I agree with Freddi and would like to add that a field indicating the
order of the outgoing legs would be very useful. For billing purposes
one could benefit very much if one new the order of the providers
that were called in a specific call.
Freddi Hansen wrote:
To me the obvious answer is to
On Tue, Nov 25, 2008 at 10:26:43PM -0600, [EMAIL PROTECTED] wrote:
Greetings List
I have connected my asterisk box with x100 2xfxo and xorcom 8xfxo and all
of them give me the error Ring/Off-hook in strange state 6.
DAHDI? Zaptel? What version?
Whenever the caller hangup, the call
Hello,
I've seen Domino 7 supports ACL and IMAP.
Have you heard of experiences in which Lotus Notes/Domino users could read
and manage voicemails recorded by Asterisk ?
In other words, is it possible with Domino to dedicate to Asterisk an
account with which, using IMAP, Asterisk could drop or
jhon digital21 wrote:
same result
I never saw the original message, what version of Asterisk and what
country are you in? Does it work for outbout okay?
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither
Hello, everyone.
Anybody know when that svn will be available again?
Regards
*Alex Montoanelli*
Administração e Gerência de Redes
Unetvale Conectividade http://www.unetvale.net
+55 48 3263 8700
___
-- Bandwidth and Colocation Provided by
Jerry Geis wrote:
/ I upgraded from 1.2 to 1.4.18
//
// After upgrading I get half channel audio on SOME phones.
//
// I have Cisco 7960 that works, I have a wireless polycom 8002 phone that
// works.
// However, my polycom 501's are getting half channel audio on EXTERNAL
calls.
//
Do you use the Asterisk GUI? Changes from it can mess with contexts in the
dialplan (extensions.conf) and the hints need to remain in the [internal]
context.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Wednesday, November 26, 2008 6:33 AM
To: 'Asterisk
Greetings List,
I have configured chan_mob for Nokia 7610. I can succefully dial from
softphone to mobile and land line numbers,
Softphone (PC) = Asterisk FXO (Nokia 7610) Destination
Number
When call is established I have to use Nokia 7610 for conversation. Is it
A little less whitespace please.
If I understand your question correctly, yes you can.
On Wed, Nov 26, 2008 at 9:10 AM, Irfan Malik [EMAIL PROTECTED] wrote:
Greetings List,
I have configured chan_mob for Nokia 7610. I can succefully dial from
softphone to mobile and land line numbers,
Hello,
I've had Asterisk and Polycom phones work perfectly with hints for the last
6 months. Suddently, I realize they've stopped working in the last few
days. I haven't changed the configuration in any way.
I have hints setup (CLI show hints does show the hints, and they seem
correct).
On Wed, Nov 26, 2008 at 11:47 AM, Richard Brady [EMAIL PROTECTED] wrote:
Hi folks
I'm not sure what I am missing but I cannot find a predefined channel
variable to identify the SIP peer/user which has initiated a call and
established the channel.
The one option is to extract it from the
On Wed, Nov 26, 2008 at 1:32 PM, Michiel van Baak [EMAIL PROTECTED] wrote:
On 09:06, Wed 26 Nov 08, Alex Montoanelli wrote:
Hello, everyone.
Anybody know when that svn will be available again?
Regards
Hey,
I can checkout stuff fine from svn.digium.com.
Maybe you can provide some more
Hello,
Is it possible, for testing, to connect an cat5 straight patch cord between
2 ports of a Digium B410P card and use these 2 ports as a normal dahdi trunk
?
I've tried this:
One port is set as NT, the other as TE.
I would expect timing to come for system hardware so I choose in
Hello,
I have created a dynamic conference into two languages (english and
russian). Client calls to confrence number and interactive choose the
language. Meetme runs with 'dMi' options. Everything works perfect if one
conference room clients have choosed the same language. If clients had
Just speaking theoretically, you should be able to do a Zap/SIP bridge just
like using a TDM???. How does this show up in the CLI interface (core show
channels)?
