Hi,
Urgently response would be wonderful, system is a Fedora Core 2.
I have a Ericsson BP250 connected to 1 port on the TE405P and another
connected to a local telco ISDN30.
I have been running CVS-HEAD from about a 2 months ago and upgraded it
again just in cause it was a version issue
Update...
Figured out it was a faulty port in the te405p, swapped to a spare port
and all it good, now to get warranty.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Bean
Sent: Saturday, 6 August 2005 10:54 AM
To: Asterisk Users Mailing
Hi,
Running Asterisk CVS-Head latest on a Dual P3 800 1Gb Ram.
For some odd reason now that I have the asterisk box almost to the stage
I want it, I hit a problem.
I have a te405p in the system, Zap/g1 is connected to the telco as an
ISDN 30, Zap/g4 is connected by ISDN Primary Rate to an
Hi,
Asterisk 1.0.7
TE405P - Port 1 - ISDN30 telco
- Port 4 - Primary Rate connection to Phone system
The system has a mixture of 20+ sip phones and the 50 odd extensions on
the phone system connected to Port 4.
What I want to accomplish is to be able to denigh access to certain
outgoing
can call NZ) exten
= _0011.,Exec(checkperms);
checkperms.agi would then match against a list.
--Rob
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Bean
Sent: Sunday, June 12, 2005 8:40 AM
To: asterisk-users@lists.digium.com
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GXP2000 and hint LED's
On Thu, 9 Jun 2005, James Bean wrote:
Has anyone got the hint function working, and maybe with the GXP2000.
I don't think the current firmware release for the GXP-2000 supports
SUBSCRIBE/NOTIFY
Did that pre-release version fix that bug where the other party can
hear you when you pressed the transfer button ?
Does it also enable the leds next to the speeddial buttons like the
snoms ?
Unfortunately not, Grandstream didn't admit to me that they were going
to program the LED's like the
Does anyone know if the parked call queue has hint built into it.
I want to program up on a series of sip phones that support subscribe
notify on the led function keys the parked call queue (791-795)
positions so that its easy for people to know there is someone in the
queue and the calls can be
]
-Original Message-
From: James Bean [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 09, 2005 5:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] GXP2000 and hint LED's
Did that pre-release version fix that bug where the other party can
hear
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
G
Sent: Thursday, 9 June 2005 1:52 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Clicks in audio with TE100P PRI
Thanks for your answer. Googling in the lists I found what
]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = all
allow = ilbc
allow = alaw
allow = ulaw
nat=disable
srvlookup=no
localnet=192.168.69.0/255.255.255.0
subscribecontext = sip
incominglimit = 1
[690]
type=friend
secret=secret
host=dynamic
callerid=James Bean 690
defaultip=192.168.69.250
Hi,
Weird issue, 2 asterisk boxes running 1.0.7, when I call iax from box 1
to box 2 it works fine, when I dial from box 2 to box 1 I get a
On Box 1
Apr 11 17:25:39 WARNING[8969]: chan_iax2.c:5553 socket_read: Call
rejected by 192.168.69.1: No authority found
On Box 2
Apr 11 17:26:07
Hi,
Got 2 asterisk boxes, Box 1 has SIP users, and Box 2 has a E1 card
connected to a TDA200, when a sip user from box 1 calls someone on the
tda200 there is no ringing noise just dead silence until the person on
the TDA picks up there extensions.
Is there a way in thse situations to supply a
Whooppss after research for several hours before posting, another
asterisk user passed on the answer to me.
Add ,r to the Dial string over the E1 to hear the ringing on the line.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Bean
Sent
:[EMAIL PROTECTED] On Behalf Of James Bean
Sent: Monday, 11 April 2005 5:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX calls between asterisk boxes works 1 way
only
Hi,
Weird issue, 2 asterisk boxes running 1.0.7, when I call iax from box 1
to box 2
b) If you are planning to use SIP make sure you configure it properly
to work with NAT.
SIP has a lot of issues with NAT. The alternative is using IAX but the
IAX deskphones
arent as feature rich as the SIP phones. Also take into account the
cost of deploying IP phones.
