdb3,crc4

bchan=1-15,17-31
dchan=16

We already tried ...

span=1,1,0,ccs,hdb3,crc4
span=1,1,0,ccs,hdb3,crc4,yellow
span=1,0,0,ccs,hdb3,crc4,yellow


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RE: dialogic was RE: [Asterisk-Users] Glare condition - How well does asteriskhandle?

[Storer, Darren]

Hi Steve,

SU If you are using CTR4, then I guess they use CTR4. :-)
SU CTR4 ==  Net 5 == various other names == EuroISDN.

Reasonable logic but bad assumption in this case.

The Dialogic Q.931 stack (D/300, DM3 etc.) is solid and quite tolerant of
ISDN 85 as are most hardware PBXs. Other (PC based) products exhibit exactly
the same fussy behaviour though; the Digi RAS products (
http://tinyurl.com/36e7l ) work well with EuroISDN but won't work with ISDN
85 so the Asterisk stack is not alone in freaking when presented with this
Frankenstein Protocol of the ISDN world. (Thanks a bunch BT/Marconi/GPT et
al who rushed ISDN85 into service because they didn't want 18 months of
effort to delay real Q.931 deployment in the UK, so they bolted a protocol
converter on the end of existing DASS line cards instead of developing a
native solution...ugly stuff!)

I would like to try to help Tim decide which version of PRI he has as I'm
local to him, let's see if he takes me up on the offer to plug a working *
box into his PRI... Even if he has ISDN85 we would still benefit from the
chance to capture the failure (using an MPA) and compare it to some good
(working) * traces from a real EuroISDN circuit. Then the fun starts trying
to find a neat way to patch the stack...

Regards

Darren
--
Comgate
TelcoInternetBroadcast


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Underwood
Sent: 28 May 2004 01:39
To: [EMAIL PROTECTED]
Subject: Re: dialogic was RE: [Asterisk-Users] Glare condition - How
well does asteriskhandle?


tim panton wrote:

 Steve Underwood wrote:

 Jason Williams wrote:

 At 09:16 27/05/2004 -0500, you wrote:

 Maybe the time and effort would be better spent finding out why the
 Digium card won't work on the NTL's PRI and either fixing it or
 providing the information and testing facility to someone who can.





 NTL's PRI uses ISDN 85  not q931 so a ne protocol stack would need
 to be written.



 I think you means ISDN 85 not EuroISDN.

 Good heavens. I thought ISDN 85 died out in about 89. :-)

 I don't know where you would get the spec these days, but it
 shouldn't be a lot of work to modify libpri to add another variant of
 ISDN.


 I should say that I don't _know_ what NTL are delivering me,
 I haven't (yet) tried it with a digium E1 card.

 What I do know is :
 1) the dialogic card claims to be running CTR4 on an E1 ISDN PRI
 2) other folks on this list have had difficulty getting digium
 cards to talk to NTL.
 3) exactly the same dialogic config works on BT and the Dutch
 PTT's E1 lines.

 So it _may_ not be a problem for me as NTL is a patchwork of smaller
 telcos, my area (Manchester) may be more up to date.

 Anyone know an easy way to tell what I've got ?
 (or will I have to ask NTL -gh)

 T.

If you are using CTR4, then I guess they use CTR4. :-)

CTR4 ==  Net 5 == various other names == EuroISDN.

It sounds like you are OK.

Regards,
Steve

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RE: dialogic was RE: [Asterisk-Users] Glare condition - How well does asteriskhandle?

[Storer, Darren]

Hi Tim,

TP So it _may_ not be a problem for me as NTL is a patchwork
TP of smaller telcos, my area (Manchester) may be more up to
TP date.
TP Anyone know an easy way to tell what I've got ?
TP (or will I have to ask NTL -gh)

Pound to a penny you have ISDN 85. It's been reported via the list
recently that only one NTL region in the UK has ISDN 110 (EuroISDN).

If you are near Manchester and you're amenable, I'd like to ask if you'd
mind me coming down to capture a trace of Asterisk failing with NTL's ISDN
85? (Pretty please etc.) I have a portable(ish) Asterisk server, with PRI,
that I can bring along and the whole thing should take between 30 minutes
and 1 hour to setup. The test can take place any time early or late
(weekend's ok too) to suit you and the needs of your business.

It would be great to move the ISDN 85 problem forward; I've lost access to
the spare ISDN 85 circuit at a local switch site as it now has a production
server on it...

There are a number of features missing from ISDN 85 and some additional
Information Elements that are sent, especially during call setup and tear
down. I'm hoping that a patch to the existing Q.931 stack is all that's
required but without some hard facts to go on it will be difficult to crack.

Regards

Darren
--
Comgate
TelcoInternetBroadcast
Tel: +44(0)700 COMGATE

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of tim panton
Sent: 27 May 2004 20:59
To: [EMAIL PROTECTED]
Subject: Re: dialogic was RE: [Asterisk-Users] Glare condition - How
well does asteriskhandle?


Steve Underwood wrote:
 Jason Williams wrote:

 At 09:16 27/05/2004 -0500, you wrote:

 Maybe the time and effort would be better spent finding out why the
 Digium card won't work on the NTL's PRI and either fixing it or
 providing the information and testing facility to someone who can.




 NTL's PRI uses ISDN 85  not q931 so a ne protocol stack would need to
 be written.


 I think you means ISDN 85 not EuroISDN.

 Good heavens. I thought ISDN 85 died out in about 89. :-)

 I don't know where you would get the spec these days, but it shouldn't
 be a lot of work to modify libpri to add another variant of ISDN.


I should say that I don't _know_ what NTL are delivering me,
I haven't (yet) tried it with a digium E1 card.

What I do know is :
1) the dialogic card claims to be running CTR4 on an E1 ISDN PRI
2) other folks on this list have had difficulty getting digium cards to
talk to NTL.
3) exactly the same dialogic config works on BT and the Dutch PTT's E1
lines.

So it _may_ not be a problem for me as NTL is a patchwork of smaller
telcos, my area (Manchester) may be more up to date.

Anyone know an easy way to tell what I've got ?
(or will I have to ask NTL -gh)

T.