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Irfan Malik
Sent: Wednesday, November 26, 2008 8:11 AM
How? Any hint?
Regards,
Irfan Malik
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, November 26, 2008 7:15 PM
Assuming you have caller id, you can call MeetMe with different parameters.
You could also write an AGI to handle the announcements and leave meetme in
Silent (No Announce) mode.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giedrius Augys
Sent: Wednesday, November
What have you tried?
On Wed, Nov 26, 2008 at 9:25 AM, Irfan Malik [EMAIL PROTECTED] wrote:
How? Any hint?
Regards,
Irfan Malik
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL
What are the lines in your dialplan for using the Mobile line? For example
exten = NXX,1,Dial(Zap/g1/${EXTEN},60)
dials a local (7 digit) number using Zap Group 1, waiting 60 seconds for
connection.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Here is the output during call,
localhost*CLI core show channels
Channel Location State Application(Data)
Mobile/Nokia-7610-e2 [EMAIL PROTECTED]:1 Ringing AppDial((Outgoing Line))
SIP/2001-09960968[EMAIL PROTECTED]:1 Ring
Dial(Mobile/Nokia-7610/0321609
2
These are the lines from my extension.conf
[phones]
; context for our phones
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 500,1,Answer()
exten = 500,2,Playback(demo-echotest) exten = 500,3,Echo
exten = 500,4,Playback(demo-echodone)
exten = 500,5,Hangup
exten =
Doug Lytle wrote:
jhon digital21 wrote:
same result
country are you in? Does it work for outbout okay?
That should have read 'outbound', that's what happens when you reply
when you're late for work.
Doug
--
Ben Franklin quote:
Those who would give up Essential
Hi Andrew and all those following this thread;
I have gotten it working like it was meant to work see my original post
quoted below. I have also included the direct link to my post...
My Original Post:
http://lists.digium.com/pipermail/asterisk-users/2008-November/222339.ht
ml
Quote:
I would try this:
exten = _.,1,Dial(Mobile/Nokia-7610/${EXTEN},60,KkTt) ; dials using mobile
nokia
7610
This should make the call Bridgeable/Transferrable.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Irfan Malik
Sent: Wednesday, November 26, 2008
Olivier wrote:
One port is set as NT, the other as TE.
I would expect timing to come for system hardware so I choose in
/etc/dahdi/system.conf :
span=1,0,0,ccs,ami
span=2,0,0,ccs,ami
What is your configuration in chan_dahdi.conf?
--
Kevin P. Fleming
Director of Software Technologies
Asterisk version : 1.4
country : France
outbound : not tested
2008/11/26 Doug Lytle [EMAIL PROTECTED]
Doug Lytle wrote:
jhon digital21 wrote:
same result
country are you in? Does it work for outbout okay?
That should have read 'outbound', that's what happens when you
On 09:06, Wed 26 Nov 08, Alex Montoanelli wrote:
Hello, everyone.
Anybody know when that svn will be available again?
Regards
Hey,
I can checkout stuff fine from svn.digium.com.
Maybe you can provide some more info about how it's not working for you.
--
Michiel van Baak
[EMAIL PROTECTED]
2008/11/26 Danny Nicholas [EMAIL PROTECTED]
Assuming you have caller id, you can call MeetMe with different
parameters. You could also write an AGI to handle the announcements and
leave meetme in Silent (No Announce) mode.
--
*From:* [EMAIL PROTECTED]
Hi folks
I'm not sure what I am missing but I cannot find a predefined channel
variable to identify the SIP peer/user which has initiated a call and
established the channel.
The one option is to extract it from the CHANNEL variable, but that is
fraught with difficulties.
Is there another
These are lines from my extensions.conf
[phones]
; context for our phones
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 500,1,Answer()
exten = 500,2,Playback(demo-echotest)
exten = 500,3,Echo
exten = 500,4,Playback(demo-echodone)
exten =
Hi,
From your answer, shall I understand it is possible to loop for one port
back to another ?
