:) heehhehehe not that newbie :)
At least for linux :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb
Sent: Jueves, 24 de Febrero de 2005 12:54 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
DiscussionSubject: Re:
[Asterisk-Users] CallerID problem
Your phones should have a timezone or GMT offset setting. It
appears your in the Central zone so set it to -6. Also, setup a NTP server
and have your phones sync with it. I find it works fine to use the *
server for the NTP server.On Thu, 2005-02
You can't change the callerid on an outgoing PSTN call (at least on
analog lines).
To modifiy the callerid on incoming calls, you could do something like
this (not tested):
[incoming-line1]
exten = s,1,setCidName(Line1: . ${CALLERID})
exten = s,2,Goto(Incoming,s,1)
[incoming]
exten =
Worked Great! Thx Julian..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M.
Sent: Lunes, 21 de Febrero de 2005 02:46 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID
You can't change
I'm doing something like that on my system --
http://muware.com/asterisk
-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Monday, February 21, 2005 1:37 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] CallerID
Guys
Guys... I see there is a callerid parameter on zapata.conf... what does that
cid modify? the callerid people see when you call them using any PSTN line?
Is there a way to send the SIP phone the incoming callerid frpm PSTN lines
asrecevied and append some string depending on the line it is
I have a regular telephone that when hooked to a standard POTS line, the
incoming callerid signal sets the time on the phone. I do this because
the phone has no internal battery and any little power blip causes the
time to reset to 1/1/98.
Is there a way to either pass the date and time along
quote who=Robert Webb
Is there a way to either pass the date and time along from my POTS
line going through a TD400P with one FXO and one FXS to my phone?
Or even have * send the date and time through the caller id when
that extension is called??
It should already pass the time from the
Hi Remco,
-Original Message-
Isn't it possibly to change caller id 'on the fly' like most
pbx's do? If
you do a call transfer you can see the local extension first and the
caller id of the incoming call afterwards?
Within the Asterisk core you probably could do that, but it is
Hi to all,
can you suggest to me the best way to avoid problems in the CDRs for anonymous
sip calls?
I have some peoples that set Send Anonymous : Yes in their Grandstream phones
and i don't receive the username as phone number that i use to make billing.
It is empty. The only place where there
Hi,
Il giorno sab, 05-02-2005 alle 11:11 +0100, Marcello Lupo ha scritto:
Hi to all,
can you suggest to me the best way to avoid problems in the CDRs for
anonymous
sip calls?
I have some peoples that set Send Anonymous : Yes in their Grandstream phones
and i don't receive the username as
Hi,
On Fri, 4 Feb 2005 21:35:19 -0500, mattf [EMAIL PROTECTED] wrote:
Hello,
patching v1.0.5 on my system removed the problem for me. But yes it seems
strange that this feature was inserted into a final release with very little
documentation of the wide implications that are caused by the
The real solution would be to fix the 'logical error' and not brake
the callerid in many situations. I thought it only affected the
manager interface, but it seems that its not only limited to the
manager as these thread and many bug reports point out.
Isn't it possibly to change caller id 'on
I saw the note about the o flag. But if I'm not mistaken we have to
add the o flag to get the old behavior back again instead of using
the o flag to get the new behavior. That means our existing (and
formerly functioning) dialplans will be broken.
It would make more sense IMHO to have a flag
I just upgraded my two asterisk boxes to 1.0.5 stable and I've noticed
that callerid is not functioning properly. My setup looks like this:
SIP Phone -- SER -- Asterisk -- Asterisk --- PSTN
No iax is being used at this time.
The problem can be best described by the following scenarios:
1.) SIP
Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Friday, February 04, 2005 11:24 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Callerid problems with 1.0.5
I just upgraded my two asterisk boxes to 1.0.5 stable and I've noticed
that callerid is not functioning
mattf wrote:
Also in CVS_HEAD preserving original callerid has been given a flag 'o' in
the dial string.
I have to wonder why the default behavior was changed to this
non-standard usage though; in what situations do we want the CLID/CNAM
of the _recipient_ to be passed to them?
. And in v1.0.5
you can patch your system to remove that feature.
MATT---
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Friday, February 04, 2005 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Callerid problems with 1.0.5
: Friday, February 04, 2005 11:12 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Callerid problems with 1.0.5
Supposedly when a call is parked and/or transferred they
wanted the callerid to reflect the person who is on that
phone call
Yikes!