 Regards,
 Steve



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RE: [Asterisk-Users] Glare condition - How well does asteriskhandle?

[Storer, Darren]

Hi,

WKH You are correct... No glare on a PRI

Really? I followed some of the regular advice that's dispensed on this list
and tried to RTFG. Interestingly it transpires that several hundred hits on
Google seem to imply that you're both wrong:

http://tinyurl.com/2vmrh

..and here are two PRI/Glare scenarios nicely documented by Intel (for their
Linux stack before someone mentions M$):

http://tinyurl.com/27her

and

http://tinyurl.com/27her

Perhaps we are disagreeing over use of terminology rather than an event that
can obviously occur. I understand why  Scott raises the issue especially
with the aggressive services that he supports using Asterisk. Perhaps Scott
could use his call loop-back stress tester code to model the problem and let
us know how Asterisk behaves in a test environment. (Although he might need
two * machines, back to back, to recreate real circuit contention problems.)

Just my 2c

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of W. Kevin Hunt
Sent: 25 May 2004 23:27
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Glare condition - How well does
asteriskhandle?


You are correct... No glare on a PRI

W. Kevin Hunt

CCIE #11841
www.huntbrothers.com

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Tuesday, May 25, 2004 3:26 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Glare condition - How well does
asteriskhandle?

On Tue, 2004-05-25 at 13:53, Scott Stingel wrote:
 Hi-

 I have an upcoming application that requires use of PRI channels that
 are primarily used for high-volume incoming traffic, but that are to
 be used for outbound calling as well.  Of course, one option is to
 have dedicated outbound channels reserved, but this is an inefficient
 use of channel resources.

 Normally PBX's are designed to have the CPE yield to an incoming call
 if a particular channel is seized by both ends at the same time (a
 condition known as glare), but I'm wondering if anyone has
 real-world experience with asterisk to say how well this is handled.

While I may be wrong, I don't think glare happens on PRI. The
difference being that the call isn't sent over a channel until there had
been communications on the D channel. This means a send and a receive.
Glare would happen on a channelized T1 where it is possible for each
end to try and seize the channel at the same time, since there isn't any
out of band communications.
--
Steven Critchfield  [EMAIL PROTECTED]

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RE: [Asterisk-Users] Glare condition - How well does asterisk handle?

[Storer, Darren]

Hi Scott,

SS Normally PBX's are designed to have the CPE yield to an incoming
SS call if a particular channel is seized by both ends at the same
SS time (a condition known as glare), but I'm wondering if anyone
SS has real-world experience with asterisk to say how well this is
SS handled.

Earlier this year I had experience of this scenario in a UK carrier and in
the end I opted for split working with the top 10 channels of the PRI
dedicated for outbound traffic. Admittedly the platform may have been a
little under specified for the job (E100P installed in an HP 1U PIII 750MHz
Rack Server) but I was surprised that it could not cope with the contention
on a single E1. The symptoms were loss of channels and excessive alarms from
the public switch during peak traffic resulting in the need to reset the PRI
from the public switch console and a quick 'init 6' for the * machine.

The problem was captured on an MPA, I'm hoping to find a copy of the trace
to share with the list. From memory it appeared that * stopped responding to
some messages from the switch which left channels in an unknown state. The
CPU was occasionally busy so perhaps the poor signalling was related to CPU
load and timing constraints.

I'll post the trace when it surfaces...

HTH

Darren
--
Comgate
TelcoInternetBroadcast
Tel: +44(0)700 COMGATE

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel
Sent: 25 May 2004 19:54
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Glare condition - How well does asterisk
handle?


Hi-

I have an upcoming application that requires use of PRI channels that are
primarily used for high-volume incoming traffic, but that are to be used for
outbound calling as well.  Of course, one option is to have dedicated
outbound channels reserved, but this is an inefficient use of channel
resources.

Normally PBX's are designed to have the CPE yield to an incoming call if a
particular channel is seized by both ends at the same time (a condition
known as glare), but I'm wondering if anyone has real-world experience
with asterisk to say how well this is handled.

Thanks
Scott Stingel

Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California  London England
www.evtmedia.com


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RE: [Asterisk-Users] DASS2 support

[Storer, Darren]

Hi Peter,

PC Has anybody got Asterisk to work with DASS2 circuits?

As Steve said there is no native support for DASS2 within *. This leaves you
with a couple of choices:

 a) Ask for a new ETSI Q.931 ISDN circuit from a new Telco (no risk to
existing PBX). This can be at a very low cost if you go to a competitive
carrier who wants your business!

 b) Obtain a protocol converter (not cheap).

If you're interested in (b), follow the link below and take your pick (but
put your cheque book on steroids first):

http://tinyurl.com/2wzh8

HTH

Darren
PS. Please be aware that if you order a new circuit in the UK you should
specify ISDN30e to ensure you receive an ETSI compliant circuit that will
work with *.
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Underwood
Sent: 13 May 2004 13:27
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] DASS2 support


Peter Corlett wrote:

My employer wants to use Asterisk, but the E1 circuit providing the current
phone system is DASS2 rather than ISDN30. Converting the E1 to ISDN30 is
not
a practical proposition at this time as it'd stop the legacy phone system
from working.

Is there any sort of hardware support for DASS2? I speculate that the E100P
should be able to deal with the electrical side of it, but I'm unsure of
driver support.

Has anybody got Asterisk to work with DASS2 circuits?

Thanks in advance.


* has no software support for DASS or DASS2

Regards,
Steve

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RE: [Asterisk-Users] Calling CHRIS BARNET (PRI / E100P / ntl)

[Storer, Darren]

Hi,

DR Thanks for your Help Chris and thanks to Darren also...
DR Now where did I put that receipt for my e100p?

Whoa, relief *might* not be too far away... I'm installing some network
equipment on ISDN85 PRI circuits on Friday and whilst I'm at the switch site
I plan to capture a trace of the Asterisk/ISDN85 problem using an MPA
(portable SS7/PRI analyser). I already have the trace for a good ISDN110
(ETSI) connection and, in conjunction with the specs. from www.sinet.bt.com
, intend to spend some time early next week looking for a quick
fix/work-around.