Intent wrote that he is
Is the E1 card an isdn card or something else? There are a several
signalling systems that can run over an E1.
When running cas you do not have a D channel for the signalling.
Instead each voice channel has a few dedicated
bits in channel 16 (hence Channel Associated Signalling). This is used
Well, you can connect to the telco using non-isdn signalling as well.
In Europe isdn is by far
the most common signalling form used on an E1. Can you find the model
number for the E1 card?
An E1 always has 30 voice channels, one signalling channel (running CAS
or
CCS) and one timing channel.
Whooppss had pri_cpe set, redid the debug as attached.
They seem the same but just in case.
James
Enabled EXTENSIVE debugging on span 1
*CLI Sending Set Asynchronous Balanced Mode Extended
[ 02 01 7f ]
Unnumbered frame:
SAPI: 00 C/R: 1 EA: 0
TEI: 000EA: 1
M3: 3 P/F: 1 M2: 3
Asterisk does not see anything coming in on the D channel. What does
zttool say about the state of the link?
zttool shows the card exists, the following information shows when
select the card
Main Screen - Alarms Ok / Span Digium Wildcard E100P E1/PRA Card 0
Current Alarms: No Alarms
Sync
Hi, I hope someone can help me with this
Asterisk 1.0.6 Zaptel 1.0.6 Libpri 1.0.6, 1 Digium E100P card installed
Panasonic TDA200 firmware v2.0.6 E1 Card Firmware 1.0.2
System is located in Australia, so as technologies go, I believe it is twist on
the euro standard for the E1 signalling.
I am no expert but I hope I can be of
help.
What is your /etc/zaptel.conf and
/etc/asterisk/zapata.conf?
There is 2 points it can be a problem that I have found,
the cable, as depending on requirements a normal patch lead or a specific pin
cross over cable maybe needed.
The other place
I just upgraded 4 boxes to 1.0.6 without issue, then I went to upgrade
my personal test box which I am playing with call logging cdr stuff on,
writing to postgres and now asterisk now crashes on boot with the
following error.
[app_while.so]Mar 10 17:20:04 WARNING[7239]: loader.c:258
Hi,
I have postgresql and * all up and running as the latest cvs-250205,
although something weird.
Every outgoing call regardless of whether or not it is answered or busy
or just rings out in the database the entry has the disposition as
ANSWERED, instead of BUSY or NOT ANSWERED.
As a test I
James Bean [EMAIL PROTECTED] wrote:
[...]
Every outgoing call regardless of whether or not it is answered or
busy or just rings out in the database the entry has the
disposition
as ANSWERED, instead of BUSY or NOT ANSWERED.
As a test I intentionally rang numbers that would be busy
James Bean [EMAIL PROTECTED] wrote:
[...]
I am sorry I did not see anything in any of the docs about analogue
lines causing ANSWERED response on all calls. Could you point me in
the right direction to a fix or setup that fixes this situation?
The only real fix is to get some form
For MySQL and other glorified flat-file databases, you would
need to postprocess the data. You may feel more confident
skipping triggers and doing this anyway.
So by that any calls that go out over the net using IAX to
the telco
are considered digital and will report correctly?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mark Benson
Sent: Thursday, 24 February 2005 8:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallTransfer
I get the impression that the
to be * nto sending the
hint to the snom phone.
Any input on this would be very much appreciated.
The one thing I have not tried is doing the hint as
exten = 691,hint,691 ???
James Bean
Snom phone SIP Trace
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.69.1:5060
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hecken, Guido
Sent: Thursday, 24 February 2005 3:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Snom phone hint exten question
exten =
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hecken, Guido
Sent: Thursday, 24 February 2005 3:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Snom phone hint exten question
exten =
Hi,
Hoping someone has run into the same issue.
I have an * 1.0.5 tdm400p and 2 fax machines on grandstream 486 boxes.
When a fax comes in, no problem receives it fine. When you try to send a
fax out just as the fax seems to be finishing the send you get a comms
error on the fax machine and it
I haven't used it in a while, but I had to put subscribecontext=sip
for the phone's (in your case the snom) sip entry.