On Feb 4, 2005, at 1:26 PM, Jay Milk wrote:
Can someone clarify what's going on here?
I'm running 1.0.5, and I see caller-id come through just fine from one
extension to the other, as well as for incoming and outgoing calls
(iax2). What are you folks seeing there?
The behavior that was
: Mark Eissler [mailto:[EMAIL PROTECTED]
Sent: Friday, February 04, 2005 9:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Callerid problems with 1.0.5
Yikes!
On Feb 4, 2005, at 1:26 PM, Jay Milk wrote:
Can someone clarify what's going on here?
I'm
Is there a way to get the callerid of the originating call to be sent
when the Cisco 7960 is doing a blind transfer to another 7960? I always
get the callerid of the 7960 doing the transfer.
Thanks,
Adam
___
Asterisk-Users mailing list
Hello all,
I'm having a problem with getting incoming callerid to a lan-connected
phone.
The Asterisk server is connected to the Internet, and a Grandstream
BT101 phone on a lan interface:
INTERNET (eth0) Asterisk (eth1) Grandstream (192.168.1.51)
The phone registers with the Asterisk
Hi,
My * (latest stable CVS) is not sending the caller id on its zap channels
(digium TDM40B). The callerid is shown on call-waiting, but is hidden if
the ringing channel is not already in a call.
The same * and configuration was working before upgrading to the latest
stable CVS.
Of course I
On Wed, 2005-01-05 at 17:02, PHP Mechanic wrote:
Howard Lowndes wrote:
Is there anyone using * in AU that has successfully extracted the CLID
from an incoming analogue PSTN phone call, and would like to spread the
word?
Yes, I have. I'm using a Voicetronix OpenLine4 card. I did have an
Is there anyone using * in AU that has successfully extracted the CLID
from an incoming analogue PSTN phone call, and would like to spread
the
word?
What I need more though is examples of anything that needs to go into
extensions.conf
You could add this line if you want
exten =
On Wed, 2005-01-05 at 16:50, James Andrewartha wrote:
Howard Lowndes wrote:
Is there anyone using * in AU that has successfully extracted the CLID
from an incoming analogue PSTN phone call, and would like to spread the
word?
Yes, I have. I'm using a Voicetronix OpenLine4 card. I did have
On Wed, 2005-01-05 at 23:22, PHP Mechanic wrote:
Is there anyone using * in AU that has successfully extracted the CLID
from an incoming analogue PSTN phone call, and would like to spread
the
word?
What I need more though is examples of anything that needs to go into
What I need more though is examples of anything that needs to go into
extensions.conf
You could add this line if you want
exten = s,1,NoOp(Caller ID on the PSTN line is ${CALLERID})
M. Tried that, but it didn't deliver ${CALLERID}
Did the caller have callerid enabled by their telco ?
On Wed, 2005-01-05 at 23:49, PHP Mechanic wrote:
What I need more though is examples of anything that needs to go into
extensions.conf
You could add this line if you want
exten = s,1,NoOp(Caller ID on the PSTN line is ${CALLERID})
M. Tried that, but it didn't deliver
On Thu, Jan 06, 2005 at 07:25:35AM +1100, Howard Lowndes arranged a set of bits
into the following:
On Wed, 2005-01-05 at 23:49, PHP Mechanic wrote:
What I need more though is examples of anything that needs to go into
extensions.conf
You could add this line if you want
exten =
On Thu, Jan 06, 2005 at 12:14:12PM +1100, Julien Goodwin wrote:
On Thu, Jan 06, 2005 at 07:25:35AM +1100, Howard Lowndes arranged a set of bits into the following:
Sure was. It was me calling myself from my mobile (cell) phone, and
that definitely has CLID enabled. In AU CLID is enabled by
Is there anyone using * in AU that has successfully extracted the CLID
from an incoming analogue PSTN phone call, and would like to spread the
word?
Note - I am only interested in analogue, not ISDN phones.
--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
Howard Lowndes wrote:
Is there anyone using * in AU that has successfully extracted the CLID
from an incoming analogue PSTN phone call, and would like to spread the
word?
