When I receveid a call (num1) in the my office (num2), I transfer the call at the num3, but the callerid is num2, in the telephone3. What can I doing for show the callerid num1? Thanks
Yahoo! Mail: gratis 1GB per i messaggi, antispam, antivirus,
Use the o flag to force the original callerid, not the num2
callerid.
example:
exten = s,1,Dial(SIP/200,30,ortT)
SKM
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
asterisk183Sent: Monday, December 12, 2005 2:59 AMTo:
asteriskSubject: [Asterisk-Users] CallerID
Transfer
It doesn't work anymore!
When Voicemail([EMAIL PROTECTED]) is executed, the context is ignored in
Asterisk 1.2.1 with realtime voicemailboxes. I upgraded from 1.2.0 to
1.2.1 and it stopped working.
When the option searchcontext=yes, and looking at the debug messages of
the database queries the
Hi I am very new to asterisk
I am facing some problems
I have installed asterisk on my fedora core 3 by tar.gz
by
#cd /usr/local
#tar -xzvf asterisk.tar.gz
#make
#make install
#make samples
i made following changes in the sip.conf and extention.conf
In sip.conf
[500]
context=fromsip
The problem has been consistent from 1.0 through CVS to 1.2, and across
different machines and distributions. Does anyone have any suggestions
on how I can deal with this? I have had echo cancellation happening, but
half-duplex speech is not acceptable.
You're not using zaptel, what are you
Hi,
as of at least Dec 9, but also today, the cvs version of the chan-capi
on sf.net gives problems dialing out. The call gets out, but no audio in
any direction. Going back to a version from Dec 4th gives a working
system again.
All against svn branch 1.2 as of Dec 9th.
Anyone else
Several times asterisk has crashed with this message:
Dec 12 09:17:09 DEBUG[6792] chan_sip.c: Setting NAT on RTP to 0
Dec 12 09:17:09 DEBUG[6792] chan_sip.c: Checking SIP call limits for device
Dec 12 09:17:09 DEBUG[6792] chan_sip.c: build_route: Contact hop: sip:[EMAIL
PROTECTED]
Dec 12
Talat:
asterisk -r means to connect to an asterisk that is already running.
Try asterisk -gc
This will start asterisk and give you a console.
If you just want to run asterisk in the background, just run
asterisk
Then you can connect to that background asterisk with
asterisk -rc
HTH
Roger
Does anybody know what is difference between Cisco phones that end with 1 and 0
(7970 vs 7971; 7960 vs 7961 and 7940 vs 7941)? So far, as I could see and read,
only difference is that button which you choose channel, lights on when is
activated. Are there any other difference?
Models that end
Dear dude,
Please, if you solve this one and you find enough time, send one e-mail on list
with explanation how did you manage to make it all work.
Thank you!
Tomislav
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
dashy dude
Sent: 1.
Hi,
i discovered that in version 1.2.0 stable if i use a variable like:
CALLERID_FOO=12345
in the extension.conf, the variable is not evaluated and left empty.
In the 1.0.9 it was not this way.
May be a bug?
Bye,
Marcello
___
--Bandwidth and Colocation
Greetings to all.
I am looking for an application that allows dialing of phone
numbers from any Windows application, either via right-click, or via function
keys.
I am successfully using AstTapi, but want to take this one
step further and make Asttapi available from applications
Hi
I have installed asterisk 1.0.9 on my laptop which is running Redhat el3.
As it is when i use ulaw / alaw codecs my calls r easily getting thru
with good quality, but when i resort to speex i am getting the error
message on console : chan_sip.c:2792 process_sdp: No compatible
codecs!
my
Hi.First off, the illegal instruction doesn't look at all pretty.The best way to start a new installation is to start asterisk thus:/usr/sbin/asterisk -vcwhen you get a clean start with no fatal or significant errors, you can close it out and then start it as a daemon
/usr/sbin/asteriskOnly then
Hello,
Iam trying to configure asterisk with a PRI E1 line. I got to a point
where incoming calls on PRI is landing on asterisk and asterisk
immediately starts throwing the below errors on the console. The call is
dropped at this point somehow.
Dec 12 15:02:57 NOTICE[5783]: chan_zap.c:7437
Unable to get our IP address, Skinny
disabledIllegal instruction
It seem a processor invalid
instruction.
