[asterisk-users] cisco iad's

[Michael Di Martino]

To Members,

I currently looking for someone who can configure Cisco IAD's Ie.. IAD 2431
And yes we are more than willing to pay for the service.
If interested please drop me an email 
m...@openaccessinc.com


Michael DiMartino | Director of IT | Open Access, Inc.
115 Bi County Blvd | Farmingdale, NY 11735
631.227.1034| 631.694.6730 FAX |631.988.6060 MOBILE
www.openaccessinc.com

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[asterisk-users] Cisco IAD's

[Michael Di Martino]

To Members,

I currently looking for someone who can configure Cisco IAD's Ie.. IAD 2431
And yes we are more than willing to pay for the service.
If interested please drop me an email 
m...@openaccessinc.com



Michael DiMartino | Director of IT | Open Access, Inc.
115 Bi County Blvd | Farmingdale, NY 11735
631.227.1034| 631.694.6730 FAX |631.988.6060 MOBILE
www.openaccessinc.com

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RE: [Asterisk-Users] Asterisk and Norstar MICS

[Michael Di Martino]

That works thanks! 

What is option 11? 


Michael

Michael DiMartino 
Director of MIS 
The telx Group, Inc. 
 telx
17 State St, 33rd Floor 
New York, NY 10004 
p 212.480.3300 X2022 
m 646.207.6603 
 Connectivity Accelerated

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gleim,
Jason
Sent: Friday, July 22, 2005 12:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk and Norstar MICS

I *believe* you can append '#' on the end of the dial string to tell
Nortel you are done dialing. I know it works on the Option 11.

Hope that helps!

Jason

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Thursday, July 21, 2005 10:38 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk and Norstar MICS

On Friday 22 July 2005 10:15, Michael Di Martino wrote:
> My current issues is a 5 second delay for call that is being
transferred
> from the Norstar units to
> the Asterisk servers VIA a PRI. Is their anything that can be done to 
> speed up the transfer on the Norstar.  Below  is my current phone 
> config.

You need to tell the norstar that you are done dialing.  It's waiting
for more digits.  Routing Service, Public DN Lengths and adjust the
correct prefix.

-A.
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[Asterisk-Users] Asterisk and Norstar MICS

[Michael Di Martino]

 
To All;
My current issues is a 5 second delay for call that is being transferred
from the Norstar units to
the Asterisk servers VIA a PRI. Is their anything that can be done to
speed up the transfer on the Norstar.  Below  is my current phone
config.

< Norstar1 >PRI< Asterisk-1 >IP-WAN< Asterisk-2
>---PRI---< Norstar2>

The Norstars are MICS 0x32 4.1 software


Thanks in advance
Mike

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RE: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread

[Michael Di Martino]

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Monday, June 27, 2005 5:26 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread

On Monday 27 June 2005 15:46, steve szmidt wrote:
> One could probably argue effectively for an Asterisk-Basic list. Or an

> Asterisk-Advanced user list. Something that makes it easier to get 
> started without being overwhelmed by 10,000-15,000 users posts. A 
> place that frequently posted links to the beginner pages on the wiki.

We've effectively argued it to death many many times over the course of
the last few years.  Check the archives -- it's been thought up and
re-thought up and dismissed each and every time.

Basic issue: nobody will want to sit on the newbie list because they'll
end up answering all the same questions over and over since nobody
really seems to want to read for themselves.

It's the same argument that comes around for forums, except that last
time I think we actually witnessed a man lose his mind on the mailing
list.  That was entertaining.  :-)


~

Their would not be so many newbie questions if their was 1. A fully
indexed searchable archive list and 2. Good solid documentation.





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RE: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread

[Michael Di Martino]

I agree with that fact the same questions get posted, but that problem
is compounded by the fact the archives are not really searchable. If the
were as lease some users would search.
The archives need to be fully indexed.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of steve
szmidt
Sent: Monday, June 27, 2005 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread

On Monday 27 June 2005 14:31, Michael Di Martino wrote:
> If this list spent at least half the time on helping other asterisk 
> admins as it does on trivial things like LiveVoips bankruptcy it just 
> might be a great list.
> As it stands now this list is kind of useless.  Most request for 
> assistance with asterisk problems go unresolved of unanswered.

Lists with this number of new members have a repetiveness of the same
questions which people sometimes get tired of answering. Which is too
bad.

However, even though it seemingly does not directly aid asterisk users
it does so indirectly. People on this list grow into becoming lemonade
stand operators and maybe even bigger service providers.

It is often done because people realize that "Wow, I can do it too!". 
Unfortunately it's not something that lives in the world of the
Internet, but enters the heavily controlled area of phones. An area
droght with difficulties for any newcomer. 

The thread is showing and giving reason to be a bit better prepared when
entering into this particular service industry. As such it is of great
importance to those wise enough to take note. For the rest it's just
noise.

One could probably argue effectively for an Asterisk-Basic list. Or an
Asterisk-Advanced user list. Something that makes it easier to get
started without being overwhelmed by 10,000-15,000 users posts. A place
that frequently posted links to the beginner pages on the wiki.

-- 

Steve Szmidt

"They that would give up essential liberty for temporary safety deserve
neither liberty nor safety."
Benjamin Franklin
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RE: [Asterisk-Users] LiveVoip is Bankrupt

[Michael Di Martino]

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Monday, June 27, 2005 3:27 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt

On Monday 27 June 2005 14:31, Michael Di Martino wrote:
> If this list spent at least half the time on helping other asterisk 
> admins as it does on trivial things like LiveVoips bankruptcy it just 
> might be a great list.
> As it stands now this list is kind of useless.  Most request for 
> assistance with asterisk problems go unresolved of unanswered.

Do you have some proof of this?  I find the list rather helpful on the
whole, with interjections of other (sometimes very OT) subjects
inbetween.

-A.
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This list has potential but it is not strong enough dealing w/ asterisk
system issues.