I'm sure you'd like to make good use of those freebie PRI circuits with
something as flexible as Asterisk!?!

Watch this space...

Darren
PS. Telesoft have a nice protocol converter solution but it's really only
cost effective (with Asterisk) when converting multiple PRIs. They should be
e-mailing pricing over soon.
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Darren Round
Sent: 12 May 2004 15:26
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Calling CHRIS BARNET (PRI / E100P / ntl)


Hi Chris

Thanks for getting back to me :)

Sounds like your having the same sort of game I am with them.  Yeah I
tracked Darren down and called him myself (one helpful chap indeed!) and he
explained how the whole thing works.  Ive rang ntl back and got to speak to
one of the senior staff there who told me that the whole of the ntl network
is ISDN85 apart from one area (sorry guys I cant remember which one) which
is 110 and there is no information about their upgrade schedule.  So it
looks like for now until the new driver is written the E100P is not
compatible with the ntl in the UK :(

Believe me Id absolutely love to change Telco's but its not really
financially viable for me as I've got these pri lines rent free as part of a
bigger deal because we buy stacks of bandwidth from ntl.  Do you know the
irony is that there are talking about putting up the cost of my internet
bandwidth if I dont start using their outdated pri service!

Thanks for your Help Chris and thanks to Darren also... Now where did I put
that receipt for my e100p?

:)

Thanks guys

Darren Round

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Chris Barnett
 Sent: 12 May 2004 14:22
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Calling CHRIS BARNET (PRI / E100P / ntl)

 Hi,

 Following the *extremely* helpful assistance of Darren Storer
 I did establish that the NTL line I currently have is ISDN 85
 (partial ETSI) whereas Asterisk and the E100P want *full*
 ETSI (ISDN 110). Not that NTL even understood what it was I
 was trying to achieve and were initially very confused at the
 difference between ISDN 85 and ISDN 110, they just kept
 repeating that their line was Q.931 and therefore should be
 fine - certainly not the case! They got there in the end though :)

 I did hit a brick wall with NTL on upgrading the line though,
 the equipment delivering the line to my premises is not up to
 the job and needs upgrading/replacing by NTL, this has meant
 it's all had to go off to their finance department for
 approval etc. etc. Basically I've got mired up in NTL red
 tape :( You may be luckier in that the equipment used to
 deliver your line has been upgraded over the last few years
 to cope with ISDN 110, you'll need to speak with your NTL
 account manager to find this out.

 I've been speaking with Darren recently and there *may* be a
 chance that ISDN 85 support could be coded into the drivers
 for *, but even if this were to happen it's a way off. If NTL
 can't upgrade your current line to ISDN 110 and you can't
 wait for any possible ISDN 85 support in * then your only
 other option is to look at changing teleco. For me it's not
 all doom and gloom as the nice people at BT have been able to put a
 *very* attractive proposal together to persuade me to switch
 from NTL and provide me with a pucker ETSI line in the
 process - so hopefully all's well that ends well!

 Chris

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of
 Darren Round
 Sent: 12 May 2004 12:43
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Calling CHRIS BARNET (PRI / E100P / ntl)


 Chris you might know the answer to my HUUGE problem

 A few weeks ago you posted this message:
 I have an ISDN PRI supplied by NTL (ex Diamond Cable,
 Nottingham) which is currently working happily with an SDX
 Index phone system. I have to replace this phone system
 shortly and I've been trying to get a * system working for
 some weeks now. I have configured the dial plan (which
 works) and all my SIP extensions (which all work) along with
 voice mail etc. etc. - all this works perfectly as an
 internal PBX. My problem comes when I try to connect it to my
 ISDN line.

 Thread:
 http://lists.digium.com/pipermail/asterisk-users/2004-April/04
 4169.html

 I have exactly the 

RE: [Asterisk-Users] Signalling C7 / SS7

[Storer, Darren]

Hi Roger,

 What hardware do you use to connect your asterisk
 box to a PSTN carrier via C7/SS7 (instead of ISDN PRI)?

There is currently no native SS7 support within Asterisk. If your need is
urgent you could source a protocol converter (PRI/C7) from a 3rd party like
Telesoft (Okeford 4000 etc.) but the costs are prohibitive compared to a
very cost effective Asterisk platform.

There is an OpenSS7 project (See: http://www.openss7.org/asterix.html ) that
is looking at such a stack for Asterisk but it appears to have stalled. Have
a look in the mailing list archive for more info.

Out of interest, why do you need SS7 on your Asterisk build? (Which features
etc. are you looking to support?)

Regards

Darren
--
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Roger
Schreiter
Sent: 10 May 2004 13:48
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Signalling C7 / SS7


Hi,

has anybody out there experience with those
server grade connections?

What hardware do you use to connect your asterisk
box to a PSTN carrier via C7/SS7 (instead of ISDN PRI)?


Thanks for any hints!
Roger Schreiter.

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RE: [Asterisk-Users] No outbound calls at a PRI possible

[Storer, Darren]

Hi Felix,

on some UK public switches I have seen similar bad call setup problems with
a release cause of 28 (Invalid number format) when using:

  pridialplan=national

Have you tried:

  pridialplan=unknown

in zapata.conf?

It seems as though the omission of the pridialplan= statement in zapata.conf
is treated by Asterisk as pridialplan=national.

We could probably give you more relevant suggestions if you would enable a
more verbose level of output and post the call setup trace results here. Try
the following command from the Asterisk CLI before making your next call:

pri debug span x

Where x = single integer digit for the PRI span that will be used to make
the outgoing call. (Eg. 1)

Please drop a note to the list (either way) with your results.

HTH

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of ePyron Felix
Deierlein
Sent: 09 May 2004 20:32
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No outbound calls at a PRI possible


Hello,

i guess the problem ist pridialplan from zapata.conf

with

pridialplan = local

it works :-). But I still get the error messages:

 Calling Number (len= 4) [ Ext: 0  TON: Unknown Number Type (0)  NPI:
Unknown Number Plan (0)
   Presentation: Unknown (67) '' ]
 Called Number (len= 9) [ Ext: 1  TON: Subscriber Number (4)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]

What pridialplan should I use with an
E1 with Euroisdn from the German Telekom (DTAG or T-Com).