See the Makefile,
what is you processor ? Pentium ? Amd ? a had this problem
on Soekris.
i had change PROC=i486, i586 and
so
De:
[EMAIL PROTECTED]
Hi Roger
#asterisk -vvvgc
gave long list to verbosity and at the end it say
Illegal instruction (core dumped)
when i run
[EMAIL PROTECTED] postfix]# asterisk -rc
Unable to connect to remote asterisk
[EMAIL PROTECTED] postfix]#
According to my last mail i told that in modules.conf i unload few
Now I have unintentionally reproduced the problem again. The good news
is that I have a backtrace:
#0 0x08063c25 in ast_do_masquerade (original=0x818e500) at channel.c:2841
2841AST_LIST_INSERT_TAIL(original-varshead,
AST_LIST_FIRST(clone-varshead), entries);
(gdb) bt
#0
On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote:
Hi,
as of at least Dec 9, but also today, the cvs version of the chan-capi on
sf.net gives problems dialing out. The call gets out, but no audio in any
direction. Going back to a version from Dec 4th gives a working system again.
All against
We have had terrible echo when calling brisbanewhere are you calling
from/to?
PaulH
Melb
- Original Message -
From: James Andrewartha [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, December 12, 2005 6:15 PM
Hi Mark
The
#/usr/sbin/astersik -vc
---(many verbosity)
Illegal instruction
[EMAIL PROTECTED] postfix]# /usr/sbin/asterisk
[EMAIL PROTECTED] postfix]# /usr/sbin/asterisk -r
Unable to connect to remote asterisk
still give me illegal instruction in the end.
As long as i
Armin Schindler schrieb:
On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote:
Hi,
as of at least Dec 9, but also today, the cvs version of the chan-capi on
sf.net gives problems dialing out. The call gets out, but no audio in any
direction. Going back to a version from Dec 4th gives a working
Hello,
This looks like the asterisk you are using was compiled for a higher class
CPU then the one running it.
Regards
Kiss Karoly
On Mon, 12 Dec 2005, Talat Ishtiaq wrote:
Date: Mon, 12 Dec 2005 15:48:11 +0500
From: Talat Ishtiaq [EMAIL PROTECTED]
To: Mark Edwards [EMAIL PROTECTED]
Cc:
I do not think that speex is installed by default.
run show translations in asterisk and see what you get.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
Hi Douglas!
2. It'd be cool if the regcontext command actually did something.
There's a myth out there that it does something like execute a command
upon registration. Even the O'Reilly The Future of Telephony seems
to think this. After reading some posts in the developer discussion I
can
I assume that it would have to use a key sequence (Ctrl+Shift+A, etc.) that
does a copy of whatever is highlighted (in any app that supports text copy)
and pastes it into the dialing app. (either Tapi or softphone)
--
--
Steven
May you have the peace and freedom that come from abandoning
/var/log/asterisk/full text file may give you a more specific error.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
- --- - - -- - -- -- - -
Hi Kiss
I am trying to run it on p3 machine i think it should be enough
Regard
Talat
On Mon, 2005-12-12 at 12:18 +0100, Kiss Karoly wrote:
Hello,
This looks like the asterisk you are using was compiled for a higher class
CPU then the one running it.
Regards
Kiss Karoly
On Mon, 12
On Fri, 2005-12-09 at 06:51 -0600, [EMAIL PROTECTED] wrote:
Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes. One
of the boxes is
On Fri, 2005-12-09 at 15:53 +, Jason Williams wrote:
chan_capi registers fine:
**
[chan_capi.so] = (Common ISDN API for Asterisk)
== This box has 1 capi
OK, I did some testing with asterisk 1.0.7.
I went into production with asterisk 1.0.9.
I am using a 2 PCI slot 1U Dell 1750 server.
I am using 2 TE110P cards. 1 PRI to Telco. 1 EM to Panasonic DBS PBX with
150 people on it. (all Panasonic people reach Telco via asterisk)
I technically put the
Branko Samardzic wrote:
Hi everyone,
I am running two Asterisk servers on two machines that have dynamic DNS due
to ISP changing IP address daily. Both servers are registered on DynDns.org
and IP update scripts work fine on both machines. However, if one machine
changes IP address, other
Does anyone know if there are plans by Digium to have a PCI-X T1 card?
If so, any timing information?
Although throughput is higher on PCI-X, is interrupt processing any
better/worse than standard PCI?
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
Thank you! Thank you! Thank you!
This is what I was looking for! If you ever come in Croatia, give me a cool.