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RE: [Asterisk-Users] LiveVoip is Bankrupt

[Michael Di Martino]

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
http://www.0xdecafbad.com
Sent: Monday, June 27, 2005 4:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] LiveVoip is Bankrupt

On Mon, 2005-06-27 at 14:31 -0400, Michael Di Martino wrote:
> If this list spent at least half the time on helping other asterisk 
> admins as it does on trivial things like LiveVoips bankruptcy it just 
> might be a great list.
> As it stands now this list is kind of useless.  Most request for 
> assistance with asterisk problems go unresolved of unanswered.
> 
> If you would like to see how a good list is run join the Qmail users 
> list and observe.
>  
The bankrupt thread is mostly now about finiding hosting for Daily
Asterisk News, which I feel is helping asterisk people, and people
whining about this thread.  

The whining seemed to be from people reading the subject line and not
even bothering to notice that the majority of the posts under this
subject were about an asterisk specific thing when I saw that.  This
isnt slashdot we should actually read more than the subjects before
commenting.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378

8
Not really. This thread does not belong on this list. It is off topic
and a waste of time for admins dealing w/  real system issues. Like I
said sign up for the qmail list and you will see how a real user list
operates.




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RE: [Asterisk-Users] IAXY setup

[Michael Di Martino]

 No problems provisioning the iaxy device. and I have two other devices
working internally.

The output form debug shows that the iaxy device is communicating w/ the
the asterisk server but 
it never registers.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan & Company, LLC
Sent: Monday, June 27, 2005 2:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAXY setup

do your firewall rules allow the proper port in?  and you had no
problems in provisioning the iaxy?  Have you tried it from the same
subnet as *?

Michael Di Martino wrote:
>  
> 
> I am attempting to get an iaxy device to connect to my asterisk box 
> over the public cloud however
> 
> It fails register and I cannot figure out why.
> 
>  Below is my iax.conf, iaxy setup file and debug output  from iax2
debug.
> 
>  
> 
> My iax.conf
> 
> [u7402]
> 
> type=friend
> 
> accountcode=iaxy
> 
> host=dynamic
> 
> secret=u7402p
> 
> context=from-iaxy
> 
> disallow=all
> 
> allow=ulaw
> 
> callerid="my iaxy" <7402>
> 
> trunk=no
> 
> notify=yes
> 
>  
> 
> Mi iaxy setup file
> 
> [EMAIL PROTECTED] iaxyprov]# cat iaxy.conf.7402
> 
> ; IAXY Provisioning description
> 
> ;
> 
> dhcp
> 
> ;ip: 216.207.244.130
> 
> ;netmask: 255.255.255.192
> 
> ;gateway: 216.207.244.129
> 
> codec: ulaw
> 
> ;codec: adpcm
> 
> server: 207.251.84.198
> 
> ;altserver: 192.168.0.2
> 
> user: u7402
> 
> pass: u7402p
> 
> register
> 
> ;heartbeat
> 
> ;debug
> 
> ;
> 
> ; Feature tuning (default is all enabled)
> 
> ;
> 
> ;disablecid
> 
> ;disablecw
> 
> ;disablecidcw
> 
> ;disable3way
> 
> [EMAIL PROTECTED] iaxyprov]#
> 
>  
> 
> Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:

> ACK   
> 
>Timestamp: 2ms  SCall: 1  DCall: 10605 [24.47.87.47:4569]
> 
> Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:

> REGREQ
> 
>Timestamp: 2ms  SCall: 10605  DCall: 0 [24.47.87.47:4569]
> 
>USERNAME: u7402
> 
>REFRESH : 60
> 
>DEVICE TYPE : iaxy2
> 
>SERVICE IDENT   : 000364000132
> 
>PROVISIONG VER  : 326528057
> 
>  
> 
>  
> 
> Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:

> ACK   
> 
>Timestamp: 2ms  SCall: 1  DCall: 10605 [24.47.87.47:4569]
> 
>  
> 
>  
> 
> 
> --
> --
> 
> ___
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> --
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> 
> No virus found in this incoming message.
> Checked by AVG Anti-Virus.
> Version: 7.0.323 / Virus Database: 267.8.1/28 - Release Date: 
> 6/24/2005
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RE: [Asterisk-Users] LiveVoip is Bankrupt

[Michael Di Martino]

If this list spent at least half the time on helping other asterisk
admins as it does on
trivial things like LiveVoips bankruptcy it just might be a great list.
As it stands now this list is kind of useless.  Most request for
assistance with asterisk problems go unresolved of unanswered.

If you would like to see how a good list is run join the Qmail users
list and observe.
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
http://www.0xdecafbad.com
Sent: Monday, June 27, 2005 2:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt

On Mon, 2005-06-27 at 07:04 -0400, Andrew Kohlsmith wrote:
> On Monday 27 June 2005 02:45, Marcel van Kaam, Fonetica wrote:
> > I think by now everybody knows that LiveVoip went down, bankrupt
etc
> > So please stop nagging about it and move on to some topics that 
> > really matter.
> >
> > If you want to discuss LiveVoip, get all together in a restaurant, 
> > eat, drink and nag and wine about it as much as you want. But do it 
> > there and not here.
> 
> I've never understood this -- people are having a decent discussion.  
> There's no flaming, there's no bashing.  Sure it's offtopic but it'll 
> die within a few more days...  Why snuff it?  I am positive we're all 
> not geographically close to discuss this in a restaurant, and setting 
> up an entirely new list is silly.
> 
> So I ask you -- what should people do?
> 
> -A.

I think he was upset that people were talking about the Daily Asterisk
News website, and people were offering to donate webspace to keep it up
and stuff and all of that happened under the "LiveVoip is Bankrupt"
subject line.  

But then I could be wrong, maybe he did not actually read anything but
the subject itself and decided to attack people to force this very
conversation, about him doing exactly what he is claiming others are
doing.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] IAXY setup

[Michael Di Martino]

 


I am attempting to get an iaxy 
device to connect to my asterisk box over the public cloud 
however
It fails register and I cannot 
figure out why.
 Below is my iax.conf, iaxy 
setup file and debug output  from iax2 debug.
 
