I have LibPri installed and working (.../wPRI).
So, if I understand Tzafrir correctly - DAHDI support for the B410P isn't
available in 1.4 at all.
Looks like I'm going back to mISDN.
Cheers
Andy
-- -Original Message-
-- From:
You could always run a shoutcast server and stream from that.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: 09 March 2009 19:02
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Hi All,
We have working pbx with asterisk 1.4.23.1
System: Centos 5.2
We are using polycom phones for pbx.
We are using sip channels for calls and all the users has set canreinvite=no
and nat=yes.
We have a issue with resuming the hold call by the polycom phone when the
call traffic is high.
We
Tilghman Lesher wrote:
On Monday 09 March 2009 01:28:49 pm Anthony Francis wrote:
Tilghman Lesher wrote:
On Friday 06 March 2009 11:24:46 pm Hooman Peiro wrote:
hi,
I'm working with asterisk on a project and I found a problem with
cdr_odbc. As we know, after answering each call
Hi, I have modified in Mobile/Setting the parameter SIP From from
tel/user to tel/tel and now I view the correct incoming number.
Thanks.
--
Salvatore.
- Original Message -
From: Christian Victor christ...@victormedia.de
To: Asterisk Users Mailing List - Non-Commercial
On Tue, Mar 10, 2009 at 4:01 AM, James Sneeringer jsnee...@gmail.com wrote:
On Mon, Mar 9, 2009 at 5:44 PM, nik600 nik...@gmail.com wrote:
Thanks, i've tested and it works (1.4.23.1).
Just 2 questions:
1) this approach seems to be an hack and not the implementation of a
feature is it really
fateme fatah wrote:
thank you Dear doug
But,I don't have any file in /var/spool/hylafax/log directory.
Sorry, I didn't see that you installed from RPM. To find out where FC
installs the files to, do the following:
updatedb -e /home (Updates the 'locate' database and excludes the home
Hi all!
On outgoing call sometimes Asterisk use/give back the caller id sent back by
called number instead of number called by me. This is annoying and
misleading statistics if other side use some exotic number. For example I
have called number 12345678 and CDR include the number 333 as callerid
thanks for your responses, I checked again and I found that I asked a wrong
question! I was supposed to ask about answer time. The answer time is not
getting save in the database.
On Tue, Mar 10, 2009 at 11:41 AM, Anthony Francis antho...@rockynet.comwrote:
Tilghman Lesher wrote:
On Monday
Hi All,
Is there a way that I can include call dialing functionality in a
webinterface. I have EyeBeam configured with a SIP user say
8440. Will I be able to design an inteface which agent can choose a number
and the Dial without punching in the number in
Eyebeam.
I tried using the .call file. ie
Hi,
When trying to cluster Asterisk boxes to gain scalability and more
performance, there is now a new simple and efficient solution for doing it.
OpenSIPS/OpenSER 1.5 can now implement traffic routing based on load.
Shortly, when OpenSIPS routes calls to a set of destinations, it is able
to
Great Job Bogdan
On Tue, Mar 10, 2009 at 12:52 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi,
When trying to cluster Asterisk boxes to gain scalability and more
performance, there is now a new simple and efficient solution for doing it.
OpenSIPS/OpenSER 1.5 can now implement
Hi,
It's not clear to me which asterisk version drops support for zaptel in favor
of dahdi.
Dahdi and zaptel can coexist in some 1.4 versions but it seems that from
1.4.22 onward, chan_zap.so is not built. Documentation within the 1.4.23.1
tarball indicates that one can keep using the
Hi,
My setup is:
IPPhone1 --- Asterisk1 with B410P Patton 4638 --- Asterisk2 ---
IPPhone2
I want to evaluate Asterisk1 in TE/PtmP mode.
So, Patton box is configured in NT/PtmP (with 3 BRI links between both
systems).
Anyway, asterisk -rx pri show spans keeps replying :
PRI span 1/0:
On Tue, Mar 10, 2009 at 6:40 AM, Kurian Thayil kurianmtha...@gmail.comwrote:
Hi All,
Is there a way that I can include call dialing functionality in a
webinterface. I have EyeBeam configured with a SIP user say
8440. Will I be able to design an inteface which agent can choose a number
and
On Tue, Mar 10, 2009 at 04:04:41AM -0700, Vieri wrote:
Hi,
It's not clear to me which asterisk version drops support for zaptel
in favor of dahdi.