I'll buy you a drink!
Tomislav
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Sent: 2. prosinac 2005 16:24
To:
On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote:
Armin Schindler schrieb:
On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote:
Hi,
as of at least Dec 9, but also today, the cvs version of the chan-capi
on
sf.net gives problems dialing out. The call gets out, but no audio in
any
On Sun, 2005-12-11 at 09:56 +1100, Brad wrote:
[snip]
Anyone able to point me in the right direction to compile this app? It
is running ubuntu..
Not really but iirc there is a Mandrake/Mandriva mpg123 SRPM floating
around that had some x86_64 patches in it. Maybe you could try to track
the
Hello again,
as of at least Dec 9, but also today, the cvs version of the chan-capi
on
sf.net gives problems dialing out. The call gets out, but no audio in
any
direction. Going back to a version from Dec 4th gives a working system
again.
[..]
error 0x1103 is 'queue full', so the capi driver
Chris Bagnall wrote:
We've recently ordered a pair of these:
http://www.soekris.com/net4801.htm
Which have a standard PCI slot into which I'm hoping a TDM card will work.
Their Belgian distributor (kd85.com) appears to have a nice range of
expanded cases that might (hopefully) take a TDM card.
Kristian Kielhofner wrote:
P.S. - You should run AstLinux on your net4801:
http://www.astlinux.org
P.P.S. - I created AstLinux, and it rocks ;)!
Kristian,
We haven't decided yet, but AstLinux is definitely on our short list.
I'm a Debian person myself (I used to be a Debian developer),
On 12/12/05, Alistair Cunningham [EMAIL PROTECTED] wrote:
Chris Bagnall wrote:
We've recently ordered a pair of these:
http://www.soekris.com/net4801.htm
Which have a standard PCI slot into which I'm hoping a TDM card will work.
Their Belgian distributor (kd85.com) appears to have a
Kristof Hardy wrote:
The problem has been consistent from 1.0 through CVS to 1.2, and
across different machines and distributions. Does anyone have any
suggestions on how I can deal with this? I have had echo cancellation
happening, but half-duplex speech is not acceptable.
You're not using
Is there a way to persist agent statuses after a
restart?
Support I have to restart Asterisk for some reason,
is it possible that all logged in (AgentCallBackLogin) would remain logged
in?
Thank you
Dov
___
--Bandwidth and Colocation provided by
there is an asterisk agi (may not be an agi but is a program, I think it
was an agi) to do this for radio stations, perhaps a google for that, I
dont remember the exact name used but do remember that someone was
speaking about mass dialing to a radio contest line and bridging to
their phone
I'll third this Cepstral is superb! And it's at the right price!
On 12/10/05, Steve Totaro [EMAIL PROTECTED] wrote:
Cepstral sounds great. You can test it for free but I will append some
message about being free until you pay for the license.
I will be purchasing a license shortly but
Somebody implemented
the Chef-Secretary function in asterisk?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I am attempting to connect asterisk to a pbx to setup a 4 port voicemail
with auto attendant.
What I would like to do is if a call comes in on Zap/1 I would like to
play the auto attendant and when they select an extension use a transfer
to send them
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Disregard, I just found what I was looking for
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,Background(pls-wait-connect-call)
exten = s,4,Flash()
exten = s,5,SendDTMF($ARG1)
exten = s,6,Hangup()
Sean Cook wrote:
I am attempting to
List ... Darren,
In order to use a provider with unusual prefix 00
i.e. 001NXXNXX and providing failover to other providers with
the usual 1NXXNXX, decided to:
1. Change dialstr like that: IAX2/$res-{path}$phone|30|HL (excerpt from
below)
if ( $res-{tech} eq IAX2 ) {
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Use the o flag to force the original callerid, not the num2 callerid.
example:
exten = s,1,Dial(SIP/200,30,ortT)
And where to put this one? On first or on second call?
--
Tomislav Parcina
[EMAIL PROTECTED]
Douglas Garstang wrote:
The issues of NAT, call limit handling and registration expiration don't sound
quite so bad. I think we can live with those, if we can in fact just get a
central location database. Do you have any suggestions or ideas about how this
can be implemented with Asterisk?
Try it out. It looks to me like it would work but I've been wrong
often. :-)
Darren Wiebe
[EMAIL PROTECTED] wrote:
List ... Darren,
In order to use a provider with unusual prefix 00
i.e. 001NXXNXX and providing failover to other providers with
the usual 1NXXNXX, decided to:
1.
