Hello,
Can you monitor the buddies status with you polycom
phones ?
Harry
--- Michael Araba [EMAIL PROTECTED] a écrit :
I am having the same problems. The polycom phones
the 501 or 601 or 301 will list more more than 7
buddies neither will the 601 with an expansion
module monitor more than
I would sugest that you look into the agent queue system. It will not let
you do exactly what you whant but nearly. And it works very well for
callcenters.
Best regards
jan
--On Saturday, November 12, 2005 11:22:24 AM +0100 Pavel Jezek
[EMAIL PROTECTED] wrote:
I would like, that incomming
Post line 17 of your extensions.conf file.
On Sat, 2005-11-12 at 18:47 -0600, Greg Blakely wrote:
I just recently upgraded to the latest HEAD, and am now getting the
following warning:
-- Including context 'fromcnet' in context 'pots'
Nov 12 18:45:17 WARNING[3035]: pbx_config.c:1697
Hi everybody,
I was not successful to make my Asteirsk receive calls. Please help me
to set it up.
My Asterisk is behind a Linksys router.
My Local IP, i.e. Router's IP is 192.168.0.1
My External IP is 24.57.xxx.xxx
My Asterisk's IP is 192.168.0.105
My X-Lite's IP is 192.168.0.103 and it is on
Hi everybody,
I was once able to type in Dial 200@context and it worked on my other
computer, but on this computer it says No such command 'dial'.
Somebody told me that it had something to do with soundcard. I activated
load = chan_oss.so in modules.conf. But when I type show modules, I
don't
Title: Re: IAX2 calls being droppped
Hi
Thanks for the reply. The host can still be pinged. In fact, it is usually only 1 user which is dropped from the session while other users are in the session. I dont think it is a problem with routing to the host.
Steven
Message: 13
Date: Sat, 12
Hi,
I wonder, how reliable is spandsp for faxing with
asterisk as an endpoint when using digital pstn lines (Sangoma
AFT-board)?
Morten Tryfoss
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Guys, is the any CLI commands or info files where you can check how
many g729 codec
license installed.
Regards,
Lito
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Hi
I'm trying to integrate Arabic numbering in asterisk
I've been using agi-stream_file(FILENAME) , but it seems that it doesn't
like to send the filename with path
Like agi-stream_file(ar/filename)
Anyone knows if it is built this way?
Cheers
That's easy...
Just go into asterisk cli and type show g729
It will tell you how many are active and how many you have in total
Regards
Zafer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angelito
Manansala
Sent: Sunday, 13 November 2005 10:31 PM
*CLI show g729
No such command 'show g729' (type 'help' for help)
this means i have no g729 codec installed, right?
On 11/13/05, Zafer Khodr [EMAIL PROTECTED] wrote:
That's easy...
Just go into asterisk cli and type show g729
It will tell you how many are active and how many you have in
Right :)
Regards,
Sahil Gupta
VoiceValley
On Sun, 13 Nov 2005, Angelito Manansala wrote:
*CLI show g729
No such command 'show g729' (type 'help' for help)
this means i have no g729 codec installed, right?
On 11/13/05, Zafer Khodr [EMAIL PROTECTED] wrote:
That's easy...
Just go into
To have dial command woking at the cli you have to have the soundcard
working and the channel for that.
Have you got sound to work on the machines at all?
Best regards
jan
--On Sunday, November 13, 2005 05:59:57 AM -0500 Zeeshan
[EMAIL PROTECTED] wrote:
Hi everybody,
I was once able to
- Original Message -
From: Angelito Manansala [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, November 13, 2005 14:31
Subject: [Asterisk-Users] How to check how many G729 codec license
installed
Guys, is the any CLI commands or info files where you can check how
I was not successful to make my Asteirsk receive calls. Please help me
to set it up.
snip
But the problem begins when I dial to my Asterisk server from my other
phone which is 514-854-7804
What is this other number? Is it a DID, ported, what?
Presume it DID...
On debugging, I get the
sahil, what is the relation of aligator and g729codec?
On 11/13/05, Sahil Gupta [EMAIL PROTECTED] wrote:
Right :)
Regards,
Sahil Gupta
VoiceValley
On Sun, 13 Nov 2005, Angelito Manansala wrote:
*CLI show g729
No such command 'show g729' (type 'help' for help)
this means i have
Do:
*CLI show translations
If you see - (lines) on the G729 row/columns than you do not have any G729
support.
RG.