Thanks


Felix

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 ePyron Felix Deierlein
 Sent: Sunday, May 09, 2004 6:48 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] No outbound calls at a PRI possible

 Hello all,

 the scenario:

 Carrier S2M-- * -S2M--Siemens
   |
   |
   SIP Clients
   and many other features

 With much help from the list, the PRI links are without
 alarms and inbound calls are working fine (from both: Carrier
 and Siemens).

 But I am not able to dial wether outbound nor to the Siemens PBX.
 I allways get the message:
   == Everyone is busy at this time


 After hours of googling and reading and trying I seek help...

 Thank you very much.

 Felix Deierlein


 My extension.conf (only important parts):
 [AtInternal]
 ;exten = 402,1,Macro(stdexten,402,Zap/g2/402)
 exten = 402,1,Dial(Zap/g2/595402)

 [ePInternal]
 include=system
 include=test
 include=AtInternal

 exten = 812,1,Macro(stdexten,812,${ePFfd})
 exten = 814,1,Macro(stdexten,814,${ePFjw})
 exten = 854,1,Macro(stdexten,854,${ePFch})
 exten = 5950,1,Macro(stdexten,812,${ePFfd})
 exten = _0.,1,Dial(Zap/g1/${EXTEN:1},60)


 [zapata.conf]
 [channels]
 language=en
 context=default
 switchtype=euroisdn
 ;pridialplan=national
 usecallerid=yes
 hidecallerid=no
 callwaiting=yes
 usecallingpres=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 cancallforward=yes
 callreturn=yes
 echocancel=yes
 echocancelwhenbridged=yes
 rxgain=0.0
 txgain=0.0
 callgroup=1
 pickupgroup=1
 immediate=no

 ;pridialplan=national
 switchtype = euroisdn
 signalling = pri_cpe
 group = 1
 channel = 1-15
 channel = 17-31


 immediate=no

 switchtype = euroisdn
 signalling = pri_net
 group = 2
 callgroup=2
 pickupgroup=2
 channel = 32-46

 my zaptel.conf
 #amt (carrier)
 span=1,1,0,ccs,hdb3,crc4
 bchan=1-15
 dchan=16
 bchan=17-31
 #hicom (siemens)
 span=2,1,0,ccs,hdb3,crc4
 bchan=32-46
 dchan=47
 bchan=48-62
 loadzone=uk
 defaultzone=uk
 channel = 48-62


 PRI Debugging Infos:
 Call to Carrier: (Destination was 899312)
 -- Executing Dial(SIP/ePfd-b455, Zap/1/899312|60) in new stack
 -- Making new call for cr 32774
  Protocol Discriminator: Q.931 (8)  len=40 Call Ref: len= 2
 (reference
  6/0x6) (Originator) Message type: SETUP (5) Bearer
 Capability (len= 3)
  [ Ext: 1  Q.931 Std: 0  Info transfer
 capability: Speech (0)
   Ext: 1  Trans mode/rate: 64kbps,
  circuit-mode
 (16)
   Ext: 1  User information layer
 1: A-Law
  (35) Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,
  Exclusive
 Dchan: 0
 ChanSel: Reserved
Ext: 1  Coding: 0   Number Specified
  Channel Type:
 3
Ext: 1  Channel: 1 ] Display (len= 6)
 [ 1Felix ]
  Calling Number (len= 7) [ Ext: 0  TON: National Number (2)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
 number not screened (0) '812' ]
  Called Number (len= 9) [ Ext: 1  TON: National Number (2)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
  Sending Complete (len= 0)
 -- Called 1/899312
  Protocol Discriminator: Q.931 (8)  len=14  Call Ref: len=
 2 (reference 32774/0x8006) (Terminator)  Message type: STATUS 

RE: [Asterisk-Users] No outbound calls at a PRI possible

[Storer, Darren]

Hello Again Felix,

first a quick apology: sorry, I re-read your e-mail and found the trace
information (lower down) that you had already posted. (It's late here, etc.)

The error messages that you reported in your last e-mail are actually
outbound Q.931 call setup messages that are being sent to DTAG from your
Asterisk machine. The direction of the message is indicated in the first
column of the trace output in the form of  or . Although these are not
error messages I am surprised to see those particular messages being
generated with your current zapata.conf settings; with pridialplan=local I
would have expected something similar to the following messages during call
setup:

 Calling Number (len=14) [ Ext: 0  TON: Subscriber Number (4)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
   Presentation: Presentation permitted, user
number not screened (0) 'X58777' ]
 Called Number (len=14) [ Ext: 1  TON: Subscriber Number (4)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'X986600' ]

(I have inserted X in the PSTN numbers above to protect the innocent
Calling and Called parties.)

Please retry pridialplan=local and pridialplan=unknown in zapata.conf and
post the trace results so we compare results. With pridialplan=local in
zapata.conf the outbound call setup from Asterisk to DTAG should look ideal.

On a different subject, how are your results with telephony calls from the
Asterisk machine to your Hicom PBX? I would have expected the zaptel.conf
entry to have been:

 #hicom (siemens)
 span=2,0,0,ccs,hdb3,crc4

...so that your Asterisk provides clocking/timing information for the Hicom.
If this configuration is not set correctly you could find that the systems
seem to communicate well at first but after a while you might see strange
PRI errors (every hour or so) that relate to clock synchronisation problems.

MfG

Darren
--
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TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Storer,
Darren
Sent: 10 May 2004 01:29
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No outbound calls at a PRI possible


Hi Felix,

on some UK public switches I have seen similar bad call setup problems with
a release cause of 28 (Invalid number format) when using:

  pridialplan=national

Have you tried:

  pridialplan=unknown

in zapata.conf?

It seems as though the omission of the pridialplan= statement in zapata.conf
is treated by Asterisk as pridialplan=national.