Anyway, chan_dahdi.conf :
[channels]
language=fr
context=isdntrunk
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
internationalprefix=00
nationalprefix=0
usecallerid=yes
Ok. You will need to modify meetme.c to allow a prompt for language as well
as name. Based on the prompt, you will provide the chosen language to the
asterisk say prompts in the routine.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giedrius Augys
Sent: Wednesday,
I was trying a 'svn ls http://svn.digium.com/svn/', and was receiving a 403
- Forbiden.
But a rising level could access the content.
Thank you and hugs
Regards
*Alex Montoanelli*
On Wed, Nov 26, 2008 at 12:17 PM, Atis Lezdins [EMAIL PROTECTED] wrote:
On Wed, Nov 26, 2008 at 1:32 PM,
For me, the Polycom loses its subscription when asterisk is restarted.
However, as long as the phone is restarted after asterisk, everything works
fine. Worth a look. (I'm running a Polycom 500, so my firmware is older than
yours.)
___
-- Bandwidth and
Hi there,
I've got the following code (for remote enquiry of the answering machine) in
my dialplan:
[mailbox]
exten = m,1,Set(TIMEOUT(digit)=4)
exten = m,2,Set(TIMEOUT(response)=0)
exten = m,3,Set(LANGUAGE()=de)
exten = m,4,Read(Pin,unavail,4)
exten = m,5,capicommand(echosquelch|no)
exten =
The phone should renew itself to asterisk periodically even after a
reboot. My setup renews the connection every 2 minutes (non-critical,
small shop).
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen
Sent: Wednesday, November 26, 2008 9:24 AM
To:
2008/11/26 Danny Nicholas [EMAIL PROTECTED]
Ok. You will need to modify meetme.c to allow a prompt for language as
well as name. Based on the prompt, you will provide the chosen language to
the asterisk say prompts in the routine.
--
*From:* [EMAIL
You can compile this code into Asterisk 1.4 to give you the ability to write
custom data for up to 20 fields. The field names in the code must match the
field names in the cdr db table. ENJOY
Dave
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2003-2005, Digium,
On Wed, Nov 26, 2008 at 04:12:58PM +0100, Olivier wrote:
Hi,
From your answer, shall I understand it is possible to loop for one port
back to another ?
/etc/dahdi/system.conf :
span=1,0,0,ccs,ami
span=2,0,0,ccs,ami
Hmm... which of those two should provide timing?
I suppose you should
I tried to add app_rxfax/app_txfax to a running asterisk 1.6.0. on Lenny.
I followed http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 :
apt-get install libspandsp1
cd /usr/src/asterisk-1.6.0.1
./configure
make clean
make menuselect
In menuselect/application menu, I can see that
Not at all, I do everything with vi
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas
Sent: Wednesday, November 26, 2008 8:51
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Hints stopped working suddently
Do you use
Good theory, but I had already tried that (and my phone re-subscribes every
60 seconds anyways)
so that's not it.
Regards,
Mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen
Sent: Wednesday, November 26, 2008 10:24
To: Asterisk Users Mailing List -
Have you tried doing core show hints and sip show peers before and after
asterisk restart to see what if anything changes?
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Wednesday, November 26, 2008 10:11 AM
To: 'Asterisk Users Mailing List -
Hi,
I'm having trouble getting asterisk to report MWI to a Cisco CCME.
I record a message in mailbox 29, but the subsequent MWI notifications
I see continue to report no messages waiting. Are they reporting for
the wrong mailbox? Is there some other option I have to set or change?
I'm running
Yes I did. Nothing changes, really. And it all looks good.