In 1.6.0 .
Dahdi and zaptel can coexist in some 1.4 versions but it seems that
from 1.4.22 onward, chan_zap.so is not built.
Hi,
It seems BRI signalling settings are missing from
http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf
I would like to add those parameters :
bri_cpe_ptmp
bri_cpe
bri_net
Is this http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conftied
to a specific Asterisk
On Tue, Mar 10, 2009 at 01:51:27PM +0100, Olivier wrote:
Hi,
It seems BRI signalling settings are missing from
http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf
I would like to add those parameters :
bri_cpe_ptmp
bri_cpe
bri_net
Is this
Hi Steve,
That worked beautifully. Thank you so much. But one question though. Imagine
if I keep a Hangup Button in the interface and it should terminate the call.
Will that be possible? This scenario happens when the user gets connected to
an invalid phone number where the user have to manually
2009/3/10 Sasa s...@shoponweb.it
Hi, I have modified in Mobile/Setting the parameter SIP From from
tel/user to tel/tel and now I view the correct incoming number.
Thanks.
Glad I could help. It took me nearly a month to figure that out. ;-)
Chris
2009/3/10 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Mar 10, 2009 at 01:51:27PM +0100, Olivier wrote:
Hi,
It seems BRI signalling settings are missing from
http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf
I would like to add those parameters :
bri_cpe_ptmp
Don't forget to mention that the BRI signalling method doesn't work in 1.4 (and
probably 1.2) ;).
Andy
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: 10 March 2009 12:51
To: Asterisk Users
If you're using a php i'd take a look at phpagi - there are others around
for various different languages too. our agents use twinkle with
auto-answer, the only reason they need to look at twinkle is if they need to
perform a transfer (that too will soon be done from the web browser), you
can do
Came across this today
http://www.techcrunch.com/2009/03/09/y-combinators-voxli-targets-gamers-
with-browser-based-group-voice-chat/
Yet another opportunity Asterisk http://www.digium.com/ / Mexuar
http://www.mexuar.com/ / PhoneFromHere
http://www.phonefromhere.com/ / (insert the other
after installing asterisk 1.4.23.1 and dahdi-linux-2.1.0.4 and at CLI module
load chan_dahdi.so receive the following:
signalling must be specified before any channels are.
CLI Warning [4663]: chan_dahdi.c:11627 process_dahdi: Ignoring signalling
Error[4663]: chan_dahdi.c:10946
I am about to setup a new machine and based on a thread in the freetel-oslec
list, I came across the idea of compiling Intel optimizations in when using
oslec w/ dahdi. So I edit
dahdi-linux-complete-2.1.0.4+2.1.0.2/linux/drivers/dahdi/dahdi_config.h
to #define CONFIG_DAHDI_MMX which on its own
Hello,
I'm having difficulties to make Asterisk (1.6.0.6) interoperate correctly
with a CISCO mediaGW in order to send faxes to the PSTN using T.38.
When Asterisk sends the initial INVITE containing the T.38 media offer in
the SDP, the CISCO answers with a 488 Not Acceptable Media.
Apparently,
Aqua Man wrote:
after installing asterisk 1.4.23.1 and dahdi-linux-2.1.0.4 and at CLI
module load chan_dahdi.so receive the following:
signalling must be specified before any channels are.
CLI Warning [4663]: chan_dahdi.c:11627 process_dahdi: Ignoring signalling
2009/3/10 Andrew Thomas a...@datavox.co.uk
Don't forget to mention that the BRI signalling method doesn't work in 1.4
(and probably 1.2) ;).
Is it me or is it not either mentioned in chan_dahdi.conf after a make
samples ?
(grep BRI /etc/asterisk/chan_dahdi.conf returns nothing)
Is it
2009/3/10 Joseph L. Casale jcas...@activenetwerx.com
I am about to setup a new machine and based on a thread in the
freetel-oslec
list, I came across the idea of compiling Intel optimizations in when using
oslec w/ dahdi. So I edit dahdi-linux-complete-2.1.0.4+
Post up your chan_dahdi.conf and we'll fix it :)
Hint - you are missing : 'signalling = fxo_ks' and 'signalling = fxs_ks' from
it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aqua Man
Sent: 10 March
2009/2/27 Matthew Fredrickson cres...@digium.com
I have a couple of suggestions:
Make sure that your timing configuration is correct in
/etc/dahdi/system.conf (that it has a valid timing source).