My 
iax.conf
[u7402]
type=friend
accountcode=iaxy
host=dynamic
secret=u7402p
context=from-iaxy
disallow=all
allow=ulaw
callerid="my iaxy" 
<7402>
trunk=no
notify=yes
 
Mi iaxy setup 
file
[EMAIL PROTECTED] iaxyprov]# cat 
iaxy.conf.7402
; IAXY Provisioning 
description
;
dhcp
;ip: 
216.207.244.130
;netmask: 
255.255.255.192
;gateway: 
216.207.244.129
codec: 
ulaw
;codec: 
adpcm
server: 207.251.84.198 

;altserver: 
192.168.0.2
user: u7402 

pass: u7402p 

register
;heartbeat
;debug
;
; Feature tuning (default is all 
enabled)
;
;disablecid
;disablecw
;disablecidcw
;disable3way
[EMAIL PROTECTED] 
iaxyprov]#
 
Tx-Frame Retry[-01] -- OSeqno: 000 
ISeqno: 001 Type: IAX Subclass: ACK    

   Timestamp: 
2ms  SCall: 1  DCall: 10605 
[24.47.87.47:4569]
Rx-Frame Retry[Yes] -- OSeqno: 000 
ISeqno: 000 Type: IAX Subclass: REGREQ 

   Timestamp: 
2ms  SCall: 10605  DCall: 0 
[24.47.87.47:4569]
   
USERNAME    : 
u7402
   
REFRESH : 
60
   DEVICE 
TYPE : iaxy2
   SERVICE 
IDENT   : 000364000132
   PROVISIONG VER  : 
326528057     

 
 
Tx-Frame Retry[-01] -- OSeqno: 000 
ISeqno: 001 Type: IAX Subclass: ACK    

   Timestamp: 
2ms  SCall: 1  DCall: 10605 
[24.47.87.47:4569]
 
 
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[Asterisk-Users] iaxy over the public cloud

[Michael Di Martino]

I am attempting to get an iaxy device to connect to my
asterisk box over the public cloud however

It fails register and I cannot figure out why.

 Below is my iax.conf, iaxy setup file and debug output
 from iax2 debug.

 

My iax.conf

[u7402]

type=friend

accountcode=iaxy

host=dynamic

secret=u7402p

context=from-iaxy

disallow=all

allow=ulaw

callerid="my iaxy" <7402>

trunk=no

notify=yes

 

Mi iaxy setup file

[EMAIL PROTECTED] iaxyprov]# cat iaxy.conf.7402

; IAXY Provisioning description

;

dhcp

;ip: 216.207.244.130

;netmask: 255.255.255.192

;gateway: 216.207.244.129

codec: ulaw

;codec: adpcm

server: 207.251.84.198 

;altserver: 192.168.0.2

user: u7402 

pass: u7402p 

register

;heartbeat

;debug

;

; Feature tuning (default is all enabled)

;

;disablecid

;disablecw

;disablecidcw

;disable3way

[EMAIL PROTECTED] iaxyprov]#

 

Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type:
IAX Subclass: ACK    

   Timestamp: 2ms  SCall: 1 
DCall: 10605 [24.47.87.47:4569]

Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type:
IAX Subclass: REGREQ 

   Timestamp: 2ms  SCall: 10605 
DCall: 0 [24.47.87.47:4569]

  
USERNAME    : u7402

  
REFRESH : 60

   DEVICE TYPE : iaxy2

   SERVICE IDENT   : 000364000132

   PROVISIONG VER  : 326528057
    

 

 

Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type:
IAX Subclass: ACK    

   Timestamp: 2ms  SCall: 1 
DCall: 10605 [24.47.87.47:4569]

 

 






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RE: [Asterisk-Users] LiveVoip is Bankrupt

[Michael Di Martino]

Hey pooch are u ever going to put up the howto's from the Atlanta
asterisk conference? You only said you would. Don't be like LiveVOIP and
follow thru on your word.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Capouch
Sent: Sunday, June 26, 2005 4:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt

Yair Hakak wrote:
> well, i can't say i'm surprised. any company whose approach to
> customers is "you are all scum trying to cheat us, don't ask
> questions, and we'll help you when we feel like it" isn't going to be
> around for a long time.
> 

I agree totally.  After seeing some of the issues people were having 
with their customer support (or better, "flying off the handle at their 
customers") I decided to stay clear of them.

Survival of the fittest . . .

B.
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RE: [Asterisk-Users] 2 servers via PRI

[Michael Di Martino]

-Original Message-
From: Altus Snyman [mailto:[EMAIL PROTECTED] 
Sent: Monday, May 16, 2005 9:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] 2 servers via PRI

Good day all
How do i set a connection between 2 asterisk servers via PRI
In Bri I would set one to NT and TE
How shoud the zapata.conf and zaptel.conf look
And how should the cable be?
All I got on the web was to set one to "pri_net"...this cant be all?
And the cable 
> pin1 <--> pin4> pin2 <--> pin5> pin3 <--> pin6> pin4 <--> pin1> pin5
<--> pin2> pin6 <--> pin3> pin5 <--> pin8> pin8 <--> pin7
Please Help and advice
Thanks Altus

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I have one asterisk server connected to 2 Norstar mics systems via two
PRI lines. Here is how I did it

My Zaptel.conf
loadzone = us
defaultzone = us
#loadzone = us-old
#loadzone=gr
#loadzone=it
#loadzone=fr
#loadzone=de
#loadzone=uk
#loadzone=fi
#loadzone=jp
#loadzone=sp
#loadzone=no

#
# PRI's
#
span=1,0,0,esf,b8zs
#clear=1-24
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
#clear=25-48
bchan=25-47
dchan=48

my Zapata.conf

[channels]
usecallerid=yes
hidecallerid=no
usecallingpres=yes
callerid=asreceived
echocancel = yes   ; You can set this to 16, 32, 64, or 128, or 256
tweak to your needs.  Try 64.  Yes=128.
echotraining = 400 ; Ast trains to the beginning of the call, num is in
millisec.  0-4000.  Try 800.
echocancelwhenbridged = yes

context = internal
switchtype = dms100
signalling = pri_net
group = 1
channel => 1-23

context = internal
switchtype = dms100
signalling = pri_net
group = 2
channel => 25-47

my extensions.conf  (partial)

;Allows access to 2000 3000 Nortel extensions
exten => _2XXX,1,Dial(${TELX-MICS2}/${EXTEN:${TELX-MICS2-MSD}})
exten => _2XXX,2,Congestion
exten => _3XXX,1,Dial(${TELX-MICS1}/${EXTEN:${TELX-MICS1-MSD}})
exten => _3XXX,2,Congestion

I hope this help[s

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[Asterisk-Users] iaxy device

[Michael Di Martino]

I am attempting to get an iaxy device to connect to my
asterisk box over the public cloud however

It fails register and I cannot figure out why.