We could probably give you more relevant suggestions if you would enable a
more verbose level of output and post the call setup trace results here. Try
the following command from the Asterisk CLI before making your next call:

pri debug span x

Where x = single integer digit for the PRI span that will be used to make
the outgoing call. (Eg. 1)

Please drop a note to the list (either way) with your results.

HTH

Darren
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TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of ePyron Felix
Deierlein
Sent: 09 May 2004 20:32
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No outbound calls at a PRI possible


Hello,

i guess the problem ist pridialplan from zapata.conf

with

pridialplan = local

it works :-). But I still get the error messages:

 Calling Number (len= 4) [ Ext: 0  TON: Unknown Number Type (0)  NPI:
Unknown Number Plan (0)
   Presentation: Unknown (67) '' ]
 Called Number (len= 9) [ Ext: 1  TON: Subscriber Number (4)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]

What pridialplan should I use with an
E1 with Euroisdn from the German Telekom (DTAG or T-Com).


Thanks


Felix

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 ePyron Felix Deierlein
 Sent: Sunday, May 09, 2004 6:48 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] No outbound calls at a PRI possible

 Hello all,

 the scenario:

 Carrier S2M-- * -S2M--Siemens
   |
   |
   SIP Clients
   and many other features

 With much help from the list, the PRI links are without
 alarms and inbound calls are working fine (from both: Carrier
 and Siemens).

 But I am not able to dial wether outbound nor to the Siemens PBX.
 I allways get the message:
   == Everyone is busy at this time


 After hours of googling and reading and trying I seek help...

 Thank you very much.

 Felix Deierlein


 My extension.conf (only important parts):
 [AtInternal]
 ;exten = 402,1,Macro(stdexten,402,Zap/g2/402)
 exten = 402,1,Dial(Zap/g2/595402)

 [ePInternal]
 include=system
 include=test
 include=AtInternal

 exten = 812,1,Macro(stdexten,812,${ePFfd})
 exten = 814,1,Macro(stdexten,814,${ePFjw})
 exten = 854,1,Macro(stdexten,854,${ePFch})
 exten = 5950,1,Macro(stdexten,812,${ePFfd})
 exten = _0.,1,Dial(Zap/g1/${EXTEN:1},60)


 [zapata.conf

RE: [Asterisk-Users] Fehler beim starten...[Translated]

[Storer, Darren]

[Literal translation from Google]

Hello,

after me up to now still nobody answered again my asks:

If I asterisk start get I the following error message:

 [app_capiCD.so]May  6 00:38:23 WARNING[16384]: loader.c:240
ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined
symbol: ast_capi_MessageNumber
May  6 00:38:23 WARNING[16384]: loader.c:408 load_modules: Loading
module app_capiCD.so failed!

I installed SUSE 9, a Fritzcard... for etc..  perhaps can help me someone
from you!

Thank you!

mfg

Markus Dohnal

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Administrator
Sent: 06 May 2004 17:12
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Fehler beim starten...


Hallo,

nachdem mir bis jetzt noch niemand geantwortet hat nochmal meine frage:

wenn ich asterisk starte bekomme ich folgende fehlermeldung:

 [app_capiCD.so]May  6 00:38:23 WARNING[16384]: loader.c:240
ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined
symbol: ast_capi_MessageNumber
May  6 00:38:23 WARNING[16384]: loader.c:408 load_modules: Loading
module app_capiCD.so failed!


Ich habe SUSE 9 installiert, eine Fritzcard... usw.


vielleicht kann mir ja jemand von euch helfen!

Vielen Dank!

mfg
Markus Dohnal
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RE: Caller ID Re: [Asterisk-Users] Re: Support Digium

[Storer, Darren]

Someone wrote:

 The BT CD50 and soldering iron plan is looking more and more like the
 one I'll be going with for now

If you don't fancy using a soldering iron to read UK CLI there's a mod to *
that my colleague, Robb Boardman, uses. By placing a certain model of Hayes
or Pace modem in parallel with * on the incoming PSTN line the CLI is
collected (before the first ring) via a serial TTY port. I'm sure it was
posted in here some while ago so if you're interested have a look in the
archives or reply to this note and see if we can encourage him to re-post
the details.

From memory the new ProSlic chip used by Digium supports UK CLI at a
physical interface level but appropriate drivers have not yet been coded.
Mark Spencer is very aware of the community's demand for international CLI;
I suspect that it's a case of ever growing demand for new functionality
verses finite implementation/support resources (both financial and human).
If we can obtain the ProSlic technical interface details does someone fancy
a spot of coding in return for a bounty...?

On the subject of line reversal detection I know of a major manufacturer
whose LLU products were recently rejected by a UK Telco for failure to
support this feature on V5 Access Network muxes. There were a number of
problems with automatic telephony equipment (E.g.. subscriber's own (CPE)
telephone answering machines) that could not detect the end of the call. One
of the strengths of the PSTN is the backward compatibility that has been
maintained (including physical standards like voltages as well as higher
protocols) for more than 100 years. I would like to echo an earlier poster's
comments about the necessity to maintain compatibility with the earlier
electro-mechanical standards for as long as we can.

Just my 2ds...

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark Elkins
Sent: 02 May 2004 23:21
To: [EMAIL PROTECTED]
Subject: RE: Caller ID Re: [Asterisk-Users] Re: Support Digium


On Mon, 2004-05-03 at 00:11, David J Carter wrote:
 Mark J Elkins wrote

 Um - Digium wants you to buy their hardware - but there is a CLID
 issue.. would it not make more financial sense to insert a dumb ISDN
 card (or two), and upgrade your PSTN to ISDN??? Would this not assist
 Digium in making sure CLID worked in the UK???

 Isn't this a bit like cutting of the nose to spite the face.

 UK PSTN lines costs £30 /Qtr  UK ISDN costs £65 /qtr, you could buy two
 X100P's every year and still be in pocket by staying with PSTN.

ISDN BRI is two lines - so that makes it £2.50 more per line  - or
£10 a year..?? no need to purchase the BT50 (a caller-ID unit? - at what
cost? you need one per line? and an RS232 interface per unit?)

 There was a post on the list in the not to distant past where someone had
 written two small scripts for getting the information from a BT50 and a
 serial modification and passing it to asterisk.