What I don't get is why the status unavailable appears when the phone is
disconnected, but the status inuse doesn't when on a call. That
unavailable works fine is some sort of proof that everything is setup
properly
Mike
2008/11/26 Tzafrir Cohen [EMAIL PROTECTED]
On Wed, Nov 26, 2008 at 04:12:58PM +0100, Olivier wrote:
Hi,
From your answer, shall I understand it is possible to loop for one port
back to another ?
/etc/dahdi/system.conf :
span=1,0,0,ccs,ami
span=2,0,0,ccs,ami
Hmm... which of
On Wednesday 26 November 2008 09:48:40 David Budny wrote:
You can compile this code into Asterisk 1.4 to give you the ability to
write custom data for up to 20 fields. The field names in the code must
match the field names in the cdr db table. ENJOY
Or you could just use the
Olivier wrote:
(As for the question of wiring: I have no idea. Refer to the
documentation or to the answers of others in this thread)
b410 manual says pins are affected this way :
3 Tx+ (TE) Rx+ (NT)
4 Rx+(TE)Tx+ (NT)
5 Rx-(TE)Tx- (NT)
6 Tx- (TE) Rx- (NT)
jhon digital21 wrote:
Asterisk version : 1.4
country : France
outbound : not tested
Someone else may need to chime in here, I'm in the US.
But, when I was doing analog (PRI all the way around now), I used to use
ztmonitor to measure in inbound/outbound volume. You may want to try
that
I am having an issue with a Rhino channelbank connected to a Digium
TE411P card. The server has 3 E1 R2 links and the fourth port is used
to connect a Rhino FXO channelbank with 12 lines. The first four ports
on the rhino are GSM adapters. From time to time I can see the channels
On Wed, Nov 26, 2008 at 05:02:16PM +0100, Olivier wrote:
I tried to add app_rxfax/app_txfax to a running asterisk 1.6.0. on Lenny.
I followed http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 :
apt-get install libspandsp1
cd /usr/src/asterisk-1.6.0.1
./configure
make clean
make
The problem is that IAX2 does not seem to support call pickup.
Bruno Castelo Branco wrote:
hi
I'm using only IAX extensions and inserted callgroup=1 and callpickup=1
for all IAX extensions in iax.conf. Didn't works for while.
thanks
Tim Panton wrote:
I think it doesn't work across
Tzafrir Cohen schrieb:
On Wed, Nov 26, 2008 at 05:02:16PM +0100, Olivier wrote:
I tried to add app_rxfax/app_txfax to a running asterisk 1.6.0. on Lenny.
I followed http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 :
apt-get install libspandsp1
cd /usr/src/asterisk-1.6.0.1
On Wed, Nov 26, 2008 at 07:16:05PM +0100, Philipp Kempgen wrote:
Tzafrir Cohen schrieb:
On Wed, Nov 26, 2008 at 05:02:16PM +0100, Olivier wrote:
I tried to add app_rxfax/app_txfax to a running asterisk 1.6.0. on Lenny.
I followed http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38
2008/11/26 Kevin P. Fleming [EMAIL PROTECTED]
Olivier wrote:
(As for the question of wiring: I have no idea. Refer to the
documentation or to the answers of others in this thread)
b410 manual says pins are affected this way :
3 Tx+ (TE) Rx+ (NT)
4 Rx+(TE)Tx+
2008/11/26 Philipp Kempgen [EMAIL PROTECTED]
Yup. libspandsp is a build-dependency (in contrast to a normal
runtime dependency). Thus you need the -dev package.
Philipp Kempgen
I was not aware of such build-dependency packages.
Does that mean I could remove build-dependency packages once
2008/11/26 Tzafrir Cohen [EMAIL PROTECTED]
aptitude install libspandsp-dev
It did it !
Thanks, very much.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
Tzafrir Cohen schrieb:
On Wed, Nov 26, 2008 at 07:16:05PM +0100, Philipp Kempgen wrote:
Tzafrir Cohen schrieb:
On Wed, Nov 26, 2008 at 05:02:16PM +0100, Olivier wrote:
I tried to add app_rxfax/app_txfax to a running asterisk 1.6.0. on Lenny.