 Below is my iax.conf, iaxy setup file and debug output  from
iax2 debug.

 

My iax.conf

[u7402]

type=friend

accountcode=iaxy

host=dynamic

secret=u7402p

context=from-iaxy

disallow=all

allow=ulaw

callerid="my iaxy" <7402>

trunk=no

notify=yes

 

Mi iaxy setup file

[EMAIL PROTECTED] iaxyprov]# cat iaxy.conf.7402

; IAXY Provisioning description

;

dhcp

;ip: 216.207.244.130

;netmask: 255.255.255.192

;gateway: 216.207.244.129

codec: ulaw

;codec: adpcm

server: 207.251.84.198 

;altserver: 192.168.0.2

user: u7402 

pass: u7402p 

register

;heartbeat

;debug

;

; Feature tuning (default is all enabled)

;

;disablecid

;disablecw

;disablecidcw

;disable3way

[EMAIL PROTECTED] iaxyprov]#

 

Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type:
IAX Subclass: ACK    

   Timestamp: 2ms  SCall: 1 
DCall: 10605 [24.47.87.47:4569]

Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type:
IAX Subclass: REGREQ 

   Timestamp: 2ms  SCall: 10605 
DCall: 0 [24.47.87.47:4569]

  
USERNAME    : u7402

   REFRESH
: 60

   DEVICE TYPE : iaxy2

   SERVICE IDENT   : 000364000132

   PROVISIONG VER  : 326528057
    

 

 

Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type:
IAX Subclass: ACK    

   Timestamp: 2ms  SCall: 1  DCall:
10605 [24.47.87.47:4569]

 






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[Asterisk-Users] iaxy over the public cloud

[Michael Di Martino]

I am trying to get an iaxy device to connect to my asterisk box
over the public cloud however

It fails register and I cannot figure out why. Below is my
iax.conf, iaxy setup file and out from iax2 debug.

 

My iax.conf

[u7403]

type=friend

accountcode=iaxy

host=dynamic

secret=u7403p

context=from-iaxy

disallow=all

allow=ulaw

callerid="my iaxy" <7403>

trunk=no

notify=yes

 

Mi iaxy setup file

[EMAIL PROTECTED] iaxyprov]# cat iaxy.conf.7402

;

; IAXY Provisioning description

;

dhcp

;ip: 216.207.244.130

;netmask: 255.255.255.192

;gateway: 216.207.244.129

codec: ulaw

;codec: adpcm

server: 207.251.84.198 

;altserver: 192.168.0.2

user: u7402 

pass: u7402p 

register

;heartbeat

;debug

;

; Feature tuning (default is all enabled)

;

;disablecid

;disablecw

;disablecidcw

;disable3way

[EMAIL PROTECTED] iaxyprov]#

 

Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX
Subclass: ACK    

   Timestamp: 2ms  SCall: 1  DCall: 10605
[24.47.87.47:4569]

Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX
Subclass: REGREQ 

   Timestamp: 2ms  SCall: 10605  DCall: 0
[24.47.87.47:4569]

   USERNAME    : u7402

   REFRESH : 60

   DEVICE TYPE : iaxy2

   SERVICE IDENT   : 000364000132

   PROVISIONG VER  : 326528057     

 

 

Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX
Subclass: ACK    

   Timestamp: 2ms  SCall: 1  DCall: 10605
[24.47.87.47:4569]






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RE: [Asterisk-Users] voicemail

[Michael Di Martino]

Ok I have added the timeout value but it still does not pick. However
jus to test voicemail function
I comment out the first line and voice does pick up. What could be
wrong.

exten => 7403,1,Dial(IAX2/u7403/1/5)
exten => 7403,2,Voicemail(u7403)
exten => 7403,102,Voicemail(b7403)
exten => 7403,103,Hangup

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Seth
Remington
Sent: Friday, June 24, 2005 12:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] voicemail

On Thu, 2005-06-23 at 23:19 -0400, Michael Di Martino wrote:
> I am trying to setup voicemail for my iaxy device, however, i cannot 
> get it to work voicemail never picks up. Below is my config.
> Am i doing anything wrong here
> 
> >From my Extensions.conf file
> exten => 7403,1,Dial(IAX2/7403/10)

You did not specify a timeout in the dial command. Change it to:
exten => 7403,1,Dial(IAX2/7403/10,xx) <--- where xx is the number of
seconds you want the Dial command to attempt to connect the call before
it returns and proceeds to the next priority (i.e. voicemail).

> exten => 7403,2,Voicemail(u7403) 
> exten => 7403,102,Voicemail(b7403) 
> exten => 7403,103,Hangup 
> 
> >From my voicemail.conf 
> [telx.com] 
> 7403 => 7403 
> 
> 
> Thanks 
> Mike

Hope that helps.