Also, you probably will probably want to use the half_full buffer
policy, and set the number
2009/2/27 Matthew Fredrickson cres...@digium.com
I have a couple of suggestions:
snip
Olivier wrote:
2009/2/25 stoffell stoff...@gmail.com mailto:stoff...@gmail.com
Hi all,
I wanted to switch from my current setup (mISDN) to the native dahdi
with b410p support
On Tue, Mar 10, 2009 at 10:27 AM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
Hello,
I'm having difficulties to make Asterisk (1.6.0.6) interoperate correctly
with a CISCO mediaGW in order to send faxes to the PSTN using T.38.
Is it possible to make Asterisk work like this?
yes, as I've
Benny Amorsen schrieb:
Klaus Darilion klaus.mailingli...@pernau.at writes:
What are the typical ways to work around the 64 groups limit?
a) Split into different Asterisks
b) Use directed pickup instead, not *8
Maybe we should change the groups from a bitmask to an AST_LIST
regards
- Santiago Gimeno santiago.gim...@gmail.com wrote:
Hello,
I'm having difficulties to make Asterisk (1.6.0.6) interoperate
correctly with a CISCO mediaGW in order to send faxes to the PSTN
using T.38.
When Asterisk sends the initial INVITE containing the T.38 media offer
in the SDP,
I'm working on a project that involves Queues with Agents that are at home
with a PSTN phone number, NOT connected via SIP phones.
In the queues.conf it clearly states that only the SIP driver supports In
Use detection of making members of a Queue available or unavailable.
I've not yet figured
- Santiago Gimeno santiago.gim...@gmail.com wrote:
**The call-file I'm using is:
Channel: SIP/08099...@outbound-
calls
MaxRetries: 3
WaitTime: 30
Set: LOCALSTATIONID=2
Set: LOCALHEADERINFO=T38 fax
Set: T38CALL=1
Set: T38TXDETECT=yes
CallerID: 2
Context: fax-out
Benny Amorsen schrieb:
Klaus Darilion klaus.mailingli...@pernau.at writes:
What are the typical ways to work around the 64 groups limit?
b) Use directed pickup instead, not *8
So I would have to implement privileges (who is allowed to pick up whose
calls) manually - not easy
klaus
On Tue, Mar 10, 2009 at 11:18 AM, Joshua Colp jc...@digium.com wrote:
- Santiago Gimeno santiago.gim...@gmail.com wrote:
This was filed as an issue and is being tracked at
http://bugs.digium.com/view.php?id=12437. Thus far
I have created a branch for Asterisk 1.4 that changes the behavior
Why does enabling the mmx in dahdi_config.h break compilation?
I get the following:
{standard input}: Assembler messages:
{standard input}:86: Error: suffix or operands invalid for `mov'
{standard input}:87: Error: suffix or operands invalid for `mov'
make[3]: ***
Joshua Colp jc...@digium.com writes:
This was filed as an issue and is being tracked at
http://bugs.digium.com/view.php?id=12437. Thus far
I have created a branch for Asterisk 1.4 that changes the behavior to accept
the incoming INVITE with
either audio and T38, or only T38 (if we only got
Softphones out of the question?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter Beckman
Sent: Tuesday, March 10, 2009 10:24 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] 1.4.23 +
- Benny Amorsen benny+use...@amorsen.dk wrote:
Joshua Colp jc...@digium.com writes:
This was filed as an issue and is being tracked at
http://bugs.digium.com/view.php?id=12437. Thus far
I have created a branch for Asterisk 1.4 that changes the behavior
to accept the incoming INVITE
Klaus Darilion klaus.mailingli...@pernau.at writes:
Maybe we should change the groups from a bitmask to an AST_LIST
That would be a serious pessimization, but I guess it would work if
every phone is in just a few groups. The optimal implementation would
probably be a bit vector.