Yes, I have. I'm using a Voicetronix OpenLine4 card. I did have an issue
with the config file, you have to set callerid=yes
Howard Lowndes wrote:
Is there anyone using * in AU that has successfully extracted the CLID
from an incoming analogue PSTN phone call, and would like to spread the
word?
Yes, I have. I'm using a Voicetronix OpenLine4 card. I did have an issue
with the config file, you have to set callerid=yes
On Wed, 5 Jan 2005 17:02:32 +1100, PHP Mechanic wrote:
Howard Lowndes wrote:
Is there anyone using * in AU that has successfully extracted the CLID
from an incoming analogue PSTN phone call, and would like to spread the
word?
Yes, I have. I'm using a Voicetronix OpenLine4 card. I did have
I fixed it using this:
http://www.voip-info.org/tiki-index.php?page=Asterisk+and+Australian+Caller+ID
Actually settings such as these should really be moved to a settings
header file.
Please elaborate...
___
Asterisk-Users mailing list
On Wed, 5 Jan 2005 17:55:50 +1100, PHP Mechanic wrote:
I fixed it using this:
http://www.voip-info.org/tiki-index.php?page=Asterisk+and+Australian+Caller+ID
Actually settings such as these should really be moved to a settings
header file.
Please elaborate...
Of the top of my head... I
Hi all,
I was wondering how the easiest way to restrict the users ability to set
caller ID would be ?
I have some users that uses IAX to connect with me. multiple numers via
iax.
on outgoing calls I would like the user to only be able to set his
range of numbers on the outgoing calls.
Is
of commercial PRIs you can set your caller ID
to anything you wish.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, December 30, 2004 8:11 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] callerid
Damon Estep wrote:
Use a separate context for the outbound calls for that customer,
check the caller ID in the dialplan before completing an outbound
call using a PATTERN MATCH, and IF the pattern does not match the
pattern of the customers numbers GOTO a step that sets the caller ID
to the
On Thu, 30 Dec 2004 [EMAIL PROTECTED] wrote:
I was wondering how the easiest way to restrict the users ability to set
caller ID would be ?
I have some users that uses IAX to connect with me. multiple numers via
iax.
on outgoing calls I would like the user to only be able to set his
Can you try to add some parameters to zapata.conf before defining channel 3,4
cidsignalling = dtmf
cidstart = polarity
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
I am using version: CVS-v1-0-12/13/04-18:46:23 with a
TDM400p (2 fxo, 2 fxs ports) and I keep getting errors
along with phantom calls:
Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363
ss_thread: Got event 17 (Polarity Reversal)...
Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434
ss_thread: CallerID
I am having a similar problem for my home setup. I receive a call from
the PSTN and have * automatically dial one or both Cisco 7940's running
SIP firmware. In my case the callerid info just says asterisk as the
PBX is actually placing the call on behalf of the incoming PSTN call on an
X100P
Is there a way to keep the incoming CallerID from the PSTN
and pass it onto the sip phone receiving the supervised call transfer?
The receptionist receives the PSTN callerID, performs a
supervised transfer, we get her local SIP callerID, not the original callers.
The main reason we
Hi,
I cannot see cid for incomming call from PSTN (Quintum gateway) to IAX
client (FireFly). Client displays blank but when I look into cdr's
/var/log/asterisk/cdr-cvs/Master.cvs, the callerid is registered
properly. Why it's not displaying?
L.
___
Heya
I have my * box connected to the Telkom PSTN, and an analogy line with callerID subscription (yes we get charged extra :).
When i call the line, it rings once, a short pause, and then the continued ringing of the phone. Using an external callerID device, it shows the number of the call
NeumannSent: Wednesday, December 01, 2004 4:42 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] CallerID
on X100P in South AfricaHeyaI have my * box
connected to the Telkom PSTN, and an analogy line with callerID subscription
(yes we get charged extra :).When i call the line, it rings
I would like to have CallerID reading from my outlook contacts. I could
export the Outlook thing to a csv file - does anyone have a script that can
read CallerID of it then?
thanks
Jens
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Are you waiting until the start of the second ring cycle before
answering the phone?
CLID information is sent in-band between the first and second ring
cycles. If you interrupt this process (by answering the phone before
transmission is complete), you will not receive the CLID information.
Hi Jim,
Thanks for your response.
I do wait after the second ring.