 Still seems the best way in the interim.

 As has been said many times in the list Digium have given us this
software,
 we don't have to give them a hard time in return. Not a fair payback.

True - the software is excellent. If they sold an ISDN BRI 4-port card
(like Fritz) - I'd buy it from them.
No intentions of bad mouthing Digium... but USA != World

--
  .  . ___. .__  Posix Systems - Sth Africa
 /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496


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RE: [Asterisk-Users] D/41 ESC dialogic ISA CARD

[Storer, Darren]

Hi Alejandro,

from memory only the newer JCT series of Dialogic cards are supported by
special drivers for Asterisk (obtained under license from Digium direct).
Please check if your D/41 card has a JCT suffix.

HTH

Darren
--
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TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Alejandro
Acosta
Sent: 22 April 2004 19:29
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] D/41 ESC dialogic ISA CARD


Hello,
   I just wanted to know if any of you has successfully (or know about)
installed the Dialogic 4xFXs ISA CARD D/41 ESC? Does it work with *?, if
so, how is the driver called?

Thanks a lot for your comments.

Alejandro Acosta,-
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RE: [Asterisk-Users] Fwd: Sasquatch, the Loch Ness Monster, UFOs and...Speech Recognition

[Storer, Darren]

Hi,

John Todd said:

 9) Speech recognition support

 Nothing towards this yet - sphinx keeps getting mentioned, though I
 don't know anyone who has had it running in anything other than a
 crippled test, or at least I don't remember anyone saying anything
 about it.

Which features do Asterisk users a) need and b) desire for a speech
recognition solution? Extensions to IVR and Auto Attendant applications are
the first couple that spring to mind but what else should/could be included?
Thoughts on size of vocabulary and API are of specific interest.

Thanks

Darren
--
Comgate
TelcoInternetBroadcast


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Todd
Sent: 08 April 2004 15:01
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Fwd: Sasquatch, the Loch Ness Monster, UFOs
and...


Every half year or so, I probably will repost this list, adding and
subtracting as the community makes advances (or ignores what isn't
required.)


Date: Thu, 9 Oct 2003 04:51:23 -0400
To: asterisk-users-lists.digium.com
From: John Todd [EMAIL PROTECTED]
Subject: Sasquatch, the Loch Ness Monster, UFOs and...

Mythical Asterisk Creatures, oft-discussed, rarely seen:

1) An advanced graphical user interface

We're getting there.  There are starting to appear a crop of PHP or
in at least one case, Flash-driven front ends for users.  These
haven't been compiled as part of asterisk-addons, but perhaps
sometime in the next month or two the code from the existing various
projects can be pushed into the addons directory.

2) An IAX2 hardware device

Any Day Now(tm).  Wasim has fallen off the face of the Earth, but
I've seen with my own two eyes a working copy of the Iaxy from
Digium, so this holds promise.  My request for a 1u 24-port IAX-based
box that takes Digium daughterboards (FXO or FXS) generated some
interest when a show of hands was asked for at the VON show... Bob
Knight seemed to have an interest and some time on his hands.  ;-)

3) A Radius CDR report module

This sort-of exists now, but again is not a completely robust
solution.  I've not implemented it yet (due to other pressing issues
of life and profit) but it should hopefully work with some of the
traditional billing systems that existing VoIP carriers are using.

4) A live-method, robust SQL-based dialplan

Not sure on this one - anyone care to comment?

5) LDAP/SQL/Radius authentication for SIP phones

I hear rumors of this existing, but again, I haven't had the time to
investigate.  The SQL-friends database hacks might be the answer for
an SQL system.

6) Robust R2 signalling support

Steve Underwood says that he's made advances... has anyone else done
any work on R2?

7) Multilingual language recordings of all existing * .gsm files

Nothing that I know of towards this end, or at least, nothing that is
available on the CVS server.  Anyone?

8) Free exchange of PSTN gateways in a centralized routing arbiter model

HO ho ho ho ho... that's a funny one.  Actually, I have someone
working on TRIP now, but I suspect that budget will get cut as soon
as another project starts to explode.

9) Speech recognition support

Nothing towards this yet - sphinx keeps getting mentioned, though I
don't know anyone who has had it running in anything other than a
crippled test, or at least I don't remember anyone saying anything
about it.


Here are this halfyear's additions:

10) Encryption

I'd love to see TLS/SRTP built into the SIP stack, to support the
Zultys and Sipura devices which now handle crypto natively.  More
clients will support this functionality; time to start building
Asterisk to work with them.  Additionally, IAX2 would be much cooler
if it had a full-channel encryption method, which I know is at least
being thought about (the aes header files have appeared in the CVS
distro.)

11) Presence.

Support for presence integration into devices would be great, and is
this year's hot-button technology.  Just simply supporting line
appearances would help out quite a bit for business users on newer
devices which support that feature, but the same technology
(subscribe/notify) could be used for more advanced presence features.
My ideas about integration into existing chat services might have
some merit, or maybe not.

12) BSD Support

We've got Asterisk compiling, now to get Zaptel/libpri working with
Digium cards...  rumors have someone Almost Done(tm)

13) High-density Zap cards

Inexpensive DS3 Zap-driven cards would be a boon for large providers.
The cards exist, there are Linux drivers, all that is required is
some GPL'ed glue code and hair-pulling to weave it into
Zaptel/libpri.  With the data mode on Asterisk, it might also be
possible to provide the equivalent of a Cisco CT3+ card that does
voice as well.


That's all I can think of at the moment.  Comments are welcome.

JT
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RE: [Asterisk-Users] B-channels resetting every 60 minutes?

[Storer, Darren]

Hi Lach,

this looks like normal behaviour to me. Most of the equipment I use issues a
restart upon initial physical connection (bad equipment can cause problems
when it doesn't do this) and then several times per hour thereafter. Once
every hour seems infrequent but I guess that this is down to individual
suppliers' interpretation of the specification documents.

HTH

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of osx
Sent: 07 April 2004 17:47
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] B-channels resetting every 60 minutes?


Hello,

As you can see are pri is being reset every 60 minutes!  Is there a way to
stop this??  Is it a Zapata configuration problem?