- Original Message -
From: Sahil Gupta [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday,
g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
g723 - - - - - - - - - - -
gsm - - 3 3 4 3 2 9 - - 131
ulaw - 5 - 1 3 2 1 8 - - 130
Yes.
- Original Message -
From: Angelito Manansala [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, November 13, 2005 1:23 PM
Subject: Re: [Asterisk-Users] How to check how many G729 codec
licenseinstalled
Hi all,
Can anyone please tell me which format music needs to be in for native
MoH if my local phones use alaw/ulaw and some gsm g729 connections
that come in through the Net.
Thanks and regards,
Patrick
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Did you load the kernel module 'capi.o' as well? This is the module which
provides the node /dev/capi20.
If you use mISDN, you don't need capiinit, which is for AVM drivers only.
Armin
On Sat, 12 Nov 2005, MBIT Technologies wrote:
Hi Guys
I'm having a problem getting CAPI to work on
Hello all,
I'm contemplating upgrading a client's asterisk system from 1.0.9 to 1.2
beta to take advantage of some of the new echo cancellers in the later
zaptel packages. Problem is, I'll be doing it without physical access to the
box and without being able to personally test the new echo
Hi,
When I try to make an outbound call via SIP I get the following logging
in the asterisk commandwindow:
Check for res for
is not a local user
is not a local user
Exactly that, needless to say the call-out fails. What could be the cause?
My sip.conf:
[general]
port = 5060
How to make sound card work. Shouldn't Linux find it automatically?
Zeeshan A Zakaria
-Original Message-
From: Jan Saell [mailto:[EMAIL PROTECTED]
Sent: Sunday, November 13, 2005 7:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dial
On Nov 13, 2005, at 8:36 AM, Chris Bagnall wrote:
Hello all,
I'm contemplating upgrading a client's asterisk system from 1.0.9
to 1.2
beta to take advantage of some of the new echo cancellers in the later
zaptel packages. Problem is, I'll be doing it without physical
access to the
box and
I now have chan_capi-cm 0.6 working with Asterisk 1.2 RC2.
But I have discovered a small problem.
I have a mix of analog and ISDN (BRI) lines coming in to my Asterisk box.
Both types of lines are fed into the same set of contexts.
In the previous version of chan_capi-cm that I was using (0.53
I have 2 voicepulse open access numbers coming in over SIP. I use them
for some testing and at other times I just comment out the register
lines and let them go to the voicepulse mailboxes.
I went to use them yesterday and they are not working. Calls go to the
voicemail and if I enable their
hey guys,
if i get the asterisk from CVS like cvs checkout -r v1-0 zaptel libpri
asterisk asterisk-addons asterisk-sounds do i get a stable one?
On 11/11/05, Mark Quitoriano [EMAIL PROTECTED] wrote:
Great! tnx matt!On 11/11/05, Matt Florell
[EMAIL PROTECTED] wrote:
It's CVS v1-0. Digium has said
Before you upgrade to 1.2 and potentially break a lot of
things, have you followed the instructions available at
http://www.voip-info.org/
wiki/view/Asterisk+zapata+gain+adjustment to adjust the
rxgain and txgain?
Don't suppose anyone knows of a 1004 Hz 0dB number I can call to test with
I'm contemplating upgrading a client's asterisk system from 1.0.9
to 1.2
beta to take advantage of some of the new echo cancellers in the later
zaptel packages. Problem is, I'll be doing it without physical
access to the
box and without being able to personally test the new echo
Well yes in theory - have you got it playing other sounds?
Best regards
jan
--On Sunday, November 13, 2005 09:37:26 AM -0500 Zeeshan
[EMAIL PROTECTED] wrote:
How to make sound card work. Shouldn't Linux find it automatically?
Zeeshan A Zakaria
-Original Message-
From: Jan Saell
thanks Vahan
you are right, I have changed 'call t1' for 'calls t1' in balance.php
and invoices.php files and then tried to create a new table named
'calls' but mysql 5 has changed syntax for 'TIMESTAMP DEFAULT' and this
is the error:
mysql CREATE TABLE calls (
- id BIGINT NOT NULL
Incrementally reduce those gains by 2db per day and listen to
your customer's feedback relative to echo. Don't bother using
milliwatt generators and ztmonitor. (Those tools are okay to
find a starting point if you have no other transmission test
sets, but will not help even one little
Hi Patrick,
It is mpe3 (.mp3)... u need to install mpg123 before it will work. You
can also use you own program if you would like, you will need to specify
all this stuff in the musiconhold.conf file + your zaptel.conf
Cheers,
MICHAEL TOOP
Tel 011 602 9309
Fax 011 656 1342
Mobile 083 364
Also AFAIK this is not available in 1.0.9, so make sure that you got
something newer than that.