-Seth



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[Asterisk-Users] voicemail

[Michael Di Martino]

I am trying to setup voicemail for my iaxy device, however, 
i cannot get it to work voicemail never picks up. Below is my config. 
Am i doing anything wrong here 

>From my Extensions.conf file 
    exten => 7403,1,Dial(IAX2/7403/10) 
    exten => 7403,2,Voicemail(u7403) 
    exten => 7403,102,Voicemail(b7403) 
    exten => 7403,103,Hangup 

>From my voicemail.conf 
    [telx.com] 
    7403 => 7403 


Thanks 
Mike






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Re: [Asterisk-Users] PRI trouble

[Michael Di Martino]

On Sun, 2005-06-12 at 20:09 -0400, Andrew Kohlsmith wrote:
> On Sunday 12 June 2005 06:10, Michael Di Martino wrote:
> > Out of the blue i started receiving the following error on my PRI line
> > which connects my asterisk server to a Norstar 0x32 key system.
> 
> Well first off, it's likely not the norstar 0x32 -- that is a 32-station 
> module.  You've likely got a MICS (my guess, I have one too).
> 
> > The asterisk zaptel.conf file was configure as follows and this config
> > worked for 6 months until friday. Nothing was changed on either system
> > prior to friday. here is teh zaptel.conf
> 
> Obviously something has.  :-)  What's the MICS say as far as its DTI 
> configuration?  What's its event log say?
> 
> > However i still get the same error. Please help we cannot connect call
> > form my norstar to asterisk w/ it dropping in 10 seconds.
> 
> Sounds like someone changed the pri signaling.
> 
> -A.
> ___
> Asterisk-Users mailing list
I am thinking it is a timing issue but i am not certain.
What type of signaling and timing do you use to connect to your 
MICS. And yes my 0x32 is a MICS.

Thanks
Mike

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Re: [Asterisk-Users] PRI trouble

[Michael Di Martino]

On Sun, 2005-06-12 at 20:09 -0400, Andrew Kohlsmith wrote:
> On Sunday 12 June 2005 06:10, Michael Di Martino wrote:
> > Out of the blue i started receiving the following error on my PRI line
> > which connects my asterisk server to a Norstar 0x32 key system.
> 
> Well first off, it's likely not the norstar 0x32 -- that is a 32-station 
> module.  You've likely got a MICS (my guess, I have one too).
> 
> > The asterisk zaptel.conf file was configure as follows and this config
> > worked for 6 months until friday. Nothing was changed on either system
> > prior to friday. here is teh zaptel.conf
> 
> Obviously something has.  :-)  What's the MICS say as far as its DTI 
> configuration?  What's its event log say?
> 
> > However i still get the same error. Please help we cannot connect call
> > form my norstar to asterisk w/ it dropping in 10 seconds.
> 
> Sounds like someone changed the pri signaling.
> 
> -A.
> ___
I thinking it is a timing issue, but i an not sure how to test.
What type of PRI signaling and timing do u use to connect to your MICS.
and yes my 0x32 is a MICS.

Thanks
Mike


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[Asterisk-Users] PRI trouble

[Michael Di Martino]

Out of the blue i started receiving the following error on my PRI line
which connects my asterisk server to a Norstar 0x32 key system.

The asterisk zaptel.conf file was configure as follows and this config
worked for 6 months until friday. Nothing was changed on either system
prior to friday. here is teh zaptel.conf

span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48

To try to fix the problem i changed the timing on span 2 from 0 to 1 

span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48

However i still get the same error. Please help we cannot connect call
form my norstar to asterisk w/ it dropping in 10 seconds.



Jun 12 10:51:51 NOTICE[213005]: PRI got event: 5 on Primary D-channel of
span 2
Jun 12 10:51:51 WARNING[213005]: No D-channels available!  Using Primary
on channel anyway 48!Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on
channel 35
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 36
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 37
Jun 12 10:51Out of the blue i started receiving the following error on
my PRI line
which connects my asterisk server to a Norstar 0x32 key system.

The asterisk zaptel.conf file was configure as follows and this config
worked for 6 months until friday. Nothing was changed on either system
prior to friday. here is teh zaptel.conf

span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48

To try to fix the problem i changed the timing on span 2 from 0 to 1 

span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48

However i still get the same error. Please help we cannot connect call
form my norstar to asterisk w/ it dropping in 10 seconds.



Jun 12 10:51:51 NOTICE[213005]: PRI got event: 5 on Primary D-channel of
span 2
Jun 12 10:51:51 WARNING[213005]: No D-channels available!  Using Primary
on channel anyway 48!Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on
channel 35
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 36
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 37
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 38
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 39
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 40
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 41
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 42
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 43
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 44
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 45
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 46
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 47:51 NOTICE
[262160]: Alarm cleared on channel 38
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 39
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 40
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 41
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 42
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 43
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 44
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 45
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 46
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 47

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[Asterisk-Users] PRI Trouble

[Michael Di Martino]

Out of the blue i started receiving the following error on my PRI line
which connects my asterisk server to a Norstar 0x32 key system.

The asterisk zaptel.conf file was configure as follows and this config
worked for 6 months until friday. Nothing was changed on either system
prior to friday. here is teh zaptel.conf

span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48

To try to fix the problem i changed the timing on span 2 from 0 to 1 

span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48

However i still get the same error. Please help we cannot connect call
form my norstar to asterisk w/ it dropping in 10 seconds.



Jun 12 10:51:51 NOTICE[213005]: PRI got event: 5 on Primary D-channel of
span 2
Jun 12 10:51:51 WARNING[213005]: No D-channels available!  Using Primary
on channel anyway 48!Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on
channel 35
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 36
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 37
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 38
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 39
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 40
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 41
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 42
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 43
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 44
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 45
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 46
Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 47


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[Asterisk-Users] Dialplan not showing up.

