I found the decision in using
Channel: Local/[EMAIL PROTECTED]/n
Denis V. Gudtsov пишет:
> Hello, All!
>
> How to specify the context in call file section Channel? Is it possible?
>
> I want to dial external number (12345) and connect it to context
> "notify", which consist of playback() comman
Yes i tried but nothing change
Regards
Farooq
Quoting yusuf <[EMAIL PROTECTED]>:
> Farooq Ahmed wrote:
> > Hi All
> > Trying to install Digium B410P on Trixbox 2. After initializing
> card driver and asterisk i m getting
> > follow message asterisk shows no port.
> > Would be kind enough if someb
hi, everyone,
i have been sufferred for the asterisk hang on problem for so long and i
just reinstalled the whole thing yesterday, but again this morning the
server hangs on again, you could not call in through PSTN line and the ppl
also could not call out throught the server, there is simply enga
Just wanted to update the list
I found the problem. In my extensions.conf
I had
exten => 21,hint(SIP/21)
It should be
exten => 21,hint,SIP/21
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Wednesday, April 04, 2007 1:41 PM
To: Ast
On 4/4/07, John Schmerold <[EMAIL PROTECTED]> wrote:
What is that tone called & where is stored and configured.
I'd like to replace the ring with an announcement that is played until
the call is picked up or put into voicemail.
The ring is called "ring" and is defined in indications.conf. Pe
VGPS is a PHP/MySQL based provisioning system intended to generate
vendor-specific configuration files for Voice-over-IP (VoIP) devices
via a generic HTTP API.
Good luck...
http://sourceforge.net/projects/vgps
On 4/4/07, Forrest Beck <[EMAIL PROTECTED]> wrote:
I know this doesn't belong on thi
> Are there any documents/examples people have come across out there about
> using DUNDi to achieve load balancing/failover between 2 or more asterisk
> boxes? I've used DUNDi in the past, but primarily as a method of ensuring
> calls between locations take the lowest cost route (i.e. directly thr
Joe Acquisto wrote:
Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite
softphones, for eval/testing. They do get registered, and can call each other,
but mostly get no audio, sometimes one way audio.
Suggestions/fixes?
joe a.
Is there NAT on both sides? Are you us
Serial number?
Andrew Joakimsen wrote:
Well I would wonder how Polycom even had any idea whom your vendor is.
On 4/2/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:
Andrew Joakimsen wrote:
> First-sale doctrine, unless your vendor did something illicit to
> obtain Polycom phones there is nothing
Ever since upgrading from 1.2.X to 1.4.2, I'm having trouble with
voicemail. When played back, the messages start out okay, but after 10
seconds or so, the playback speed starts to increase and the voice
becomes illegible. It seems like some kind of audio timing problem.
Phone calls seem okay
"J. Oquendo" <[EMAIL PROTECTED]> Wrote: 4/4/2007 5:58 PM:
> On Wed, 04 Apr 2007, Joe Acquisto wrote:
>
>> >> iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j
>> >> ACCEPT
>> >> iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j
>> >> ACCEPT
>> >> ipta
t;
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I am getting the message "chan_gtalk.c:853 gtalk_alloc: no gtalk capable
clients to talk to." What does it mean? How can I find or make a "gtalk
capable client"?
It is Asterisk SVN-trunk-r59043.
"Jabber show connected" shows 1 connected jabber user.
"Jabber debug" periodically shows "JABBER: m
Je suis absent du 2/04/2007 au 11/04/2007.
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On Wed, Apr 04, 2007 at 11:51:21PM +0200, bram kortleven wrote:
> Well, I'm experiencing a similar problem with my setup... debian etch,
> asterisk 1.4.2, zaptel 1.4.1, ... I cannot find the chan_zap.so module
> file anywhere, tried recompiling with zaptel 1.4.0... no change... I
> tried 'make menu
2007/4/3, Tzafrir Cohen <[EMAIL PROTECTED]>:
On Mon, Apr 02, 2007 at 08:30:57PM +0300, Giedrius Augys wrote:
> Hi,
> I have FRITZ!Card PCI card. I have installed misdn-1.1.0 on stable
debian
> 3.1 with 2.6.8. kernel. Then I reboot system, and it doesn't boot, it
stops
> near "Apache2 starting..