Regards,
-Ryan
On Thu, 2004-11-11 at 12:35, Jim Van Meggelen wrote:
Are you waiting until the start of the second ring cycle before
answering the phone?
CLID information is sent in-band between the first and second ring
Hi,
I'm having a problem with callerid. It is recieved fine by the fxo (it
appears in the cdr, and voicemail app gets it fine), but it is passed to
the internal phones works about 25% of the time.
The internal phones are all analog, a dvg-1120M (mgcp firmware) and a
quicknet phonejack.
There
Yes I am currently using a build with that patched. I think the
complicating factor here may be that I am using both callerID *and*
distinctive ring.
Thanks for the suggestion, but as this is still a problem please let me
know if you have any further! :-)
Kind regards,
Elliot.
Have you tried
]
Sent: Tuesday, 9 November 2004 6:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID+Distinctive ring in Australia
Whoops. Apologies for sending this more than once. I thought a sendmail
upgrade had broken, but it was just slow :-)
Someone
,
PaulH
Hawthorn
Aust
-Original Message-
From: Elliot Mackenzie [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 9 November 2004 6:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID+Distinctive ring in Australia
Whoops. Apologies for sending this more
: Elliot Mackenzie [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 9 November 2004 6:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID+Distinctive ring in Australia
Whoops. Apologies for sending this more than once. I thought a sendmail
upgrade had broken
Hi,
- Original Message -
From: Dan [EMAIL PROTECTED]
I have upgraded the Asterisk Server after several months and now there is
an issue with the CallerID Name information.
When I call from DIAX (IAX2) to ATA186(SIP) I get the correct CallerID
name/number.
When I call from ATA186(SIP) to
I have a situation involving both caller id and distinctive ring in
australia that appears to be having issues. I am using the CVS snapshot
current as of an hour ago.
The distinctive ring was working ok until i arranged for the telco to
turn on callerid: now the distinctive ring detection
I have a situation involving both caller id and distinctive ring in
australia that appears to be having issues. I am using the CVS snapshot
current as of an hour ago.
The distinctive ring was working ok until i arranged for the telco to
turn on callerid: now the distinctive ring detection
Whoops. Apologies for sending this more than once. I thought a
sendmail upgrade had broken, but it was just slow :-)
Someone mentioned there are some patches required to make callerid work
in the UK. Do similar patches apply for use in Australia? How would I
make my own modifications? Is
Elliot Mackenzie wrote:
I have a situation involving both caller id and distinctive ring in
australia that appears to be having issues. I am using the CVS snapshot
current as of an hour ago.
The distinctive ring was working ok until i arranged for the telco to
turn on callerid: now the
Hi,
I have upgraded the Asterisk Server after several months and now there is an
issue with the CallerID Name information.
When I call from DIAX (IAX2) to ATA186(SIP) I get the correct CallerID
name/number.
When I call from ATA186(SIP) to DIAX(IAX2) I get the correct number, but the
CallerID
Dan wrote:
Hi,
I have upgraded the Asterisk Server after several months and now there
is an issue with the CallerID Name information.
When I call from DIAX (IAX2) to ATA186(SIP) I get the correct CallerID
name/number.
When I call from ATA186(SIP) to DIAX(IAX2) I get the correct number, but
the
Hi,
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
I have upgraded the Asterisk Server after several months and now there is
an issue with the CallerID Name information.
When I call from DIAX (IAX2) to ATA186(SIP) I get the correct CallerID
name/number.
When I call from
I have been using Asterisk 1.0-RC2 successfully with a channelized T1
circuit for quite a while now but after upgrading to 1.0.0 callerid no
longer works properly.
Debug output from a channel shows what actually is received through DTMF
from the carrier:
[ TYPE: DTMF (1) SUBCLASS: * (42) ]
, September 20, 2004 10:37 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] CallerID in Queue
How can I bring the Caller ID when the calls enter call queue and answer
by X- lite or kphone?
I've tried many configuration but no luck that it only shows the
AgentLogin's exten..
Thanks!
R Wong
How can I bring the Caller ID when the calls enter call queue and answer by X-
lite or kphone?
I've tried many configuration but no luck that it only shows the AgentLogin's
exten..
Thanks!