On 11/13/05, Jan Saell [EMAIL PROTECTED] wrote:
Well yes in theory - have you got it playing other sounds?
Best regards
jan
--On Sunday, November 13, 2005 09:37:26 AM -0500 Zeeshan
[EMAIL
Guys. Has anybody been able to compile spandsp-0.0.2pre21c against 1.2rc2?
Seems spandsp-0.0.2pre21c is broken. :(
Compiles great against 1.2rc1 but no luck so far with rc2.
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Hi,
Not natively, but you can run a Bash command in your extensions.conf
use Lame or Sox to do the conversion for you.
Cheers,
MICHAEL TOOP
Tel 011 602 9309
Fax 011 656 1342
Mobile 083 364 2370
Web www.bizcall.co.za
Kuniyoshi Murata wrote:
Hi * users,
Is that possible to make
I dont think I understand what to do with extensions.conf ... do I need
to create an extension 720727 that * answers under [default]
below is what im getting in the console when I dial my FWD account
along with a piece of my iax.conf
that deals with fwd account ..
is this not considered a line
Can someone help me here. I installed asterisk briefly just to see if it
would install on my suse 9.3 system and now I can get rid of a cron job that
goes every minute.
I've deleted the asterisk user, looked in all of the cron files I can think of
but it is beyond me.
Here is the line that
On 13 Nov 2005, at 13:08, Tim Ashman wrote:
Can someone help me here. I installed asterisk briefly just to see
if it
would install on my suse 9.3 system and now I can get rid of a cron
job that
goes every minute.
I've deleted the asterisk user, looked in all of the cron files I
can
On Sunday November 13 2005 10:21 am, Jens Vagelpohl wrote:
On 13 Nov 2005, at 13:08, Tim Ashman wrote:
Can someone help me here. I installed asterisk briefly just to see
if it
would install on my suse 9.3 system and now I can get rid of a cron
job that
goes every minute.
I've
Hi ,
Can Any one let us know how to implement this solution "Playing Music at the Back Ground while the conversation is on and recording the same in real time" using asterisk . PSTN connectivity is on ISDN protocol.
regards
kiran
___
yes it appears that you do.
On Sun, 2005-11-13 at 13:12 -0500, Health Masters wrote:
I dont think I understand what to do with extensions.conf ... do I
need to create an extension 720727 that * answers under [default]
below is what im getting in the console when I dial my FWD account
along
Incrementally reduce those gains by 2db per day and listen to
your customer's feedback relative to echo. Don't bother using
milliwatt generators and ztmonitor. (Those tools are okay to
find a starting point if you have no other transmission test
sets, but will not help even one
On Sun, 13 Nov 2005, Faris Raouf wrote:
I now have chan_capi-cm 0.6 working with Asterisk 1.2 RC2.
But I have discovered a small problem.
I have a mix of analog and ISDN (BRI) lines coming in to my Asterisk box.
Both types of lines are fed into the same set of contexts.
In the
freebsd 5.4, asterisk-1.0.9_2, broadvoice byod plan.
I can't get this all working together. I followed the instructions on BV
website to the letter and tried every other combination I could think
of, but for incoming calls I only get
BV's message
The party you're trying to reach is busy and can
On Sun, 2005-11-13 at 13:59 -0500, synrat wrote:
freebsd 5.4, asterisk-1.0.9_2, broadvoice byod plan.
I can't get this all working together. I followed the instructions on BV
website to the letter and tried every other combination I could think
of, but for incoming calls I only get
BV's
On Nov 13, 2005, at 1:59 PM, synrat wrote:
freebsd 5.4, asterisk-1.0.9_2, broadvoice byod plan.
I can't get this all working together. I followed the instructions
on BV
website to the letter and tried every other combination I could think
of, but for incoming calls I only get
BV's message
Zeeshan wrote:
Hi everybody,
I was once able to type in Dial 200@context and it worked on my other
computer, but on this computer it says No such command 'dial'.
Somebody told me that it had something to do with soundcard. I activated
load = chan_oss.so in modules.conf. But when I type show
Hi,
Any experiences with Voip Reach. The prices seem quite
tempting. They are also listed as Digium partner.