Douglas Garstang wrote:
Someone tell me how this sounds please. We will know the IP addresses of all our phones,
and the users/extensions on those phones because we will be the ones provisioning them.
We therefore write a script that reads from some source (file/database etc) and somehow
[EMAIL PROTECTED] wrote:
Hi Kevin and the list,
Yes, please, you must.
Why? The CVS server is not going away any time soon, and there are no
changes in that project nor any commits happening.
___
--Bandwidth and Colocation provided by Easynews.com
Hi,
In Asterisk voicemail... if I record a temporary greeting, how the
heck do I delete it and go back to using the normal greetings again?!
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update
Steven wrote:
Does anyone know if there are plans by Digium to have a PCI-X T1 card?
If so, any timing information?
We cannot say anything about our future product plans, sorry :-(
Although throughput is higher on PCI-X, is interrupt processing any
better/worse than standard PCI?
No, it's
On Mon, 2005-12-12 at 08:49 -0500, Matt wrote:
there is an asterisk agi (may not be an agi but is a program, I think it
was an agi) to do this for radio stations, perhaps a google for that, I
dont remember the exact name used but do remember that someone was
speaking about mass dialing to
0-4-2 Options, Temporary Greeting, Delete Temporary Greeting
Matt wrote:
Hi,
In Asterisk voicemail... if I record a temporary greeting, how the
heck do I delete it and go back to using the normal greetings again?!
___
--Bandwidth and Colocation
Go back into voicemail , mailbox options, press 4 to record temp greeting .
Once in this level of the menu you should have options 1 2. Option # 2
will be delete temp greeting.
Matt wrote:
Hi,
In Asterisk voicemail... if I record a temporary greeting, how the
heck do I delete it and go back
Kevin P. Fleming wrote:
Steven wrote:
Does anyone know if there are plans by Digium to have a PCI-X T1 card?
If so, any timing information?
We cannot say anything about our future product plans, sorry :-(
Although throughput is higher on PCI-X, is interrupt processing any
better/worse
Could DUNDI help him?
Or maybe a OpenSER plus Asterisk environment...
Denis.
On 12 de dez de 2005, at 12:41, Kevin P. Fleming wrote:
Douglas Garstang wrote:
The issues of NAT, call limit handling and registration expiration
don't sound quite so bad. I think we can live with those, if we
I finally got my issue resolved. It actually had nothing to do with my SIP.conf
file. The problem was how I was trying to set the callerid in my
extensions.conf file.
Anyway, do you have other voip providers that are working?
Do incoming calls work at all prior to timing out?
Are you NAT'ed, or
Thank you that will do it.
Wow that was slightly not intuitive :)
On 12/12/05, Steve Blair [EMAIL PROTECTED] wrote:
Go back into voicemail , mailbox options, press 4 to record temp greeting .
Once in this level of the menu you should have options 1 2. Option # 2
will be delete temp greeting.
I think I've got something whipped up with the queue... however, it
would be nice to not have to pick the phone up when it rings, but
rather have asterisk just queue as you speak of.. and connect the call
to your handset, already off-hook.. is there a way to do that?
On 12/12/05, trixter aka Bret
I posted on the wiki how to use cepstral with weather.agi. I am no
programmer so it is a hack but hey, it works great. I like William the
best but I have only tried William, Diane, and David. If someone finds
one that is better let me know.
Thanks
Steve
I'll third this Cepstral is
I wasted a lot of time on this and never figured it out. Finally went
with Madplayer. If you find a solution, please let us know.
On Sun, 2005-12-11 at 09:56 +1100, Brad wrote:
[snip]
Anyone able to point me in the right direction to compile this app?
It
is running ubuntu..
Not
Patrick wrote:
On Sun, 2005-12-11 at 09:56 +1100, Brad wrote:
[snip]
Anyone able to point me in the right direction to compile this app? It
is running ubuntu..
Not really but iirc there is a Mandrake/Mandriva mpg123 SRPM floating
around that had some x86_64 patches in it. Maybe you could
Try PRI debug span 1 and see if that sheds any light on the problem.
Thanks,
Steve
Hello,
Iam trying to configure asterisk with a PRI E1 line. I got to a point
where incoming calls on PRI is landing on asterisk and asterisk
immediately starts throwing the below errors on the console. The
This is called Zone Paging and can be implemented on *
Thanks,
Steve
I am asking all the VOIP Gurus and any developers out there if a
product
exists and if not if anyone would want to help me develop such
product.