R Wong
The information transmitted is intended only for the person or entity to which it is
addressed
Hi,
need a quick help ... it should be easy but ...
exten =_9898,1,Answer
exten =_9898,2,VoiceMailMain([EMAIL PROTECTED])
Accepting overlap call from '342' to '9' on channel 0/2, span 3
-- Executing Answer(Zap/8-1, ) in new stack
-- Executing VoiceMailMain(Zap/8-1, @domain) in new
[EMAIL PROTECTED] wrote:
Accepting overlap call from '342' to '9' on channel 0/2, span 3
-- Executing Answer(Zap/8-1, ) in new stack
-- Executing VoiceMailMain(Zap/8-1, @domain) in new stack
As you can see there variable CALLERID is empty, why ?
Sending a question again doesn't mean
Hello Andreas,
Thursday, September 2, 2004, 2:28:33 PM, you wrote:
AS [EMAIL PROTECTED] wrote:
Accepting overlap call from '342' to '9' on channel 0/2, span 3
-- Executing Answer(Zap/8-1, ) in new stack
-- Executing VoiceMailMain(Zap/8-1, @domain) in new stack
As you can see there
In my zapata.conf, I have
callerid=unknown
so if an incoming call doesn't set or suppresses it's callerid then my phone
will show unknown. I have found that if the callerid on the incoming call
is suppressed, then the call goes straight to Voicemail.
Has anyone seen this problem?
Simon Brown
Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown
Sent: Friday, 20 August 2004 13:51
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] CallerId
In my zapata.conf, I have
callerid=unknown
so if an incoming call doesn't set or suppresses
Simon Brown wrote:
In my zapata.conf, I have
callerid=unknown
That doesn't look right to me. Try:
callerid=Unknown
Cheers
Darryl
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email: [EMAIL PROTECTED]
Phone: 08 81228363
Office: 08 81226361
If you want to live up to the whole There is more
Hi all,
I've been trying to wrap my mind around this one for several days now.
How can I 'debug' the CallerID reception on a Zap/POTS channel? I have
a POTS line with CallerID and a Digium TDM11B card right now. I have my
signalling set to ks for both sides, can make and receive calls just
Christopher L. Wade wrote:
Hi all,
I've been trying to wrap my mind around this one for several days now.
How can I 'debug' the CallerID reception on a Zap/POTS channel? I
have a POTS line with CallerID and a Digium TDM11B card right now. I
have my signalling set to ks for both sides, can
Soren Rathje wrote:
Try immediate=no in context=external. FSK type callerid is usually received after the first ring and before the second ring.
Have already tried that, the only thing it does is make the messages
slightly less random. I end up getting more of the notice messages than
I do the
Hello All,
Does anyone tried to use CallerID in Eastern Europe (Russia/Ukraine)?
Our teleco provides CallerID, as well as AON, then can send _callerid_, as well as AON
signals non of those 2 works on TDM400P card with FXS ports.
They are using Siemence systems.
How can I debug this and
I want to send date information that comes in PSTN-Caller*ID from
Asterisk to a H.323-PSTN gateway (an AudioCodes one) on the other
side, but Caller *ID in the phone shows
00/00/00 12:00 a.m.
36.
That is No ANI, no date.
I've found a solution to ANI problem (perhaps patch h323 channel)
: [Asterisk-Users] Callerid via PRI
When receiving multiple calling numbers via a PRI for a call setup, I
cannot find the ability to select between either first or last. Is
there a way to do this currently? If not, can anyone help me in
getting
this to work? Here is the dump of the PRI
We are receiving phone-calls at one Asterisk box, it forwards them to a
different location over IAX. Calls are answered on the other end of this
connection.
However, the callerID seems to be lost in the process.
The 1st box shows the original callerID, on the 2nd box callerID shows
the
On Wed, 14 Apr 2004, Markus Mayer wrote:
The 1st box shows the original callerID, on the 2nd box callerID shows
the callerID of the 1st box. Apparently the 1st Asterisk box replaces
the original callerID with its own.
Try this one:
Hi,
bristuff 0.0.2rc20 will add support for HOLD/RETRIEVE, SUSPEND/RESUME
and isdn transfers in an experimental way.
It also features a zaptel that works on 2.6 (and does not freeze),
together with optimized qozap drivers. Load tests have shown that it
is possible to have 6 quadBRI cards in a
Well, Once upon a time, I had problems receiving callerid, and then one
day, Mark was logged into my asterisk box helping with something else,
and I asked him about this, and he showed me a nice tweak to some source
file that made it work.