[Michael Di Martino]

I recently 
updated my sip.conf and extensions.conf files and after 
shutting 
down asterisk and restarting it (asterisk -cvvv)
it shows and 
empty dialplan (show dialplan)
*CLI> 
show dialplan-= 0 extensions (0 priorities) in 0 contexts. 
=-
 
What could 
cause somthing like this
 
below is a 
copy of my extensions.conf file located in 
/etc/asterisk/
[EMAIL PROTECTED] asterisk]# more 
extensions.conf [general]static=yeswriteprotect=yes
 
[bogons]exten => 
_.,1,Congestion
 
[from-sip]exten => 
1001,1,Dial(SIP/1001,20)exten => 1001,2,Voicemail(u1001)exten => 
1001,102,Voicemail(b1001)exten => 1001,103,Hangup
 
exten => 
7301,1,Dial(SIP/7301,20)exten => 7301,2,Voicemail(u7301)exten => 
7301,102,Voicemail(b7301)exten => 7301,103,Hangup
 
exten => 
7302,1,Dial(SIP/7302,10)exten => 7302,2,Voicemail(u7302)exten => 
7302,102,Voicemail(b7302)exten => 7302,103,Hangup
 
exten => 
1999,1,VoicemailMain(${CALLERIDNUM})
 
[EMAIL PROTECTED] asterisk]# 

 
 
And here is 
my sip.conf file:
 
[EMAIL PROTECTED] asterisk]# more sip.conf 
[general]port=5060bindaddr=0.0.0.0allow=allcontext=bogons
 
[1001]type=friendcontext=from-sipusername=1001callerid="1001"<1001>[EMAIL PROTECTED]host-dynamicdtmfmode=rfc2833
 
[7301]type=friendcontext=from-sipusername=7301callerid="mdm" 
<7301>[EMAIL PROTECTED]host=dynamicdtmfmode=rfc2833
 
[7302]type=friendcontext=from-sipusername=7302callerid="Moses" 
<7302>[EMAIL PROTECTED]host=dynamicdtmfmode=rfc2833[EMAIL PROTECTED] 
asterisk]# 

Regards, 
Michael 
DiMartino Director of MIS 
The telx Group, 
Inc. 17 
State St, 33rd Floor New York, NY 10004 
T: 212.480.3300 
X2022 C: 
646.207.6603   
 
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[Asterisk-Users] Fedore CORE 2

[Michael Di Martino]

Does 
Asterisk support Fedora Core-2 (2.6 Kernel)?
 

Regards, 
Michael 
DiMartino Director of MIS 
The telx Group, 
Inc. 17 
State St, 33rd Floor New York, NY 10004 
T: 212.480.3300 
X2022 C: 
646.207.6603   
 
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FW: [Asterisk-Users] USB Controller ztdummy

[Michael Di Martino]

Thanks
How do you use the rtc replacement with zaptel 
extensions


From: Giudice, Salvatore 
[mailto:[EMAIL PROTECTED] Sent: Friday, April 15, 2005 5:11 
PMTo: Michael Di MartinoSubject: RE: [Asterisk-Users] USB 
Controller ztdummy


Looks like you don’t 
have a usb controller or if you compiled a custom kernel, you may have not 
installed/loaded a usb module. If the system is single cpu, you can use a rtc 
replacement with zaptel extensions or you can buy even a lowly fxo card. Either 
will supply a clock.
 




From: Michael 
Di Martino [mailto:[EMAIL PROTECTED] Sent: Friday, April 15, 2005 3:52 
PMTo: Asterisk Users Mailing 
List - Non-Commercial DiscussionSubject: [Asterisk-Users] USB Controller 
ztdummy
 

 


I am attempting my fist 
asterisk install.

I do not have any digum 
cards so use ztdummy. 

However according to the 
Asterisk install guide it says to look for one of the 


following USB controller 
chips.

UHCI USB controller of OHCI 
USB Controller. 

 

However, when I run lsmod to 
determine which have neither show up.

Can I continue the install 
and if I do will I have timing issues?

 

This is the output of 
lsmod
[EMAIL PROTECTED] root]# 
lsmodModule  
Size  Used 
byipv6  
184288  10 
parport_pc 
19392  0 
lp  
8236  0 
parport    
29640  2 
parport_pc,lpautofs4    
10624  0 
sunrpc    
101064  1 
e1000  
68492  0 
floppy 
47440  0 
sg 
27552  0 
microcode   
4768  0 
binfmt_misc 
7176  1 
dm_mod 
33184  0 
button  
4504  0 
battery 
6924  0 
asus_acpi   
8472  0 
ac  
3340  0 
ext3  
102376  2 
jbd    
40216  1 
ext3ata_piix    
5380  3 
libata 
29312  1 
ata_piix,[permanent]sd_mod 
16384  5 
scsi_mod   
91344  3 sg,libata,sd_mod[EMAIL PROTECTED] root]# 


Regards, 
Michael 

  


 
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[Asterisk-Users] USB Controller ztdummy

[Michael Di Martino]

 

I am 
attempting my fist asterisk install.
I do not 
have any digum cards so use ztdummy. 
However 
according to the Asterisk install guide it says to look for one of the 

following 
USB controller chips.
UHCI USB 
controller of OHCI USB Controller. 
 
However, 
when I run lsmod to determine which have neither show up.
Can I 
continue the install and if I do will I have timing issues?
 

This is the output of lsmod
[EMAIL PROTECTED] root]# 
lsmodModule  
Size  Used 
byipv6  
184288  10 
parport_pc 
19392  0 
lp  
8236  0 
parport    
29640  2 
parport_pc,lpautofs4    
10624  0 
sunrpc    
101064  1 
e1000  
68492  0 
floppy 
47440  0 
sg 
27552  0 
microcode   
4768  0 
binfmt_misc 
7176  1 
dm_mod 
33184  0 
button  
4504  0 
battery 
6924  0 
asus_acpi   
8472  0 
ac  
3340  0 
ext3  
102376  2 
jbd    
40216  1 
ext3ata_piix    
5380  3 
libata 
29312  1 
ata_piix,[permanent]sd_mod 
16384  5 
scsi_mod   
91344  3 sg,libata,sd_mod[EMAIL PROTECTED] root]# 


Regards, 
Michael 

  
 
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RE: [Asterisk-Users] new install

[Michael Di Martino]

 > From: Michael Di 
Martino  > Sent: Friday, 
April 15, 2005 3:15 PM > To: 
asterisk-users@lists.digium.com > Subject: 
[Asterisk-Users] new install

 > I am 
attempting my fist asterisk install.
 > I do 
not have any digum cards so use ztdummy. 
 > However according to the Asterisk 
install guide it says to look for one of the 
 > following USB controller 
chips.
 > UHCI 
USB controller of OHCI USB Controller. 
 