With the onslaught of new homes that are wired with networking
Due to problems with SIP transfers and agents, we are using blind
transfers in asterisk(# key) for all calls. With 1.2.1, Asterisk is
doing a native bridge regardless.
Dial(SIP/phone,,to)
Using the above dial string and I see on the console that Asterisk is
attempting a native bridge. This
Also, you can apparently use Cepstral voices with Festival through a
wrapper.
http://www.cepstral.com/cgi-bin/downloads?page=other
Festival Wrapper - This Perl script creates a directory called
cepstral_swift in the current working directory for each voice
directory named as an argument. When
Look on mantis for some patch to do hdlc in hardware, it might help.
Zoa
Steve Totaro wrote:
Try PRI debug span 1 and see if that sheds any light on the problem.
Thanks,
Steve
Hello,
Iam trying to configure asterisk with a PRI E1 line. I got to a point
where incoming calls on PRI is
Thanks.
It works fine. I was just curious about
any collateral damages.
Thanks again,
benchev
On Monday 12 December 2005 16:42, Darren Wiebe wrote:
Try it out. It looks to me like it would work but I've been wrong
often. :-)
Darren Wiebe
[EMAIL PROTECTED] wrote:
List ... Darren,
In order
On Mon, 2005-12-12 at 08:41 -0700, Jason Becker wrote:
Patrick wrote:
On Sun, 2005-12-11 at 09:56 +1100, Brad wrote:
[snip]
Anyone able to point me in the right direction to compile this app? It
is running ubuntu..
Not really but iirc there is a Mandrake/Mandriva mpg123 SRPM
You need to recompile asterisk after you install speex.
On 12/12/05, Steven [EMAIL PROTECTED] wrote:
I do not think that speex is installed by default.run show translations in asterisk and see what you get.StevenMay you have the peace and freedom that come from abandoning all hope of
having a
Steve Underwood wrote:
Wouldn't anything new and high performance now be PCI-E, and not PCI-X?
I know hardly anything but video cards, and the occassional high end
RAID card, uses PCI-E, but it seems like that would be the direction for
a new card.
Yes, I assumed he meant PCI Express, even
I've been doing AGI now for 2 years, and this problem is making me feel
like I just started. :) I don't have this problem on pre 1.2
installations, so I'm assuming either this is something new, or I've
missed something in the change logs or on wiki.
Scenario:
Customer disables caller id on
Douglas Garstang wrote:
Someone tell me how this sounds please. We will know the IP addresses of all
our phones, and the users/extensions on those phones because we will be the
ones provisioning them. We therefore write a script that reads from some
source (file/database etc) and somehow
Steven wrote:
Questions:
How are other upgrading asterisk on production systems?
Are you buying duplicate Digium cards to test configs and reduce downtime?
We have purchased a duplicate server (with a spare TE410P Digium card).
Our primary interface to the telco is a T1 and we use channel
How could I make a modem-based banking software dialing out through
Asterisk? I have tried to use an ATA with a lossless compression but the
remote modem did not connected. Is it possible to use our Junghanns
QuadBRI card as a modem on a dedicated channel and sharing it as a COM
port via
Hey,
Thanks for the suggestion. I did but couldnt understand any of it. Here
is the link if anyone wants to try.
http://pastebin.ca/33394
Iam trying the suggested hardware hdlc patch. Will keep you guys posted.
dushyanth
Try PRI debug span 1 and see if that sheds any light on the problem.
Armin Schindler wrote:
On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote:
Armin Schindler schrieb:
On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote:
Hi,
as of at least Dec 9, but also today, the cvs version of the chan-capi
on
sf.net gives problems dialing out. The call gets out, but no audio in
Interesting, I just came along a product that can do all of this and
more, it's called Asterisk. You can find more info here:
http://www.asterisk.org/
http://www.digium.com/
http://www.voip-info.org/wiki-asterisk
http://www.google.com/
On 12/12/05, Goran Donev [EMAIL PROTECTED] wrote:
I am
Hello!
Soon i will add a second asterisk to my setup and of course i want it to use
the same postgresql-db as the first one. Basically it's about the cdr-uniqueid.