Is the service decent? Why that flash(Mozilla doesn't like most
most of it, Konqueror went strait to bed).
Any IAX2 termination?
Thanks in advance,
benchev
On 11/13/05, Paul [EMAIL PROTECTED] wrote:
I have 2 voicepulse open access numbers coming in over SIP. I use themfor some testing and at other times I just comment out the registerlines and let them go to the voicepulse mailboxes.I went to use them yesterday and they are not working. Calls go to
Yeah but it appears that Teliax just charges you if cross that limit for
every minute you go over. So it's not a soft limit, that is just
marketing spin. It's a hard limit. Be so much nicer if Starbucks were to
use small, medium and large and ITSPs would just advertise fixed minute
plans as
snacktime wrote:
On 11/13/05, *Paul* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
I have 2 voicepulse open access numbers coming in over SIP. I use them
for some testing and at other times I just comment out the register
lines and let them go to the voicepulse mailboxes.
Patrick wrote:
Hi all,
Can anyone please tell me which format music needs to be in for native
MoH if my local phones use alaw/ulaw and some gsm g729 connections
that come in through the Net.
Well, normally SLIN or ULAW/ALAW, but you could do g729 or GSM. The thing is,
you want to encode
The softcap seems to include inbound minutes. Looks to me like it is
better to get your incoming DID's from one provider and deal with
termination separately.
If someone like me offered unlimited consulting ** for $150/month but
then said there was a ** softcap of 2 hours I would be called all
Mark Quitoriano wrote:
hey guys,
if i get the asterisk from CVS like cvs checkout -r v1-0 zaptel libpri
asterisk asterisk-addons asterisk-sounds do i get a stable one?
Yes, although version 1.2 in in release candidate stage and should be released
later this week.
This means that if you were
Patrick wrote:
On Wed, 2005-11-09 at 12:45 +, Are wrote:
We want to intergrate AstBill with a Groupeware or CRM but want input
what people will prefeer.
On our list today we have
http://www.sugarcrm.com/crm/
http://www.vtiger.com/
http://www.egroupware.org/
A couple more worth
I've got a voicemail server I made from four X100P cards (off eBay),
Fedora Core 4, connected to a Toshiba DK40 system. I'm using
Asterisk 1.0.9, and Zaptel 1.0.9.2.
It works great, except the card which receives a majority of the
activity occationally will go into a 'Red' alarm and
I have personally done this recently, and my advice is definately DO IT.
In my situation, I noticed a marked improvement in echo and general audio
quality.
I too had gain settings that were out of whack compared to what others had
experienced, but as long as your following the documented
Hi everyone,
Thank you to Andrew Kohlsmith and Rich Adamson for their input.
Do you use any wctdm kernel options? I see that you're from South
Africa, so
wctdm may need a different country identification in order to properly
interface to your telephone network? (opermode kernel module
Rafael R. GV wrote:
thanks Vahan
you are right, I have changed 'call t1' for 'calls t1' in balance.php
and invoices.php files and then tried to create a new table named
'calls' but mysql 5 has changed syntax for 'TIMESTAMP DEFAULT' and this
is the error:
- starttime TIMESTAMP
Matt Riddell wrote:
Patrick wrote:
Hi all,
Can anyone please tell me which format music needs to be in for native
MoH if my local phones use alaw/ulaw and some gsm g729 connections
that come in through the Net.
You can have all the codec versions of the moh file. Asterisk shall pick
the
I would like to ask you a question releted to Asterisk and libpri. I
was trying to find solution in internet but it failed. I have some
computers with Digium T1/E1 cards connected to terminating PBX's. In
most cases everything is allright and i can place calls to PSTN. But
there are PBX's such
Hi everyone,
We are glad to announce to all the Asterisk community the first
release of DeStar[1], a web based interface to manage the Asterisk
PBX.
DeStar provides high-level abstraction above the Asterisk
configuration, making it real easy to quickly setup a basic PBX, but
simultaneously
overlapdial setting should work but you also have to set immediate=no
if you want overlapdial to work
greetings
mk
2005/11/13, Piotr Dydycz [EMAIL PROTECTED]:
I would like to ask you a question releted to Asterisk and libpri. I
was trying to find solution in internet but it failed. I have some
A_ Navone,
You cannot use a Y connector on a data (ethernet) connection, you must
use a switch or and older hub to accomplish this.