We have a * box with a single port T1/pri card installed.

Thanks

lach

Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 2 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 3 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 4 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 5 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 6 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 7 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 8 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 9 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 10 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 11 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 12 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 13 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 14 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 15 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 16 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 17 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 18 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 19 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 20 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 21 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 22 successfully restarted
on span 1
Apr  7 09:00:07 VERBOSE[114696]: -- B-channel 23 successfully restarted
on span 1
Apr  7 10:00:08 VERBOSE[114696]: -- B-channel 2 successfully restarted
on span 1
Apr  7 10:00:08 VERBOSE[114696]: -- B-channel 3 successfully restarted
on span 1
Apr  7 10:00:08 VERBOSE[114696]: -- B-channel 4 successfully restarted
on span 1
Apr  7 10:00:08 VERBOSE[114696]: -- B-channel 5 successfully restarted
on span 1
Apr  7 10:00:08 VERBOSE[114696]: -- B-channel 6 successfully restarted
on span 1
Apr  7 10:00:08 VERBOSE[114696]: -- B-channel 7 successfully restarted
on span 1
Apr  7 10:00:08 VERBOSE[114696]: -- B-channel 8 successfully restarted
on span 1
Apr  7 10:00:08 VERBOSE[114696]: -- B-channel 9 successfully restarted
on span 1
Apr  7 10:00:08 VERBOSE[114696]: -- B-channel 10 successfully restarted
on span 1
Apr  7 10:00:08 VERBOSE[114696]: -- B-channel 11 successfully restarted
on span 1
Apr  7 10:00:09 VERBOSE[114696]: -- B-channel 12 successfully restarted
on span 1
Apr  7 10:00:09 VERBOSE[114696]: -- B-channel 13 successfully restarted
on span 1
Apr  7 10:00:09 VERBOSE[114696]: -- B-channel 14 successfully restarted
on span 1
Apr  7 10:00:09 VERBOSE[114696]: -- B-channel 15 successfully restarted
on span 1
Apr  7 10:00:09 VERBOSE[114696]: -- B-channel 16 successfully restarted
on span 1
Apr  7 10:00:09 VERBOSE[114696]: -- B-channel 17 successfully restarted
on span 1
Apr  7 10:00:09 VERBOSE[114696]: -- B-channel 18 successfully restarted
on span 1
Apr  7 10:00:09 VERBOSE[114696]: -- B-channel 19 successfully restarted
on span 1
Apr  7 10:00:09 VERBOSE[114696]: -- B-channel 20 successfully restarted
on span 1
Apr  7 10:00:09 VERBOSE[114696]: -- B-channel 21 successfully restarted
on span 1
Apr  7 10:00:09 VERBOSE[114696]: -- B-channel 22 successfully restarted
on span 1
Apr  7 10:00:09 VERBOSE[114696]: -- B-channel 23 successfully restarted
on span 1
Apr  7 11:00:10 VERBOSE[114696]: -- B-channel 1 successfully restarted
on span 1
Apr  7 11:00:10 VERBOSE[114696]: -- B-channel 2 successfully restarted
on span 1
Apr  7 11:00:10 VERBOSE[114696]: -- B-channel 3 successfully restarted
on span 1
Apr  7 11:00:10 VERBOSE[114696]: -- B-channel 4 successfully restarted
on span 1
Apr  7 11:00:10 VERBOSE[114696]: -- B-channel 5 

RE: [Asterisk-Users] Siemens EWSD 13

[Storer, Darren]

Hi,

I had exactly the same symptoms today with a co-located * connected to a
Public Switch here in the UK. The problem was solved by insisting that the
Telco turned on CRC4 at their end and then, after an 'init 6', layer two
settled down on both systems.

I was taught that if you are connecting to a full specification Q.931
circuit, CRC4 should be enabled by default; in the event that one end does
not support CRC4 the other end should auto-negotiate back and the circuit
should still align without problems. Having said all of this I have yet to
see auto-negotiation of CRC4 on any equipment (Public Network or CPE) and
suspect that I was not told the truth in the first place...

Selection of CRC4 seems to be random from Telco to Telco even on an install
by install basis within the same Carrier. It's the first thing to check when
new kit appears to be unstable..

HTH

Darren
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: 07 April 2004 14:59
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Siemens EWSD 13


Hi all,

Has anyone got any experience with hooking Asterisk up with a
Siemens EWSD 13 switch over a E1/PRI ?
We're located in Belgium (Europe) and one of our telecom partners
uses this switch.

We connected one of our TE410P ports with their switch, but the status
light on the TE410P card keeps blinking red.
On their side they are getting a DSA (distance service alarm) error, so
this normally means the devices 'see' eachother.. but there are still
problems with the signalling.

Our config below is the same as we are using for MCI, one of our other
telecom partners.

We tried changing the LBO and timing, but no luck.
As you see the signalling is carried over channel 16 (default).

TX and RX have also been regularly switched, so no luck..

Their switch is providing the timing.

The telecom operator has double checked the asterisk config several
times, and it's conform to their setup.

The only thing they couldn't find in the Asterisk config is a
'multiframing' option. But I presume this is automatically detected or
set by default ?
They also tried normal/single(?) framing, but no difference.

The card has also been tested with our MCI E1, and works flawlessly, so
no hardware issue.

Anyone got any further ideas ?

Any info or help greatly appreciated!

Our config,

*** zaptel.conf ***
span=1,1,6,ccs,hdb3,crc4,yellow
bchan=1-15
bchan=17-31
dchan=16

*** zapata.conf ***
[channels]
switchtype=euroisdn
signalling=pri_cpe
pridialplan=unknown

group=1
channel = 1-15,17-31

other zapata standard config



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RE: [Asterisk-Users] res_motv: Request for Comment

[Storer, Darren]

YES PLEASE.

Wonderful Stuff! In my opinion just what the project needs. I deployed and
supported many GPL and commercial SmoothWall (firewall) installs and was
forced to poll a web page from time to time to see if any of my customers
needed an urgent security patch applying...not a satisfactory way to manage
many machines deployed across several countries.