/Benny
IAX2 also support InUse, is a good choice for Agents at home because IAX is
nat friendly :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter Beckman
Sent: martes, 10 de marzo de 2009 12:24 p.m.
To:
Enjoyed the podcast :)
Does anyone have any idea what the pricing structure will be for this?
are we talking $10/channel? $100/channel? Does this log into the Skype
network as multiple users? One global user for the business as a whole?
Do I have to have 1 user login per inbound channel?
Hello,
Thanks everybody for the answers.
Could be. Would you post the Cisco config relevant to this?
dial-peer voice 5 voip
description ** **
preference 1
destination-pattern 1…
voice-class codec 1
session protocol sipv2
session target ipv4:1.1.1.1
session transport udp
dtmf-relay rtp-nte
[channels]
usecallerid=yes
callerid=asreceived
cidsignalling=bell
hidecallerid=no
callwaiting=yes
musiconhold=default
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=400
Enjoyed the podcast :)
Does anyone have any idea what the pricing structure will be for this?
are we talking $10/channel? $100/channel? Does this log into the Skype
network as multiple users? One global user for the business as a whole?
Do I have to have 1 user login per inbound channel?
On Tuesday 10 March 2009 05:31:38 Hooman Peiro wrote:
thanks for your responses, I checked again and I found that I asked a wrong
question! I was supposed to ask about answer time. The answer time is not
getting save in the database.
Answer time = calldate + duration - billsecs
--
Tilghman
On Tuesday 10 March 2009 03:11:57 Anthony Francis wrote:
Tilghman Lesher wrote:
On Monday 09 March 2009 01:28:49 pm Anthony Francis wrote:
Tilghman Lesher wrote:
On Friday 06 March 2009 11:24:46 pm Hooman Peiro wrote:
hi,
I'm working with asterisk on a project and I found a problem with
On Tue, Mar 10, 2009 at 04:20:47AM -1000, Aqua Man wrote:
after installing asterisk 1.4.23.1 and dahdi-linux-2.1.0.4 and at CLI
module load chan_dahdi.so receive the following:
signalling must be specified before any channels are.
CLI Warning [4663]: chan_dahdi.c:11627
--- On Tue, 3/10/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
But it will still be able to build vs. zaptel, and read
configuration
from zapata.conf .
See
http://svn.digium.com/svn/asterisk/branches/1.4/Zaptel-to-DAHDI.txt
Sorry to get back on this silly compilation issue. Also,
On Tue, Mar 10, 2009 at 05:00:38PM +0100, Benny Amorsen wrote:
Klaus Darilion klaus.mailingli...@pernau.at writes:
Maybe we should change the groups from a bitmask to an AST_LIST
That would be a serious pessimization,
Also space-wise
but I guess it would work if
every phone is in just
On Tue, Mar 10, 2009 at 12:19 PM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
dial-peer voice 5 voip
description ** **
preference 1
destination-pattern 1…
voice-class codec 1
session protocol sipv2
session target ipv4:1.1.1.1
session transport udp
dtmf-relay rtp-nte
fax-relay ecm
That appeared to work and the dahdi.conf I sent already to userslist. new
error message is :unable to create channel of type Zap' (cause 0 - Unknown)
Thanks
Date: Tue, 10 Mar 2009 18:37:35 +0200
From: tzafrir.co...@xorcom.com
To: asterisk-users@lists.digium.com
Subject: Re:
Olivier wrote:
Hi,
My setup is:
IPPhone1 --- Asterisk1 with B410P Patton 4638 --- Asterisk2 ---
IPPhone2
I want to evaluate Asterisk1 in TE/PtmP mode.
So, Patton box is configured in NT/PtmP (with 3 BRI links between both
systems).
Anyway, asterisk -rx pri show spans keeps
Olivier wrote:
2009/2/27 Matthew Fredrickson cres...@digium.com
mailto:cres...@digium.com
I have a couple of suggestions:
Make sure that your timing configuration is correct in
/etc/dahdi/system.conf (that it has a valid timing source).
Also, you probably will
On Tue, Mar 10, 2009 at 09:38:34AM -0700, Vieri wrote:
--- On Tue, 3/10/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
But it will still be able to build vs. zaptel, and read
configuration
from zapata.conf .