After recompiling zaptel, did you recompile Asterisk?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bram
kortleven
Sent: Wednesday, April 04, 2007 14:51
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ZAP device reference in Zapt
Hi Phillip,
Thanks for replying. I do have all the item you listed in below email
perviously.
I reformatted my machine with FC5 this time and loaded up Asterisk
1.4.2with Asterisk-Addon
1.4 with MySQL modules now. I am sure the problem is related to FC6. I was
pulling my hair out for a while lol
Well I would wonder how Polycom even had any idea whom your vendor is.
On 4/2/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:
Andrew Joakimsen wrote:
> First-sale doctrine, unless your vendor did something illicit to
> obtain Polycom phones there is nothing they can do about it.
What they can do i
On Wed, 04 Apr 2007, Joe Acquisto wrote:
> >> iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j
> >> ACCEPT
> >> iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j
> >> ACCEPT
> >> iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p TCP -j
Well, I'm experiencing a similar problem with my setup... debian etch,
asterisk 1.4.2, zaptel 1.4.1, ... I cannot find the chan_zap.so module
file anywhere, tried recompiling with zaptel 1.4.0... no change... I
tried 'make menuselect', and going to the channels-part, chan_zap is
marked XXX -> depen
Alberto Alonso wrote:
> I am trying to use call parking. I have the following
> in features.conf
>
> [general]
> parkext => 700
> parkpos => 701-720
> context => parkedcalls
>
> When I try #700 from my softphone asterisk just passes it
> and doesn't interpret it.
>
> Can someone tell me what I
Brian McEntire wrote:
> One question... are there any places to get extra sound files like
> "activated" or "deactivated" or "do not disturb is..." ?? I didn't
> find them in the sounds directory after a vanilla install of the
> latest stable asterisk 1.4.
Maybe the asterisk-sounds tarball has s
On Wed, 4 Apr 2007, Brian McEntire wrote:
One question... are there any places to get extra sound files like
"activated" or "deactivated" or "do not disturb is..." ?? I didn't
find them in the sounds directory after a vanilla install of the
latest stable asterisk 1.4.
They are in the asterisk
I've got a system where I'm integrating a Nortel Option 11c with a
Trixbox 2.0.0 system using a Sangoma A101 T1 card. (Running on a Dell
PowerEdge 350)
We've got things mostly up and running and all seems well... except...
If I call from a SIP extension (X-lite soft phone) dialing 9 where
I am trying to use call parking. I have the following
in features.conf
[general]
parkext => 700
parkpos => 701-720
context => parkedcalls
When I try #700 from my softphone asterisk just passes it
and doesn't interpret it.
Can someone tell me what I am missing?
I am using asterisk-1.2.17
Thanks
If you are using Asterisk 1.4 you should look at the autofill configuration
option in queues.conf. For versions prior to that, I'm not sure there is a
solution.
On 4/4/07, Jordan Novak <[EMAIL PROTECTED]> wrote:
I am using rrmemory for my queues. I have noticed that the application
will only
I know this doesn't belong on this list but... I am looking to see if
anyone is using Polycom and knows of a web based software for
creating/managing the cfg files for polycom phones. I see that the
AsteriskNow will add provisioning support for Polycom phones. Since
it is still in beta, I was j
One question... are there any places to get extra sound files like
"activated" or "deactivated" or "do not disturb is..." ?? I didn't
find them in the sounds directory after a vanilla install of the
latest stable asterisk 1.4.
As I couldn't find such files under 'sounds', I created them by "hand"
"Joe Acquisto" <[EMAIL PROTECTED]> Wrote: 4/4/2007 4:24 PM:
>
>> Easiest method in a nutshell...
>>
>> iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j
>> ACCEPT
>> iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j
>> ACCEPT
>> iptables -A INPUT -s
There wasn't a setting, but I set it to rfc2833.
