Hi,
I've installed spandsp-0.0.4pre16
With this:
cd ~
svn co https://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addons
agx-ast-addonshttps://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addonsagx-ast-addons
cd agx-ast-addons/trunk
./build.sh
I've got this:
CMake Error in
Hello,
On my Lenny system, I've got libspandsp.a, libspandsp.la files and so on
present in /usr/lib.
How could I write a shell script that would read among those files and tell
installed spandsp is version 0.0.4pre12 or version 0.0.6pre3 ?
This is something autoconf tools must be able to do but
Hi All,
When our users receive a voicemail we send it attached to an email. It used to
work fine, encoded in wav49 and read by Windows media player. Recently the
default player in the company has become VLC which is unable to read wav49. I
am trying to use OGG/VORBIS instead of wav49. I can't
2009/3/10 Ali Jawad alijaw...@gmail.com
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
Hi!
Ist it possible with Asterisk to send SIP keep-alives with CRLF instead
of OPTIONS (qualify)? The OPTIONS are very noisy :-)
thanks
klaus
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asterisk-users mailing list
To
Hi!
AFAIS the incoming SUBSCRIBE is handled in the same context as INVITE -
but how should I handle the SUBSCRIBE in the context?
thanks
klaus
SUBSCRIBE sip:u+431234...@foobar.at:5160 SIP/2.0
Via: SIP/2.0/UDP
192.168.2.82:39982;branch=z9hG4bK-d8754z-3116e1207913aa4e-1---d8754z-;rport
My understanding of current SIP MWI handling is:
- no matter if an endpoint subscribed to receive message summaries, Asterisk
will a summary to it if sip.conf mailbox entry is filled.
- I couldn't find any SIP hardphone setting (i used a Thomson ST2030), that
would make the hardphone send a
I finally solved the issue by changing the resolution and the width of the
TIFF file to one that is accepted by the fax standard. In my case I changed
to a resolution of 96x96 and a width of 1728.
Now I am able to send faxes, but something weird is happening, the fax
received in the fax-machine
On Tue, Mar 10, 2009 at 02:11:58PM +0100, Christian Victor wrote:
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
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bex Vincent
Sent: Wednesday, March 11, 2009 3:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] VLC
Hi All,
When our users receive a voicemail we
Hi friends,
Do we have any way to prevent more than one Agent being logged in from the
same extension?
Also is there a way to limit an agent from logging in from more than one
extension?
I searched too much, but didn't find a solution.
Please help. Thanks in advance.
Shanavaz.
Santiago Gimeno wrote:
I finally solved the issue by changing the resolution and the width of
the TIFF file to one that is accepted by the fax standard. In my case
I changed to a resolution of 96x96 and a width of 1728.
Now I am able to send faxes, but something weird is happening, the fax
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shanavaz E A
Sent: Wednesday, March 11, 2009 9:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Multiple Agent Login
Hi friends,
Do we have any
2009/3/11 Håkan Källberg h...@simulina.se
Hello!
Does anyone of you have Caller Presentation working in the other
direction?? My mv370 is working well, execpt the Caller ID on outgoing
GSM calls. This works with the SIM card/Provider I am using if I put
the SIM card in a telephone, but not
On Wed, Mar 11, 2009 at 04:16:43PM +0100, Christian Victor wrote:
2009/3/11 Håkan Källberg h...@simulina.se
Does anyone of you have Caller Presentation working in the other
direction?? My mv370 is working well, execpt the Caller ID on outgoing
GSM calls. This works with the SIM
Hello all.
Please, I'd like to know if somebody can help me with this problem.
I have successfully configured a PBX with Asterisk 1.4 and a Digium analog card
with 4 ports.
This PBX has a lot of incoming and outgoing calls, and works perfect in
general, but there are some extrange cases where
Rosa De Santis wrote:
Hello all.
Please, I'd like to know if somebody can help me with this problem.
I have successfully configured a PBX with Asterisk 1.4 and a Digium analog
card with 4 ports.
This PBX has a lot of incoming and outgoing calls, and works perfect in
general, but there
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Wednesday, March 11, 2009 11:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with
On Wed, 11 Mar 2009, Rosa De Santis wrote:
Hello all.
Please, I'd like to know if somebody can help me with this problem.
I have successfully configured a PBX with Asterisk 1.4 and a Digium analog
card with 4 ports.