I followed
Olivier schrieb:
2008/11/26 Philipp Kempgen [EMAIL PROTECTED]
Yup. libspandsp is a build-dependency (in contrast to a normal
runtime dependency). Thus you need the -dev package.
I was not aware of such build-dependency packages.
Does that mean I could remove build-dependency packages once
BTW (sorry for hijacking the thread):
What package satisfies the dependency on resample on Debian
Lenny?
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan
In sip.conf do you have [EMAIL PROTECTED]
Lincoln
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Mark G. Thomas [EMAIL
PROTECTED]
Sent: Wednesday, November 26, 2008 11:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] sip MWI
OK. I know it's been a few weeks since my original post. Things have been busy
;-)
Based on help from the trixbox forums and the asterisk-users mailing list, I
have located 30 asterisk core dump files in /tmp. These date from 10/30/08 to
10/24/08. Today is 10/26/08. So this does agree with the
On Wed, Nov 26, 2008 at 08:33:25PM +0100, Philipp Kempgen wrote:
BTW (sorry for hijacking the thread):
What package satisfies the dependency on resample on Debian
Lenny?
It's not yet packaged.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406
On Wed, Nov 26, 2008 at 08:12:51PM +0100, Philipp Kempgen wrote:
Tzafrir Cohen schrieb:
On Wed, Nov 26, 2008 at 07:16:05PM +0100, Philipp Kempgen wrote:
Tzafrir Cohen schrieb:
On Wed, Nov 26, 2008 at 05:02:16PM +0100, Olivier wrote:
I tried to add app_rxfax/app_txfax to a running
On Wed, Nov 26, 2008 at 12:59:49PM -0700, Douglas Mortensen wrote:
OK. I know it's been a few weeks since my original post. Things have been
busy ;-)
Based on help from the trixbox forums and the asterisk-users mailing list, I
have located 30 asterisk core dump files in /tmp. These date from
Hi all,
I have a question regarding connection of two Asterisk servers to our PBX. Each
Asterisk server has one PCI E1 card, and they are in failover mode with Linux
HA. On our PBX we have only one E1 card towards Asterisk servers.
My question is how to connect these two Asterisks to one E1
Redfone
On Wed, Nov 26, 2008 at 4:04 PM, dubravko caric
[EMAIL PROTECTED] wrote:
Hi all,
I have a question regarding connection of two Asterisk servers to our PBX.
Each Asterisk server has one PCI E1 card, and they are in failover mode with
Linux HA. On our PBX we have only one E1 card
Pessoal,
Me ocorreu uma dúvida:
Imagine que tenho uma rede com um IP válido e um router compartilhando essa
internet para 3 micros.
Eu gostaria de colocar 3 ramais softphone nessas 3 máquinas cujo servidor
fica fora da rede.
Como fica a configuração do meu roteador para que isso funcione
Olivier wrote:
Doc says it should be turned on in those instances where a BRI is
daisy-chained and terminated on the B410P in NT mode.
So after reading this again, I would I should have turned it on.
Do you agree ?
B410P card1 - port1 - TE mode -- B410P card2 - port1 - NT mode
Tzafrir Cohen wrote:
On Wed, Nov 26, 2008 at 08:33:25PM +0100, Philipp Kempgen wrote:
BTW (sorry for hijacking the thread):
What package satisfies the dependency on resample on Debian
Lenny?
It's not yet packaged.
Right, 'it' is Digium's redistribution of some resampling code, and it
is
Hi,
I'm trying to get HANGUPCAUSE on my cdr the problem I'm facing is that this
option:
endbeforehexten=yes
is not working at least on asterisk 1.6.0.1, so if I put yes o no I cant set
CDR value with that value. It seems to finish the CDR record before h is
executed.