On Wednesday 04 April 2007 12:49, Noah Miller wrote:
> Hi Mike -
>
> > Well, when I restart the cli as requested below and go the addition steps
> > of setting verbose to 25 and turning sip debug on for the phone in test,
> > I don't get ANYTHING on
> Easiest method in a nutshell...
>
> iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p TCP -j
> ACCEPT
> iptables -A INPUT -s YOUR_HOST -i ETHERNET_CARD -d PBX_SERVER -p UDP -j
> ACCEPT
> iptables -A INPUT -s PBX_SERVER -i ETHERNET_CARD -d YOUR_HOST -p TCP -j
> REJECT
> iptable
Hi,
Today's setup is :
Legacy PBX1 with E1 --- Leased line - Legacy PBX2 with E1
Prospective setup is :
PBX1 --- Asterisk GateWay1 (with Digium E1) -- IP network --
Asterisk GW2 (with Digium E1) - PBX2
Is there a way to tunnel, transport or translate Q.SIG signals bet
I am using rrmemory for my queues. I have noticed that the application
will only distribute or dial one number at a time. Is there a different
strategy that will allow the queue to distribute more than one call at a
time? I don't want to use ringall because that would tie up thirteen of
my trunks e
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
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Hello,
I just upgraded my system from 1.2.10 to 1.4.2
Now I am having problems with speex codec.
sound is totally garbled and destroyed.
In 1.2.10 speex codec worked ok.
As a SIP client I am using ekiga with
narrowband speex (8000bps) enabled.
Any ideas?
Regards,
Jure Petrovic
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
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Ah!
Got it. Hard coding CallerID is a good idea and thank you for the example.
I decided to try the Noop(DB(...)) to see what was getting passed and
the empty CALLERID was the issue.
I decided to skip that and implement a global DND since that's what I
wanted anyway so I just set DND/ALL=1 in t
Zaptel has no direct code relationship with Asterisk. Your error is
because zaptel is trying to use a member no longer exists in newer
kernels. However you are using fedora, and fedora included that change
in older kernel. I found this in xpp/xbus-core.c
/*
* As part of the "inode diet" the priva
Hi Mike -
Well, when I restart the cli as requested below and go the addition steps of
setting verbose to 25 and turning sip debug on for the phone in test, I don't
get ANYTHING on the console. Sounds like it's a phone issue after all,
right?
I've got the same symptoms for BOTH the Sipura 2002
Oops, sorry.
My explanation was not clear. I replaced that function in
the codecs/Makefile in asterisk.
I will attach the modified your patch.
I hope this explanation helps.
--
Xiu YuShen
2007/4/4, Tzafrir Cohen <[EMAIL PROTECTED]>:
On Wed, Apr 04, 2007 at 06:50:50PM +0900, Xiu YuShen wrote:
>
Just wanted to update the group.
I copied the config file to a working server and the hints worked
without any problems.
Can anyone tell me if they have a working system using hits and
SVN-branch-1.4-r59289 or newer.
Eric Hall
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
On attempting to make Zaptel 1.2.16 on FC5, I get the following messages:
/usr/src/zaptel-1.2.16/xpp/xbus-core.c: In function 'debugfs_open':
/usr/src/zaptel-1.2.16/xpp/xbus-core.c:171: error: 'struct inode' has no member
named 'u'
make[3]: *** [/usr/src/zaptel-1.2.16/xpp/xbus-core.o] Error 1
mak
Joe Acquisto wrote:
Attempts to do SIP thru firewall (IPCop) are unsuccessful. using
x-lite softphones, for eval/testing. They do get registered, and can
call each other, but mostly get no audio, sometimes one way audio.
Suggestions/fixes?
joe a.
Easiest method in a nutshell...
iptables -
Chis.
Contact me off line if you are interested i can go to you system via ssh and
then tell you what happend.
Regards.