Some time later, I must have done hundreds of CVS
Adam Goryachev wrote:
I am in Australia, which I think expects callerid at a different time to
other countries Although other people have told me callerid is
working correctly for them
From what I've been able to guess at Telstra sends a short ~50ms chirp
to the phone, the caller id and
Adam Goryachev wrote:
I am in Australia, which I think expects callerid at a different time to
other countries Although other people have told me callerid is
working correctly for them
Forgot to mention there is a patch for this, but it won't patch cleanly
against current CVS...
G'day Adam,
This drove me nuts for a few days just recently (only fixed it yesterday
in fact, and I've not had a chance to update any doco anywhere yet).
On Wed, 14 Apr 2004, Adam Goryachev wrote:
Actually, now that I look at the file again, I can also see:
Line: 80
/* Typically, how many
Duane wrote:
Adam Goryachev wrote:
I am in Australia, which I think expects callerid at a different time to
other countries Although other people have told me callerid is
working correctly for them
Forgot to mention there is a patch for this, but it won't patch
cleanly against current
they actually send the caller-id info after the SECOND ring.
Now of course if the au indications were changed to combine the first
and second ring to appear as one ring, no other changes would be needed
??
Gary
On Wed, 14 Apr 2004 00:36:16 +1000, Duane wrote:
Adam Goryachev wrote:
I am in
On Wed, 2004-04-14 at 00:52, Vic Cross wrote:
G'day Adam,
This drove me nuts for a few days just recently (only fixed it yesterday
in fact, and I've not had a chance to update any doco anywhere yet).
On Wed, 14 Apr 2004, Adam Goryachev wrote:
Actually, now that I look at the file
On Wed, 14 Apr 2004, Gary wrote:
they actually send the caller-id info after the SECOND ring.
That depends. ;) If you define ring as 'single burst of ring voltage',
then it does come after the second ring. That's how * will have to look
for it, after all.
It changes if you have Distinctive
On Thu, 2004-04-08 at 08:54, Martin Schenkelberg wrote:
Thank you problem solved.
I tried to use the (R) Button on my phone to place call on HOLD but Asterisk
says something of PRI Error : Dont know how to post-handle message of Tye
HOLD (36)
Is this feature not implemented in Bri-Stuff
Thank you problem solved.
I tried to use the (R) Button on my phone to place call on HOLD but Asterisk
says something of PRI Error : Dont know how to post-handle message of Tye
HOLD (36)
Is this feature not implemented in Bri-Stuff ?
Thanks again
Am Mittwoch, 7. April 2004 16:54 schrieb
Hi all,
i have an ISDN Phone connected to an HFC-S based card, all works fine but is i
call the Phone from a SIP User Agent or over PSTN Line the Phones Display
shows the correct CallerID but with a leading 0 .
I cant find this in the config files, how can is solve this?
Dialing Out with the
Hi,
use prilocaldialplan=local in zapata.conf.
--
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
On Mon, 5 Apr 2004, AstGrp waxed:
I am having an issue with Callerid (INBOUND). I have a system set up
with 4 companies sitting behind the system. On all of the companies
except of one of them, it displays callerid withh 'asterisk'. The other
company displays the callerid of the person
Resolved the issue... It turned out to be a problem with the ISP
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Posted At: Monday, April 05, 2004 2:21 PM
Posted To: Asterisk User Group
Conversation: CallerID
Subject: [Asterisk-Users] CallerID
I
, 2004 1:41 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] CallerID
Subject: Re: [Asterisk-Users] CallerID
On Mon, 5 Apr 2004, AstGrp waxed:
I am having an issue with Callerid (INBOUND). I have a system set up
with 4 companies sitting behind the system. On all of the companies
I am having an issue with Callerid (INBOUND). I have a system set up
with 4 companies sitting behind the system. On all of the companies
except of one of them, it displays callerid withh 'asterisk'. The other
company displays the callerid of the person calling.
Zapata.conf
[channels]
Hi all,
i have a TDM20B in my astbox and i have configured my channels as follows:
usecallerid=yes
signalling=fxo_ks
context=tel1
group=5
callerid=101
channel = 13
callerid=102
channel = 14
But if i make a connection to the manager interface the callerid in the
events is not set:
Event:
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