 > However, when I run lsmod to 
determine which have neither show up.
 > Can I 
continue the install and if I do will I have timing 
issues?
 

 > Regards,  > Michael 
   >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
This is the output of 
lsmod
[EMAIL PROTECTED] root]# 
lsmodModule  
Size  Used 
byipv6  
184288  10 
parport_pc 
19392  0 
lp  
8236  0 
parport    
29640  2 
parport_pc,lpautofs4    
10624  0 
sunrpc    
101064  1 
e1000  
68492  0 
floppy 
47440  0 
sg 
27552  0 
microcode   
4768  0 
binfmt_misc 
7176  1 
dm_mod 
33184  0 
button  
4504  0 
battery 
6924  0 
asus_acpi   
8472  0 
ac  
3340  0 
ext3  
102376  2 
jbd    
40216  1 
ext3ata_piix    
5380  3 
libata 
29312  1 
ata_piix,[permanent]sd_mod 
16384  5 
scsi_mod   
91344  3 sg,libata,sd_mod[EMAIL PROTECTED] root]# 
  

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[Asterisk-Users] new install

[Michael Di Martino]

I am 
attempting my fist asterisk install.
I do not 
have any digum cards so use ztdummy. 
However 
according to the Asterisk install guide it says to look for one of the 

following 
USB controller chips.
UHCI USB 
controller of OHCI USB Controller. 
 
However, 
when I run lsmod to determine which have neither show up.
Can I 
continue the install and if I do will I have timing issues?
 

Regards, 
Michael   

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[Asterisk-Users] dial plan

[Michael Di Martino]

I have just 
inherited a Asterisk box which is configured as follows.
 
10 internal 
Sip Phones 
 
3 Pots 
Lines
 
1 voip 
provider (SIP)
 
Call come in 
over the pots lines however Outbound goes out thru the VOIP 
provider.
However 
looking at the configs I cannot figure out what controls how call are sent 
out.
In other 
words where in the config files does it determine that all outbound 

calls go to 
the VoIP provider?
 
 
Thanks in 
advance.
Mike
 

  

 
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[Asterisk-Users] Asterisk PBX Manager

[Michael Di Martino]

Title: Asterisk PBX Manager







Does anyone on this list have any experience Thirdlane.com's Asterisk PBX Manager?

And if so what do you think of it?


Regards,

Michael DiMartino

Director of MIS

The telx Group, Inc.

17 State St, 33rd Floor

New York, NY 10004

T: 212.480.3300 X2022

C: 646.207.6603

 



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[Asterisk-Users] need info

[Michael Di Martino]

What is the unsubscribed address?

Thanks
Michael

-Original Message-
From: Steve Underwood [mailto:[EMAIL PROTECTED] 
Sent: Friday, February 18, 2005 11:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Re: quadbri and spandsp

You need to use the caller parameter. Something like:

Channel:Zap/G1/
Application:txfax
Data:/root/fax.tif|caller

might work better.

Regards,
Steve

Blas wrote:

>Yes. This is my process:
>
>1.- Create a /tmp/sample.call
>--
>Channel: Zap/G1/X  <--- Here fax machine number
>Application: txfax
>Data: /root/fax.tif
>--
>
>2.- Shell in a linux terminal:
>---
>mv /tmp/sample.call /var/spool/asterisk/outgoing/
>---
>
>I don't have any 'fax' extension in my extensions.conf
>
>Is correct my process?
>
>Thank you. Blas.
>
>
>
>  
>

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[Asterisk-Users] DTMF Payload type

[Michael Di Martino]

 To All 
I am using a SNOM 190 w/Asterisk server. 
Here is my sip.conf 
[7501] 
type=friend 
context=external 
username=7501 
callerid="Telx 7501" <7501> 
[EMAIL PROTECTED] 
host=dynamic 
dtmfmode=rfc2833 

My question is this. With above settings in my sip.conf specially
"dtmfmode=rfc2833" 
What should my "DTMF Payload Type:" be set to on my SNOM 190 phone.
Currently it is set to 101. 

Should it be set to rfc2833? 

Regards, 
Michael DiMartino 

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[Asterisk-Users] DTMF Payload Type:

[Michael Di Martino]

Title: DTMF Payload Type:







To All

I am using a SNOM 190 w/Asterisk server. 

Here is my sip.conf

[7501]

type=friend

context=external

username=7501

callerid="Telx 7501" <7501>

[EMAIL PROTECTED]

host=dynamic

dtmfmode=rfc2833


My question is this. With above settings in my sip.conf specially "dtmfmode=rfc2833"

What should my "DTMF Payload Type:" be set to on my SNOM 190 phone. Currently it is set to 101.


Should it be set to rfc2833?


Regards,

Michael DiMartino

Director of MIS

The telx Group, Inc.

17 State St, 33rd Floor

New York, NY 10004

T: 212.480.3300 X2022

C: 646.207.6603

 



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[Asterisk-Users] SNOM 190 and dtmf

[Michael Di Martino]

I have the dtmfmode in sip.conf 
set to use rfc 2833
however, when my users have to enter pin numbers to join let say 
someone's
conference bridge the pin is received twice.
 
Any ideas on how to solve 
this?
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RE: [Asterisk-Users] Passing PIN Numbers

[Michael Di Martino]

Title: Passing PIN Numbers



I have the dtmfmode in sip.conf 
set to use rfc 2833
however, when my users have to enter pin numbers to join let say 
someone's
conference bridge the pin is received twice.
 
Any ideas on how to solve this?


From: Rene Kluwen [mailto:[EMAIL PROTECTED] 
Sent: Monday, January 17, 2005 1:41 PMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 
Passing PIN Numbers

This is a long shot, I am not sure if it will solve 
your problem:
 
Did you try to change dtmfmode in 
sip.conf?
 