Polo
[EMAIL PROTECTED]
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Chris Blunt
Enviado el: Miércoles, 04 de Abril de 2007 10:53 a.m
Hi, how can I see in the console only my commands and its results?
There is any way to disable the activity logger in the console by command?
Thanks
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(This subthread is more appropriate to -users than to -dev, so it is
crossposted only to mark its transition. Please reply on the -user list
only.)
What are the cheapest prices for (humans) transcribing voicemail to
text as a service? The absolute cheapest, regardless of (known) quality
-
Hello again
I tried the "yum install kernel-smp-devel" this seemed to download an
updated version that was not the same as the version running, so I backed it
out using "rpm -e kernel-smp-devel"
I then proceeded to do "uname -r" to verify the kernel version (output:
2.6.9-42.0.3.ELsmp) and did "
Dovid B wrote:
> I have created this before. I have to dig up the dial plan. The way I
> created it is it would call user1. User1 had the option to take the
> call, pass it to the next user or send it to VM. If he passed it to
> the next user, User2 had the same options as user1 and it flows down
>
Tried this...it worked...but is this the best way?
== sip.conf ==
[general]
context=conference
allowguest=yes
[guest]
type=friend
nat=yes
host=dynamic
canreinvite=no
context=conference
--- Richard OSS <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I am configuring a conferencing server an
Hi,
I am configuring a conferencing server and need to
allow SIP clients guest access.
In iax.conf, I can allow guest access to the
[conference] context with this entry
=== iax.conf ==
[guest]
type=user
host=dynamic
context=conference
So anyone connecting without username/password will be
Hello,
I'm wondering how it would be best for user to manage a whitelist-backlist
of incoming calls to be screened :
1. Would you choose typical cases (allow-forbid internal-external calls) ?
2. Would you "teach" users regular expressions (so that then can receive
calls from mobile) ?
Regards
_
Hello, All!
How to specify the context in call file section Channel? Is it possible?
I want to dial external number (12345) and connect it to context
"notify", which consist of playback() command:
Channel: SIP/12345
Callerid: auto <12345>
MaxRetries: 3
RetryTime: 40
WaitTime: 50
Context: notif
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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Tomislav Parcina wrote:
Rolz wrote:
I was wondering if anyone knows how accurate the values are when you do a
"sip show peers" from the CLI.
My configuration is:
Asterisk box (192.168.1.102) -> gigabit switch <- PC running x-lite
(192.168.1.100)
the CLI reports 101 ms delay
however, ping is
I have created this before. I have to dig up the dial plan. The way I
created it is it would call user1. User1 had the option to take the call,
pass it to the next user or send it to VM. If he passed it to the next user,
User2 had the same options as user1 and it flows down the list. Also every
- Original Message -
From: "Olle E Johansson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, April 03, 2007 8:59 AM
Subject: Re: [asterisk-users] 603 Error
2 apr 2007 kl. 10.16 skrev Dovid B:
Hi Guys,
I started getting this error
Hello,
I'm using Realtime to select extensions out of a database so that we can
provision inbound tollfree on the fly. Once I 'catch' the inbound, I want
to "get out" of realtime and use the regular extensions again. I thought I
could just use the goto statement and go to another context/en
Do you know of any general User Portal applications for various Asterisk
installations or are the Druid, Trixbox (sort of) etc all installation
specific and not platform transferable?
After looking yesterday this doesn't seem to exist.
Regards,
Dean Collins
[EMAIL PROTECTED]
+1-212-203-4357
Hello everybody,
I'm trying to understand how can I set the MoH class for parked calls.
I set the "incoming" class for calls, and it works correctly. When I park the
call, the music on hold is ok, but when I close the communication on the
parking side, the parked call gets the default music on hol
Trying to create an extension that will toggle an enum value in our
database...
exten => s,1,MYSQL(Connect connid localhost myuser tmppass asterisk)
exten => s,n,MYSQL(Query resultid ${connid} UPDATE\ night_service\ SET\
status=(SELECT\ CASE\ status\ WHEN\ \'y\'\ THEN\ \'n\'\ ELSE\ \'y\'\
END
Try the biz list.