Since it could be possible that a record with the same uniqueid is written to
the cdr-table by both machines i'm lookin for a
On 12/12/05, Talat Ishtiaq [EMAIL PROTECTED] wrote:
Hi Kiss
I am trying to run it on p3 machine i think it should be enough
Sure, but if you got asterisk precompiled with optimizations for PIV
for instance, it won't work.
If you compiled asterisk yourself, be sure you're not using any flags
We've currently got 4 servers, and anytime we make any major
modifications to the servers, the phones have to be rebooted. We've got
about 55 cisco 7940's (which is going to steadily increase over the next
few months), does anyone know of a way to reboot the phones without
using the telnet
Yes, I meant Express.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
- --- - - -- - -- -- - --
Kevin P. Fleming [EMAIL PROTECTED] wrote in
Fellow list members,
I was wondering if anyone else out there has had
issues with Asterisk after doing a reload with a number of users
on.
I fixed a minor bug in my dial plan and did a
reload, and I seemed to have a corrupt config file afterwards. I am also
considering there may have been
Ok.. here is another pet peeve i have.. that maybe someone can answer...
When I do a call file.
How can I make the call be transfered to the second party BEFORE the
first party picks up? In other words.. right now if it's busy it will
keep trying.. however if it's ringing... it waits until PARTY
My guess would be the issue most likely happened in copying the config
file over.
Why don't you use something like SCP (Secure Copy).. it's like SSH...
but for files.
On 12/12/05, Franklin Webb [EMAIL PROTECTED] wrote:
Fellow list members,
I was wondering if anyone else out there has had
Hello all,
I need to install a production server with BRI support.
I know that exists bristuff, misdn, chan_capi ...
I have hcfpci based cards.
For a very stable environment, what driver should I use??
Thanks in advance
Pedro Nunes
___
--Bandwidth
Hey all, I have a bunch of Zultys ZIP2 phones, and they work fine with
asterisk except on one point, when I make an outgoing call via PSTN/Zap, the
call connects all is fine, but if I try to enter in any DTMF tones to
navigate a menu at the receiving end, the tones are never recognized by the
Pedro Nunes wrote:
I need to install a production server with BRI support.
I know that exists bristuff, misdn, chan_capi ...
I have hcfpci based cards.
For a very stable environment, what driver should I use??
My personal experience is only on using zaptel, it's also the most
'mature'
Hello,
Using * 1.2.1 with chan_capi CVS on a Diva server I am mostly happy.
However when a phone redirects a call (user forward) and all ISDN
channels are busy, the call goes out through an IAX connection and it
takes a few seconds to get a ring state from the remote * server. This
makes the
Hi,
I would like to make a list of all incoming and outgoing calls. From,
to, date, duration and for incoming, whether the calls were taken or not
and if yes, by which extension.
How do I do this? Put a line behind my dial command in the dialplan to
save the variables to a file? Any better
AMP does this by default, and then you can view the call log detail with
something like PHPMyAdmin. Otherwise:
http://www.google.ca/search?hl=enq=asterisk+call+logmeta=
Lots of stuff there. hth
-Original Message-
From: Arik Funke [mailto:[EMAIL PROTECTED]
Sent: Monday, December 12, 2005
I would like to make a list of all incoming and outgoing calls. From,
to, date, duration and for incoming, whether the calls were taken or not
and if yes, by which extension.
best is to use the built-in cdr options. (search voip-info.org with
asterisk cdr)
You can try the mysql cdr, it uses
Hello,
Iam not sure whether this patch will work for TE110P as the module i
load is wcte11xp and the patches specified at
(http://bugs.digium.com/view.php?id=5313) seems to patch the file
wct4xxp.c in zaptel.
Can anyone please confirm ? Or Wat the heck shuld i just go ahead and try ?
Dushyanth
Anyone using one of these as a QOS device in an Asterisk environment?
If so, does it work well?
Do you know what exactly it prioritizes? SIP only? IAX?
I bought one to play around with but read that it also prioritizes streaming
media in general..
The last thing I want is for this thing to
I am working with Xten lite for now. I am able to register
in but when I call out
I cant hear anything. The caller on the other end can
hear me just fine.
Any ideas?
I can get SIP to work fine internally.
I also have all the ports open in the firewall including
1 20
-J
For my asterisk installation in my lab, I will install the Linux ES v4 distribution (with kernel 2.6) ontoa Dell Power Edge 1650 with ~16GB of Raid-1 hard disk space.Before installing Linux, what should I set the following disk partitions to?: (root)/ /boot swap /usr /home /tmp /var
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