This seems like it has been removed from the wiki. Has it
changed or
is this incorrect?
Hi James,
I have just found out that all you need to do is make
I am going to now sit in a corner and go quietly insane
while playing
the banyo with no strings.
Still doesn't work, I dialed in an outside line and picked up the
receive on extension 691, yet the light on the snom phone
did not come
on. I dialed out of extension 691 to an
this:
http://www.phanderson.com/iom141.html
-Original Message-
From: James Bean [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 19, 2005 11:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] External relay triggered by Asterisk
extension
:36:04 +1000, James Bean
[EMAIL PROTECTED] wrote:
Putting bt-karen in the destination of the snom doesn't work, i.e.
pushing the button the phone says no such destination.
exten = 691,hint,SIP/bt-karen
exten = 691,1,SetMusicOnHold(random)
exten = 691,2,Dial(SIP/bt-karen,30,tr) exten
Hi,
I need to have it so that if someone is on their sip phone that any
other attempts to contact that phone will result in a transfer to
voicemail.
Someone mentioned there might be a setting like numofcalls = 1 in the
sip.conf so that only 1 call would every be sent to the sip phone but I
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af James Bean
Sendt: 19. februar 2005 08:14
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Snom phone hint exten question
Hi,
I am sorry
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af James Bean
Sendt: 19. februar 2005 08:14
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Snom phone hint exten question
Hi,
I am sorry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thorben Jensen
Sent: Saturday, 19 February 2005 8:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Snom phone hint exten question
Unfortunately
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thorben Jensen
Sent: Saturday, 19 February 2005 8:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Snom phone hint exten question
Unfortunately
No its setup in the snom as 691 not bt-karen I will test that now.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thorben Jensen
Sent: Saturday, 19 February 2005 8:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thorben Jensen
Sent: Saturday, 19 February 2005 8:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Snom phone hint exten question
Have you
Just after some peoples impressions if they have used this phone.
It has 10 function buttons which I am hoping can be individually
programmed for destination to accept hints from asterisk.
Any input would be very much appreciated.
James
___
at 08:38 +1000, James Bean wrote:
Just after some peoples impressions if they have used this phone.
It has 10 function buttons which I am hoping can be individually
programmed for destination to accept hints from asterisk.
What do you mean by this?
I'm not sure I understand.
If you
Has anyone every setup an external open/close relay, off say a serial
interface, and have an extension trigger the relay?
Why I ask is I have a student accomodation where I am installing an
asterisk box to supply phone services to the tenants, there is already
an intercom system in the main
and doing this or
have any better ideas or suggestions?
As an extra note I am in Australia so not all brands are available down
here.
James Bean
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk
Hi,
I am sorry to be asking this but the wiki is down and has been for a
couple of days and I need to get this working before Monday to get my
live system setup.
Trying to get the Snom 190's and soon to arrive 3com 3102's to use the
function keys and for the life of me I can't work it out from
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MultiLine Sip Phones
would an option where you could view it from a website or on your
computer be good?
coz there are a few ones out there like that, one in the wiki
James Bean wrote:
Sorry Newbie
callerid=James Bean 690
defaultip=192.168.69.250
dtmfmode=rfc2833
mailbox=690
[bt-karen]
type=friend
secret=apassword
host=dynamic
callerid=Karen Colomb 691
defaultip=192.168.69.251
dtmfmode=info
mailbox=691
___
Asterisk-Users mailing list
[EMAIL PROTECTED
=
alawdisallow =
ulawnat=disablesrvlookup=nolocalnet=192.168.69.0/255.255.255.0subscribecontext
= sip
[snom-james]type=friendsecret=passwordhost=dynamiccallerid="James
Bean"
690defaultip=192.168.69.250dtmfmode=rfc2833mailbox=690
[bt-karen]type=friendsecret=passwordhost=dynamiccallerid=&quo
Trying to setup asterisk and zaptel on a Fedora Core 3. Its all working
after reading up on udev but I still get errors.