The usual caveats about reviewing the 'phone home source code apply of
course as does an opt out for certain Carriers/official organisations that
prefer to remain anonymous.

Regards

Darren
--
Comgate
TelcoInternetBroadcast

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark Spencer
Sent: 07 April 2004 04:31
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] res_motv: Request for Comment


I've been considering the nature of Asterisk, its security, the bug
tracker, and more...  And i've come up with an interesting idea: A
message of the version.  The idea is that Asterisk has a compile time
32-bit unsigned int version which is incremented whenever some major new
bug is fixed.  When Asterisk starts up (and periodically, maybe once per
day), it sends a packet with the version number to a server at Digium,
along with a message level (INFO,MINOR,MAJOR,CRITICAL) and the Digium
server replies (if it receives the packet, if not, it might get sent again
in a day) with any INFO, MINOR, MAJOR, or CRITICAL messages which are
associated with that version of the code.  In this way, an asterisk
administrator could easily see if there were any major issues, critical
security updates, etc, that his system might need to be updated for.

Now, of course, any time you put a call home feature in, there are
people who will be concerned about privacy.  Clearly it will be able to be
disabled, but I want to run my idea about deployment by everyone here and
see if you guys had some ideas.  The idea would be that *new* installs
(make samples) would have the feature turned on for MAJOR level by
default, and that any existing install (e.g. /etc/asterisk/sip.conf
exists, but not /etc/asterisk/motv.conf) would have the file created at
the next make install based upon prompting the installer.

Any feedback on:

a) The idea itself -- is it a good one or is it stupid?

b) The way to make it deployed without sneaking a call home in on
anybody that doesn't want it?

Thanks!

Mark

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RE: [Asterisk-Users] PRI issues with TE410P

[Storer, Darren]

Hi Azher,

 They advertised on TeleVision and then we had a rush of
 calls landing on the system (about 30 calls in 1-2 secs and 60 in 3-4
 seconds time).

this sounds like a high number of _simultaneous_ call attempts for a PRI
connected system to support. Your comments about a gradual load increase
presenting no problem would seem to back this fact up. On most Public
Telephone Switches (in the UK) it is advised that no more than 5 call
attempts per second should be attempted via PRI if you seek stability.
(Above 5 calls attempts per second Network Congestion messages can be
sometimes be observed on a protocol analyser. These congestion messages can
prove fatal for certain applications such as high quality video conferences
via hardware that aggregates multiple B channels.) It should be remembered
that PRI connections are based upon subscriber signalling specifications and
that spiky mass call events are traditionally hosted on platforms that are
connected via SS7 signalling.

I know of similar stability problems with Dialogic and Acculab ISDN
connected systems where mass call events, in excess of 5 call attempts per
second, have proved too much of a spike for the Public Switch. In certain
switch architectures concentrator equipment can also be adversely affected
by this kind of deployment which can cause disruption for many customers
(not just the one running the PRI IVR platform).

On slower CPU systems ( 800 MHz) I have noticed occasional Euro ISDN PRI
signalling problems if a high percentage of the 30 channels on an E100P are
busy with simultaneous outbound/inbound calls and glare occurs. On the
protocol analyser trace it was evident that some messages via the D channel
from the Public Switch went unacknowledged by the Asterisk system. If you
encounter this problem a quick work around is to split the channels of your
single PRI into two trunks to stop the contention between inbound and
outbound telephony traffic. (E.g. g1 for incoming and g2 for outgoing calls)

I have yet to flex a TE410P with significant traffic levels as all my busy
systems are using E100P cards but I have just finished commissioning a
TE410P system (with help from Rob Boardman - thanks) in Munich that will be
stress tested over the next two weeks; I hope to have news on stability when
the tests are completed.

HTH

Darren
--
Comgate (UK)
TelcoInternetBroadcast

To:  [EMAIL PROTECTED]
Subject:  RE: [Asterisk-Users] PRI issues with TE410P
From:  Azher Amin [EMAIL PROTECTED]
Date:  Thu, 25 Mar 2004 09:09:18 +0500

Hi Tan,

I have already turned off the HT from the BIOS (while it was enabled,
asterisk was not even stable with 2 pri's).

Further I am using the same kernel (2.4.25) with SMP enabled, that u
mentioned in your email.

Basically I think Scott is right in his emails, that libpri's and zaptel
driver for t4xxp are not really optimized and they really need tunings
in case when there is certain increase in number of calls   Also I
cannot afford to get a system higher than the current specs.

Yesterday I had the same situation, I gave the IVR solution using te410p
to a company. They advertised on TeleVision and then we had a rush of
calls landing on the system (about 30 calls in 1-2 secs and 60 in 3-4
seconds time). What I have observed is:

1. Asterisk is stable upto 2 PRI's ... no call drops no errors.
2. As soon as I plug in the 3rd pri (3rd pri is confirmed stable), it
starts dropping the calls current landing as well as the previous stable
landed calls. All calls get dropped and PRIs' start syncing / unsyncing.
3. On the other hand I have also a dialogic server with 4 pri's in it,
and it was a rock solid ... going to full 120 calls ... btw: server is
just P3 700Mhz, and I have not noticed a single dropped call.
4. My configurations were checked by Mike (@ digium) ... and according
to him all things are fine, zaptel,zapata,extensions.conf ...

Thus when there is a smooth and gradual load of upto 3 pri's system will
remain stable, and when there is an instant call load it becomes shaky.

Also I will request those who have successfully done similar setup, to
post/advise their linux flavor (and updates), asterisk configurations,
kernel configuration, other system tweakings.

I want Mark from Digium to take special interest in solving and
recommendation on these issues, coz I have confirmed orders of 3 te410p
boards and one e100p (for which I mailed to Greg as well), but now I am
feeling reluctant to face the client.

Regards
Azher Amin
---
http://www.consulttech.com.pk


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, March 24, 2004 5:03 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] PRI issues with TE410P

Before going to gentoo or another version of linux, try the following:

1) turn hyperthreading off in the BIOS. It's probably called something
like virtual cpus.

2) Use a vanilla kernel from kernel.org e.g.