This PBX has a lot of incoming and outgoing calls, and works perfect in
Hi,
on file agents.conf use the option multiplelogin=no
--
Humberto Figuera - Using Linux 2.6.26
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603
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Hi gang,
I upgraded the E1000 driver on my machine from 7.3.20-k2-NAPI
to 8.0.9-NAPI. This unfortunately did nothing to resolve the problem. The
best workarounds I've come up with are:
1. use -l on scp and ftp
2. install wondershaper QOS and limit throughput to 32K.
These are
On Wed, 11 Mar 2009, Danny Nicholas wrote:
I upgraded the E1000 driver on my machine from 7.3.20-k2-NAPI to
8.0.9-NAPI. This unfortunately did nothing to resolve the problem.
The best workarounds I've come up with are:
1. use -l on scp and ftp
2. install wondershaper QOS and limit
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, March 11, 2009 1:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] update on Odd
Hi,
Until now I've been using my Digium B410P cards with misdn 1.0.x.
I would like to upgrade my systems and am now wondering which is the best
route to take:
- use the latest release of misdn v1
- upgrade to the latest stable kernel and use the built-in misdn v2
- use misdn v2 as a seperate
Vieri wrote:
- use the latest release of misdn v1
- upgrade to the latest stable kernel and use the built-in misdn v2
There is no support for mISDN v2 in Asterisk to my knowledge.
- use misdn v2 as a seperate package (disable misdn in the kernel)
See above.
- use dahdi's support for misdn
On Wed, Mar 11, 2009 at 02:56:58PM -0500, Kevin P. Fleming wrote:
Vieri wrote:
- use the latest release of misdn v1
- upgrade to the latest stable kernel and use the built-in misdn v2
There is no support for mISDN v2 in Asterisk to my knowledge.
It is available in a separate, out of
On Wed, 11 Mar 2009, Bex Vincent wrote:
When our users receive a voicemail we send it attached to an email. It
used to work fine, encoded in wav49 and read by Windows media player.
Recently the default player in the company has become VLC which is
unable to read wav49. I am trying to use
On Wed, 11 Mar 2009, Vieri wrote:
Hi,
Until now I've been using my Digium B410P cards with misdn 1.0.x.
I would like to upgrade my systems and am now wondering which is the
best route to take:
If it aint broke, don't fix it...
Saying that, I can feel a need even now to look at 1.4 - I
Idle curiosity: I like the look and feel of the Grandstreams, but it's
been my experience that the speakerphones suck (esp. when compared to the
pretty damn flawless Polycoms). I've used the BT-100/101 and GS-2000;
have any of their newer models changed that?
Thanks!
-Ken
--
This message has
On 12/03/2009 10:06 a.m., Ken D'Ambrosio wrote:
Idle curiosity: I like the look and feel of the Grandstreams, but it's
been my experience that the speakerphones suck (esp. when compared to the
pretty damn flawless Polycoms). I've used the BT-100/101 and GS-2000;
have any of their newer models
I just got some GXP2000s and they seem to have decent speaker phones. I
think I saw something about improved speaker phone in the sales lit.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken
Hello,
With an extensions.ael enabled system, I keep getting whatever I change into
my astup.call file :
[Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least
one of app or extension (or keyword message/pdu) must be specified, along
with tech and dest in file
On Wed, Mar 11, 2009 at 5:29 PM, Olivier oza-4...@myamail.com wrote:
Hello,
With an extensions.ael enabled system, I keep getting whatever I change
into my astup.call file :
[Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least
one of app or extension (or keyword
I have been using a number of the Grandstream GXP-2000 (74 in production),
GXP-2010 (1 in production), and BT-200 (15 in production) with great success.
The only issue that we have had is killing power supplies, not sure if this is
related to our power source or product. So far they have
Hi all,
I enabled recording (mixmonitor) in queue and process started after
queue member pick the call. But recording will stop after picking up
by another extensions of call transfer/parking in the same call. Is
it possible to continue to record the call for call parking/transfer,
how?
Rgds,
Sorry, I forgot to mention that I am using Asterisk 1.2.30
Hi friends,
Do we have any way to prevent more than one Agent being logged in from the
same extension?
Also is there a way to limit an agent from logging in from more than one
extension?
I searched too much, but didn't find a
hello,
I try to open the debug to compile dahdi with wcb4xxp,
=base.c
#ifdef DEBUG_LOWLEVEL_REGS
if (unlikely(DBG_REGS))
drv_dbg(b4-dev, read 0x%02x from 0x%p\n, ret, b4-addr +
reg);
#endif
if
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