I'm using
Kevin P. Fleming schrieb:
Tzafrir Cohen wrote:
On Wed, Nov 26, 2008 at 08:33:25PM +0100, Philipp Kempgen wrote:
BTW (sorry for hijacking the thread):
What package satisfies the dependency on resample on Debian
Lenny?
It's not yet packaged.
Right, 'it' is Digium's redistribution of some
2008/11/26 Kevin P. Fleming [EMAIL PROTECTED]
Olivier wrote:
Doc says it should be turned on in those instances where a BRI is
daisy-chained and terminated on the B410P in NT mode.
So after reading this again, I would I should have turned it on.
Do you agree ?
B410P card1 - port1
2008/11/26 Kevin P. Fleming [EMAIL PROTECTED]
snip
However, chan_dahdi + wcb4xxp + libpri do not currently support NT
point-to-multipoint mode anyway, so this configuration cannot work with
the current code.
My (original) question was :
Shall I turn NT 100 ohm termination when directly
Olivier wrote:
normally if there are 2 devices you want termination on on both, when
there are more than 2 in a chain, the one at each end gets terminated,
not the middle ones.
if its really a star, not a chain electrically - experiment a bit
depends on the lengths of each arm what is best.
Jon Pounder wrote:
on real short lengths of wire (all in the same room) probably won't make
a difference either way.
Agreed. All our lab testing of cards (short cables) does not involve
enabling termination and it works fine.
--
Kevin P. Fleming
Director of Software Technologies
Digium,
On Wednesday 26 November 2008 13:59:49 Douglas Mortensen wrote:
OK. I know it's been a few weeks since my original post. Things have been
busy ;-) Based on help from the trixbox forums and the asterisk-users
mailing list, I have located 30 asterisk core dump files in /tmp. These
date from
On Wed, Nov 26, 2008 at 04:20:44PM -0600, Tilghman Lesher wrote:
On Wednesday 26 November 2008 13:59:49 Douglas Mortensen wrote:
OK. I know it's been a few weeks since my original post. Things have been
busy ;-) Based on help from the trixbox forums and the asterisk-users
mailing list, I
dubravko caric wrote:
Hi all,
I have a question regarding connection of two Asterisk servers to our
PBX. Each Asterisk server has one PCI E1 card, and they are in failover
mode with Linux HA. On our PBX we have only one E1 card towards Asterisk
servers.
My question is how to connect
Is there some kind of splitter which, on one side can accept two E1
connections from Asterisks and on the other side one E1 link from PBX. This
splitter must also recognize towards which one of two E1 links on Asterisk
side it should send signals to. eg. when primary Asterisk fails this
Somebody know some work around for it?
I still trying to find a solution but nothing seems to work
thanks
Eric ManxPower Wieling wrote:
The problem is that IAX2 does not seem to support call pickup.
Bruno Castelo Branco wrote:
hi
I'm using only IAX extensions and inserted callgroup=1 and
Versions
- Asterisk 1.4.22
- DAHDI Linux 2.0.0
- DAHDI Tools 2.0.0
- Libpri 1.4.7
- Addons 1.4.7
Here is chan_dahdi.conf
;
; DAHDI telephony interface
[trunkgroups]
[channels]
context=from-pstn
switchtype=national
signalling=fxo_ks
rxwink=300
hidecallerid=no
callwaiting=yes
Hello,
I am looking for a softphone which supports RPID (displaying the called
party name) and BLF features. I couldn't find one so far...
Any idea whether such a softphone exists?
Thanks! __Yehavi:
___
-- Bandwidth
Hello
I asked the same thing some time ago, but nobody answered.
I founded some workaround.
Use this in your dialplan:
exten = _7.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:1})
exten = _7.,n,Pickup(${EXTEN:[EMAIL PROTECTED])
This worked for me.
Cosmin
--- On Thu, 11/27/08, Bruno Castelo Branco
Hi Steve,
yes I know about RedFone, in fact I'm already using it on three
locations. now I'm looking for similar solution but with PCI cards.
Thanks
/davor
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
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