[EMAIL PROTECTED] ~]# ztcfg -v
Zaptel Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 04: FXS Kewlstart
extensions.conf. Does
it work with another (soft)phone?
Regards,
Joris
On Oct 15, 2004, at 1:51 PM, James Bean wrote:
I am having a problem with my new SNOM190 and my asterisk box.
Incoming calls to the SNOM work perfectly, but when i dial-out I get a
Not Found: number dialed on the SNOM
Hi,
I do apologise I only have a basic understanding of VoIP and H323, here
is my situation, any help would be very much appreciated.
I am trying to coax my asterisk 1.0.1 box using oh323 0.6.3b with
openh323 13.5 pwlib v1.6.6 (I purchased 1 G.729 license from digium
and installed it
I am having a problem with my new SNOM190 and my asterisk
box.
Incoming calls to the SNOM work perfectly, but when i
dial-out I get a "Not Found: number dialed" on the SNOM display
everytime I try, nothing shows up on the console of the asterisk box so its not
even touching it.
I have the
Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P
--On Wednesday, October 13, 2004 16:04 +1000 James Bean
[EMAIL PROTECTED]
wrote:
a) Ensure you actually have the callerid service provided to your
line,
this is usually an extra charge from telstra (AFAIK)
Yep my analog
, and tried a some of the same ideas. No
result.
But at least we both know that a few people in Australia are using
Asterisk!
Later,
PaulH
-Original Message-
From: James Bean [mailto:[EMAIL PROTECTED]
Sent: Wednesday, 13 October 2004 4:05 PM
To: Asterisk Users Mailing List - Non
I am trying to setup a SNOM 190 with my asterisk box but having a few
problems
When a call comes in it connects and rings and I can talk no problems...
If I try to call out with the phone I get...
NOTICE[-165364816]: chan_sip.c:7561 handle_request: Unknown SIP command
'PUBLISH' from
Title: Passing CallerID to SIP phone from TDM400P
Hi,
Sorry, newbie, I want to pass the incoming callerid information through to my sip phone but when an incoming call gets passed through it says asterisk on the display instead of the number.
Being in australia callerid information is
callerid=Karen 691
defaultip=192.168.69.251
dtmfmode=inband
mailbox=691
That would be what I would do.
On Oct 13, 2004, at 12:38 AM, James Bean wrote:
Hi,
Sorry, newbie, I want to pass the incoming callerid information
through to my sip phone but when an incoming call gets passed through
Sorry, I explained this wrong.
I am wanting the callerid of the incoming caller from my analogue
line
on the TDM400P to be passed TO the sip phone so the sip phone display
shows the phone number of the incoming caler from the call on the
TDM400P.
It shows any callerid information from
FC2, Asterisk 1.0.0, Zaptel 1.0.0
TDM400P Port 1 FXS Port 4 FXO
Standard analogue handset plugged in with pstn line.
Problem:
I have 2 analog phones that I use, when plugged directly into pstn line
both phones work perfectly, dialing no issues. When I plug the handsets
into the TDM400P, one
Sorry to post such a newb set of questions but I have been hammering
about trying to get Asterisk running on FC2 machine reading everything
available (I think that is what stuffed me, shouldn't have read it all
:-) ).
Config
FC2 running Asterisk 1.0.0, with the h323 compiled in and installed
Scenario,
I got some very good help earlier from Joseph getting me up and started
but I have a couple of small problems still.
Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4
Analog dialout line and Analog handset plugged in.
Problems:
1.
Incoming calls work and the phone
Specs
FC2, Asterisk 1.0.0, Zaptel 1.0.0
TDM400P Port 1 FXS Port 4 FXO
Standard analogue handset plugged in with pstn line.
Problem:
When I go to dialout it drops numbers on the outgoing number.
Keys dialed from handset were
9 0418800185
I tried hitting the keys slowly as well as at my
Sorry very very very newbie here,
I just started setting up a asterix box as a test environment for my
work to see if it is a viable solution.
I have a standard TMD400P Development Kit with a FXS and FXO module on
it, and a standard analog handset plugged into the FXS module and a
Analog phone
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