Regards,
Humberto
2 SIP phones on Y data connector on 1 ethernet -
will that cause problems ?
thx in advance
Are there any features in asterisk that might be used to effect a
background dial task?
I want to be able to start a Dial task running, in chich the execution
can be automated via DTMF tones (e.g., voicemail retrieval), but not
block the originating thread of execution. The dial plan or AGI
John Biundo wrote:
Are there any features in asterisk that might be used to effect a
background dial task?
I want to be able to start a Dial task running, in chich the execution
can be automated via DTMF tones (e.g., voicemail retrieval), but not
block the originating thread of execution.
I took a look at vgetty as a solution for my home telephony needs, but the
lack of documentation (at least beginner-level documentation) led me to
give up. I have a couple of GNU/Linux (gentoo Debian) and FreeBSD boxes
at home. I have a POTS line and a digital cable modem. I am not
interested
I have been trying to get the value in the Referred-By header from
within the
parkandannounce application without success. I first tried getting it via
ast_channel assuming the SIP driver would request a channel for the inbound
call and Referred-By could be found that way. Maybe it can but I
[EMAIL PROTECTED] ~]# cat /etc/modprobe.conf
snip
install wcfxs /sbin/modprobe --ignore-install wcfxs /sbin/ztcfg
snip
alias wctdm wcfxs
alias wct2xxp wct4xxp
alias eth3 hisax
[EMAIL PROTECTED] ~]#
The above looks different then my setup, and is likely the result of a
previous asterisk
Asterisk can do all this and more. I would suggest starting by using this
project http://asteriskathome.sourceforge.net/. You can also check out this
site http://www.asteriskdocs.org they just published a book and can download
for free that is full of great information.
BTW when I set up my
Im still tring to figure out how to route my calls out of a secondary
termination provider when my primary fails. I have no idea how to even
attempt this.. It seems like it would be something simple...
Regards
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hi All Asterisk Montreal Users,
I invite you to join me for a coffee/beer, this Tuesday the 15th.
Meeting will be hold at pub St-Paul (they have good beer, nachos and chicken
wings...)
in the old Montréal at 5pm.
Thank you for coming!
Adrien
--
Adrien Laurent - CIO
514-284-2020 x 202
[EMAIL
Are you running off the rpms or compiled version?
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On Nov 13, 2005, at 4:31 PM, Noah Swint wrote:
Are you running off the rpms or compiled version?
Compiled. Actually, I had to compile and install it twice because
the first time I didn't have Zaptel installed (which needs to be
installed first, apparently).
Do you suppose it makes a
Hi all,
We have an asterisk server inside a network using an iax provider. The
firewall is based on Openbsd, and we would like to use PF's QOS
capabilities to ensure optimum quality.
We need to provide good throughput for other applications, so we need to
use scheme that borrows bandwith, that
I get 8 to 12 of these messages at the beginning of a call to a Cisco
7940 SIP phone:
Nov 13 17:27:56 NOTICE[27203]: rtp.c:1146 ast_rtp_raw_write: RTP
Transmission error to 192.168.1.254:32472: Operation not permitted
Other than these messages, the call proceeds normally.
From the code I
Thanks for the suggestion.
Sorry if my requirements weren't clear. I need a general purpose run
this in the background capability. The voicemail was simply an
example, and probably a bad one. I want the Dial command to go off on a
separate thread so the main thread can continue. What
On 11/13/05, Miloš Kocbek [EMAIL PROTECTED] wrote:
overlapdial setting should work but you also have to set immediate=no
if you want overlapdial to work
greetings
mk
unfortunatelly it didn't help :(
--
dydyczp
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John E. Elkin wrote:
Im still tring to figure out how to route my calls out of a secondary
termination provider when my primary fails. I have no idea how to even
attempt this.. It seems like it would be something simple...
Something like this, which is a basic idea, not a totally working
I've got a voicemail server I made from four X100P cards (off eBay),
Fedora Core 4, connected to a Toshiba DK40 system. I'm using
Asterisk 1.0.9, and Zaptel 1.0.9.2.
It works great, except the card which receives a majority of the
activity occationally will go into a 'Red' alarm and
Hello:
So far I have been using Asterisk with SIP and VoIP only.
I just received a couple a Zaptel cards from Digium (one
analog 2 FXS + 2 FXO, one T1), but I am hesitant to install
them because I am afraid I may break the kernel or something.