Rene Kluwen
Chimit
 

  - Original Message - 
  From: 
  Michael Di Martino 
  To: asterisk-users@lists.digium.com 
  
  Sent: Friday, January 14, 2005 3:53 
  PM
  Subject: [Asterisk-Users] Passing PIN 
  Numbers
  
  To All If anyone can shed any light on this 
  it would be greatly appreciated. My phones are unable to enter pins numbers correctly when 
  required by the party they are calling. 
  For example I was given an 
  outside number to attend conference bridge. After the call was connected it 
  required me to enter a 4 digit PIN. Now here is the problem whenever I enter a 
  pin it is received twice. For example if the PIN is 1234 they receive it as 
  12341234.
  Any ideas what could be 
  wrong? 
  BTW we are using SNOM 190 ip phones 
  (sip) 
  Regards, Michael DiMartino Director of MIS The telx Group, Inc. 17 State St, 33rd Floor New York, NY 10004 T: 212.480.3300 X2022 C: 646.207.6603   
  
  

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[Asterisk-Users] Passing PIN Numbers

[Michael Di Martino]

Title: Passing PIN Numbers






To All

If anyone can shed any light on this it would be greatly appreciated.

My phones are unable to enter pins numbers correctly when required by the party they are calling.


For example I was given an outside number to attend conference bridge. After the call was connected it required me to enter a 4 digit PIN. Now here is the problem whenever I enter a pin it is received twice. For example if the PIN is 1234 they receive it as 12341234.

Any ideas what could be wrong?


BTW

we are using SNOM 190 ip phones (sip)



Regards,

Michael DiMartino

Director of MIS

The telx Group, Inc.

17 State St, 33rd Floor

New York, NY 10004

T: 212.480.3300 X2022

C: 646.207.6603

 



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[Asterisk-Users] sip phones

[Michael Di Martino]

sip phones






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[Asterisk-Users] Sip

[Michael Di Martino]

Regards,
Michael Di Martino
Director of MIS
The telx Group
Office: 212 480 3300  X.2022
Cell: 646 207 6603
[EMAIL PROTECTED]
--
Sent from my BlackBerry Wireless Handheld

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[Asterisk-Users] GUI

[Michael Di Martino]

Title: GUI






I am looking for a good Asterisk GUI to manage my server. Any Suggestions?


Regards,

Michael DiMartino

Director of MIS

The telx Group, Inc.

17 State St, 33rd Floor

New York, NY 10004

T: 212.480.3300 X2022

C: 646.207.6603

 



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[Asterisk-Users] (no subject)

[Michael Di Martino]

What is the general consensus on the Polycom SIP Phones?

I am getting random gargled up sounds on mine and I really do think it is the Polycom


Regards,

Michael DiMartino


 



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RE: [Asterisk-Users] Nortel Phones.

[Michael Di Martino]

I am using the Nortel M7310

I have my Asterisk connected to  my Norstar 0x32 MIC software version
4.1 VIA a PRI. Works great



-Original Message-
From: Julio Arruda [mailto:[EMAIL PROTECTED] 
Sent: Monday, October 25, 2004 1:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Nortel Phones.

Remarks inline:

Cian O'Sullivan wrote:
> Hello,
> 
> I am wondering if anyone is using the Nortel 2001 2002 or 2004 phones 
> on their asterisk implementation.  My local dealer says they are not 
> compatible with any open source implementations.  Is there a phone 
> compatibility list somewhere?

First, 2 disclaimers...I work for Nortel and I don't speak for Nortel
(huh ?)...
The I2004 for sure (the original one, I understand there are distinct
versions) would run just a protocol known as Unistim (you may say is
like the Skinny protocll for a cisco phone).

[], 
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[Asterisk-Users] SIP phones

[Michael Di Martino]

Title: SIP phones






I am looking for a loud ringing SIP phone. I am presently using the Polycom  and just cannot loud enough to hear it over the din in a collocation room.


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[Asterisk-Users] Asterisk and Norstar 0X32 MICS

[Michael Di Martino]

Title: Asterisk and Norstar 0X32 MICS







I have the following setup a Norstar MICS 0X32 with 8 POTS Lines connected to the PSTN, and one ASTERISK server connected to the Norstar MICS VIA a PRI line.

Now here is the problem I cannot get the MICS to accept a call from the ASTERISK SERVER when that call is for an outside line(meaning dialinag a 10 digit # to someone outside my phone sys). However, I can dial a 4 digit ext from the asterisk server to an analog phone off the Norstar MICS.

Does anyone have any idea how I can get the Norstar to accept outbound calls from Asterisk? 


Thanks in advance

Mike


Regards,

Michael 

 



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[Asterisk-Users] Asterisk and Norstar 0X32 MICS

[Michael Di Martino]

I have the following setup a Norstar MICS 0X32 with 8 POTS Lines
connected to the PSTN, and one ASTERISK server connected to the Norstar
MICS VIA a PRI line.

Now here is the problem I cannot get the MICS to accept a call from the
ASTERISK SERVER when that call is for an outside line(meaning dial a 10
digit # to someone outside my phone sys). However, I can dial a 4 digit
ext from the asterisk server to an analog phone off the Norstar MICS.

Does anyone have any idea how I can get the Norstar to accept outbound
calls from Asterisk? 

Thanks in advance
Mike
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[Asterisk-Users] Newbie

[Michael Di Martino]

I am interested in setting up an Asterisk server as my  home phone system.

I ultimately want one 10 digit phone number, three  extensions, and an auto attendant  
My current phone service provider is Vonage, I have one line with call waiting.

My concern is will I need to add additional lines if I want the auto attendant  handle 
multilple calls.
For example a call comes in and the auto attendant sends the call to ext 1. Now while 
the person on ext 1 
Is still conversating can another call be handled by the auto attendant?
Regards,
Michael Di Martino
Director of MIS
The Telx Group
Office: 212 480 3300 X2022
Cell: 646 207 6603
[EMAIL PROTECTED]
--
Sent from my BlackBerry Wireless Handheld

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[Asterisk-Users] asterisks and vonage

[Michael Di Martino]

to start with i am new to asterisks and i am also a telcom idiot.
 
with that said i have one vonage line i would like to hook up in my soon to be built 
Asterisk ippbx server.
Now with the one Vonage (with call waiting) line can i receive more one call using an 
auto attendant route the call the approiate extention?
 
thanks
mike
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