See
Asterisk Project Security Advisory - AST-2009-002
++
| Product | Asterisk |
Santiago Gimeno wrote:
Hello,
Thanks everybody for the answers.
Could be. Would you post the Cisco config relevant to this?
dial-peer voice 5 voip
description ** **
preference 1
destination-pattern 1…
voice-class codec 1
session protocol sipv2
session target ipv4:1.1.1.1
session
The Asterisk Development Team is pleased to announce the first release
candidate of Asterisk 1.4.24, tagged as version 1.4.24-rc1. Release candidate
1.4.24-rc1 is available for immediate download at http://downloads.digium.com/
In addition to other bug fixes, this release candidate fixes several
The Asterisk Development Team is pleased to announce the first release
candidate of Asterisk 1.6.0.7, tagged as version 1.6.0.7-rc1. Release candidate
1.6.0.7-rc1 is available for immediate download at http://downloads.digium.com/
In addition to other bug fixes, this release candidate resolves an
The Asterisk Development Team is pleased to announce the second release
candidate of Asterisk 1.6.1, tagged as version 1.6.1.0-rc2. Release candidate
1.6.1.0-rc2 is available for immediate download at http://downloads.digium.com/
In addition to other bug fixes, this release candidate adds
Thanks for the tip. Sadly, it didn't work. I keep getting the same error:
[Mar 10 18:49:48] WARNING[18855]: app_fax.c:176 phase_e_handler: Error
transmitting fax. result=11: Far end cannot receive at the resolution of the
image.
regards,
Santi
On Tue, Mar 10, 2009 at 6:36 PM, Matthew
Greetings listers,
I am running Asterisk 1.4.21.2 on Suse 11.0 on a
Dual Processor Dell Poweredge 1650. I recently attempted to update the BIOS
and now have this happen:
When the machine starts up, Asterisk runs fine. When I do a large wget or
scp, the local SIP to
What are the differences, or where do i find docs on the difference
between the 1.6.0.x and 1.6.1.x release?
Thanks!
jlc
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
Joseph L. Casale wrote:
What are the differences, or where do i find docs on the difference
between the 1.6.0.x and 1.6.1.x release?
Check the CHANGES file in the Asterisk source directory.
Leif Madsen.
___
-- Bandwidth and Colocation Provided by
Greetings!
We are using cisco 7940 phone with SIP and asterisk. We would like to be
able to have phone directories available on the phones that are sourced from
active directory. Are their any scripts that can connect to the AD server
via LDAP and then create the directory.html file for the
Joseph L. Casale wrote:
What are the differences, or where do i find docs on the difference
between the 1.6.0.x and 1.6.1.x release?
Thanks!
jlc
A good place to find that out is to look at the CHANGES file in the Asterisk
source. This file tells the of new features/behavior added since the
Hi All,
In RFC 2617 in Section 1.2 Access Authentication Framework states the below
mentioned:
A user agent that wishes to authenticate itself with an origin
server--usually, but not necessarily, after receiving a 401
(Unauthorized)--MAY do so by including an Authorization header
Danny Nicholas wrote:
Greetings listers,
I am running Asterisk 1.4.21.2 on Suse 11.0
on a Dual Processor Dell Poweredge 1650. I recently attempted to
update the BIOS and now have this happen:
When the machine starts up, Asterisk runs fine. When I do a large
On Tuesday 10 March 2009 13:32:37 Elizabeth Steinke wrote:
Greetings!
We are using cisco 7940 phone with SIP and asterisk. We would like to be
able to have phone directories available on the phones that are sourced
from active directory. Are their any scripts that can connect to the AD
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson
Sent: Tuesday, March 10, 2009 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Odd occurrence
Danny
Typed the dahdi restart command and the output is listed in the email as per
your request.
#[0;37;40m*CLI da##[K##[Kstop now#[7Gclear#[K#[7Gdahdi restart
#[0;37;40mDestroying channels and reloading DAHDI configuration.
Initial softhangup of all DAHDI
Hello everyone!
I installed Asterisk following the instructions of the book
Asterisk: The Future of Telephony. (very nice book)
However, I failed.
I installed zaptel, libpri and asterisk (in this order).