Since Asterisk is not tested under SuSE, I prefer
Thank you
this is what I made to solve it:
1.- create table 'calls' using: starttime TIMESTAMP DEFAULT
now() NOT NULL, (it can be starttime TIMESTAMP DEFAULT
CURRENT_TIMESTAMP NOT NULL, too) :
CREATE TABLE calls (
id BIGINT NOT NULL AUTO_INCREMENT,
sessionid CHAR(40) NOT NULL,
uniqueid
On Nov 13, 2005, at 6:09 PM, Rich Adamson wrote:
About a year and a half ago when I was running a couple of x100p's
there was an issue associated with disconnecting the pstn line from
the card. If I recall correctly, if the pstn line was removed for
more then a second or so (a couple of
In my telecom experience, overlap on a PRI isn't sending digits as
INFORMATION messages, but instead means to send the digits as DTMF tones
over the B channel. This is pretty common when connecting to the PSTN
where the carrier requires an authorization code, either for billing or
as an access
So what you ant to do is FORK and kick the first proc into background.
You may be able to do this with the Local Channel...
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
John Biundo
Sent: Sunday, November 13, 2005 8:24 PM
To: Asterisk
I recently implemented a Sipura SPA-2002 with one of my Asterisk
installations. On internal calls, the SPA generates ringtone as expected.
However, when I dial out via my IAX-based service provider, I hear both the
telco-generated ringtone as well as the SPA-generated ringtone. Sometimes,
the
Hi everybody,
After running make clean; make install, asterisk starts
installing itself but then terminates after some time with the following error:
/usr/bin/ld: cannot find -lidn
collect 2: Id returned 1 exit status
make[1]: *** [app_curl.so] Error 1
Hi,
I just installed Asterisk on a new computer, running VM
Ware. Installation doesn't seem to be successful. When I try to run Asterisk
with asterisk -vc, it says app_voicemail.so: load module failed, returning -1
Loading module app_voicemail.so failed!
After this
Look
at the Wiki, Its looks like you missed a library or
two...
http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ZeeshanSent: Sunday, November 13, 2005 10:16 PMTo:
asterisk-users@lists.digium.comSubject:
About a year and a half ago when I was running a couple of x100p's
there was an issue associated with disconnecting the pstn line from
the card. If I recall correctly, if the pstn line was removed for
more then a second or so (a couple of times), the card would go into
some unknown
I recently implemented a Sipura SPA-2002 with one of my Asterisk
installations. On internal calls, the SPA generates ringtone as expected.
However, when I dial out via my IAX-based service provider, I hear both the
telco-generated ringtone as well as the SPA-generated ringtone. Sometimes,
After running make clean; make install, asterisk starts installing itself but
then
terminates after some time with the following error:
/usr/bin/ld: cannot find -lidn
collect 2: Id returned 1 exit status
make[1]: *** [app_curl.so] Error 1
make[1]: Leaving directory
Hi,
Please forgive my ignorance.. Could you please elaborate more on the
raw music on hold? Is that a MOH class or something else? How do we
configure it?
Just in case my previous questions migh have been misleading... I'd
like to know if is possible to play a MOH file from the beginning (not
Hi all,
i'm new to this list, i wanted some
information about the Digium cards. I need these cards
to connect to the PSTN from home. All i need is to
connect to the PSTN and have one more port for
connecting my land line (local) phone, please advise
me on what i should buy to get this
ok .. I guess I should have read the iax.conf a little better.. it says
it right in there...
Thanks..
trixter aka Bret McDanel wrote:
yes it appears that you do.
On Sun, 2005-11-13 at 13:12 -0500, Health Masters wrote:
I dont think I understand what to do with extensions.conf
Hi,
I am running into a issue with a TDM04B card. When dialing
out I get an noticeable (extreme to some people) echo, in that I can hear
myself. The person on the other line doesnt hear any echo and the call
sounds perfect to them.
I checked and tested a few things as per
Hi all,
i'm new to this list, and amazed with
Asterisk. I wanted some information about the Digium
cards. I need these cards to connect to the PSTN from
home. All i need is to connect to the PSTN and have
one more port for connecting my land line (local)
phone, please advise me on
On Nov 13, 2005, at 6:59 PM, [EMAIL PROTECTED] wrote:
I took a look at vgetty as a solution for my home telephony needs,
but the
lack of documentation (at least beginner-level documentation) led
me to
give up. I have a couple of GNU/Linux (gentoo Debian) and
FreeBSD boxes
at home. I
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