The Installation of Zaptel is successful and my TDM400P is correctly
detected:
# zttool
markus wrote:
I installed Asterisk following the instructions of the book
Asterisk: The Future of Telephony. (very nice book)
However, I failed.
You 'failed' because you installed Asterisk 1.6.0.6, which contains a
very large number of changes compared to Asterisk 1.4, which is what the
Zen Kato wrote:
When we use svn branches-1.4 such as:
# svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4
# svn checkout http://svn.digium.com/svn/libpri/branches/1.4 libpri-1.4
There aren't currently any branches for DAHDI, so you can either grab trunk (the
latest
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
markus wrote:
Hello everyone!
I installed Asterisk following the instructions of the book
Asterisk: The Future of Telephony. (very nice book)
However, I failed.
I installed zaptel, libpri and asterisk (in this order).
If you are using
On 7/03/2009 4:58 a.m., Klaus Darilion wrote:
Hi!
What are the typical ways to work around the 64 groups limit?
What we actually do is store a pickup group with a caller id.
So the AsteriskDB has ${DB/pickup/${CALLERID(num)}} and we set
pickupmark to the same.
That way when someone dials 29
Useful VIM command:
:%s/#\[.;3.;40m//g
On Tue, Mar 10, 2009 at 10:03:03AM -1000, Aqua Man wrote:
Typed the dahdi restart command and the output is listed in the
email as per your request.
*CLI dahdi restart
Destroying channels and reloading DAHDI configuration.
Initial softhangup of
Thank you, Doug, for precious information.
Best regards,
Marco Signorini.
===
INGEGNI Tech S.r.l.
http://www.ingegnitech.com
Doug Lytle wrote:
Main fax server:
Mandriva 2008.1
Kernel 2.6.24.5 (Compiled for source)
(1) Intel(R) Xeon(TM) CPU 2.80GHz
Digium TE110P (23
--- On Tue, 3/10/09, markus antro...@googlemail.com wrote:
Now I am missing
/usr/lib/asterisk/modules/chan_zap.so.
I searched through the mailing list and forums.
They say, that chan_zap.so is build in channels/ in my
working
directory.
But it's not too strange, that chan_zap.so was not
Vieri wrote:
Sorry to barge in like this but I would like to know if chan_zap.c is
supposed to be present in 1.4.23.1.
As documented in the CHANGES file that comes with 1.4.23, the answer is
no. chan_zap.c was renamed to chan_dahdi.c, but still supports Zaptel.
Please read the
Vieri wrote:
--- On Tue, 3/10/09, markus antro...@googlemail.com wrote:
Now I am missing
/usr/lib/asterisk/modules/chan_zap.so.
I searched through the mailing list and forums.
They say, that chan_zap.so is build in channels/ in my
working
directory.
But it's not too strange, that
--- On Tue, 3/10/09, Kevin P. Fleming kpflem...@digium.com wrote:
Sorry to barge in like this but I would like to know
if chan_zap.c is supposed to be present in 1.4.23.1.
As documented in the CHANGES file that comes with 1.4.23,
the answer is
no. chan_zap.c was renamed to chan_dahdi.c,
On Tue, 2009-03-10 at 15:44 -0500, Kevin P. Fleming wrote:
markus wrote:
I installed Asterisk following the instructions of the book
Asterisk: The Future of Telephony. (very nice book)
However, I failed.
You 'failed' because you installed Asterisk 1.6.0.6, which contains a
very large
Vieri escribió:
--- On Tue, 3/10/09, Kevin P. Fleming kpflem...@digium.com wrote:
Sorry to barge in like this but I would like to know
if chan_zap.c is supposed to be present in 1.4.23.1.
As documented in the CHANGES file that comes with 1.4.23,
the answer is
no. chan_zap.c was
Hi all,
I think running the macroexclusive application if it is run after hangup (on
h extension) crashes asterisk. This has happened a lot of times since i
started using the macro exclusive application.
There is a situation in my dialplan when after the user hangsup the call, i
execute the macro
I wish it was available too - I have just had to back dahdi out of a
system and revert to misdn after a whole day of testing.
PaulH
Andrew Thomas wrote:
I have LibPri installed and working (.../wPRI).
So, if I understand Tzafrir correctly - DAHDI support for the B410P isn't
available in
91 matches
Mail list logo