- Original Message -
From: "Mattias Andersson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, April 04, 2007 12:31 AM
Subject: [asterisk-users] Lithuania
Hi All!
Maybe a little of topic.
Bout coming from Sweden a
Are you on a VPS ?
- Original Message -
From: Chris Blunt
To: asterisk-users@lists.digium.com
Sent: Wednesday, April 04, 2007 11:55 AM
Subject: Re: [asterisk-users] Zaptel 1.4.1 Install Modules CentOS
Hi Tzafir / List.
Thank you for your reply.
I have run: m
It can very well be that your TISP is playing that (if you are using
voip). The way to test is to have
Exten => s,1,Answer
exten => s,2,Playback(tt-monkeys)
If you still hear ringing before it plays the file then there isnt much that
you can do.
- Original Message -
From: "John S
Hi,
It seems to perfectly match what I was after :
- Alice calls Bob and Bob doesn't answer (busy ? not there ?).
- Alice hangs up and dials something (*41 for instance).
- Whenever Bob is hanging up a call (that would prove Bob is back and
probably available), a call from Alice to Bob is trigger
Is there a way to cause asterisk to accept all calls without any authentication?
Mark
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On Wed, 4 Apr 2007, Brian McEntire wrote:
Don't think that was it unless I still have a typo. Here's my line
from extensions.conf:
exten => _#78,n,Set(DB(DND/${CALLERID(num)})=1)
in the CLI issued 'reload' after saving the updated extensions.conf
and then picked up the phone and dialed #78. St
Greetings list,
There have been quite a few posts on the list over the last few months about
using DUNDi to ensure users are always reachable even when logged into
different asterisk boxes (as part of a load balancing cluster).
For example, yesterday, this was in a post: (Olle Johansson)
" In c
On Wed, Apr 04, 2007 at 10:11:01AM +0300, Tzafrir Cohen wrote:
> Hi
>
> On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote:
> > monk*CLI> zap show channels
> > No such command 'zap show' (type 'help' for help)
> >
> > Does that mean I dont have ZAP support in Asterisk?
>
> Maybe.
>
On Wed, Apr 04, 2007 at 02:31:54PM +0100, Chris Blunt wrote:
> Hi Tzafir / List
>
> Here is some more information obtained from the commands you gave me:
>
> 2.6.9-42.0.3.ELsmp #1 SMP Fri Oct 6 06:21:39 CDT 2006 i686 i686 i386
> GNU/Linux
>
> kernel-2.6.9-42.EL
> kernel-smp-2.6.9-42.EL
> kernel-
Hi Tzafir / List
Here is some more information obtained from the commands you gave me:
2.6.9-42.0.3.ELsmp #1 SMP Fri Oct 6 06:21:39 CDT 2006 i686 i686 i386
GNU/Linux
kernel-2.6.9-42.EL
kernel-smp-2.6.9-42.EL
kernel-ib-1.0-1
kernel-devel-2.6.9-42.0.3.EL
kernel-2.6.9-42.0.3.EL
kernel-smp-2.6.9-42.
When I call into my Asterisk system, before the call is picked up, I
hear a ring tone.
What is that tone called & where is stored and configured.
I'd like to replace the ring with an announcement that is played until
the call is picked up or put into voicemail.
TIA
_
Don't think that was it unless I still have a typo. Here's my line
from extensions.conf:
exten => _#78,n,Set(DB(DND/${CALLERID(num)})=1)
in the CLI issued 'reload' after saving the updated extensions.conf
and then picked up the phone and dialed #78. Still getting this error:
[Apr 4 08:56:20] W
Joe Acquisto wrote:
. . .
http://sourceforge.net/projects/stun/
Which is linked from:
http://www.vovida.org/applications/downloads/stun/
That's what I'm running.
Gordon
Thanks. Looking there, why would I need a "stun client" if the
device/softdevice already has STUN suppor
On Wed, 4 Apr 2007, Joe Acquisto wrote:
. . .
http://sourceforge.net/projects/stun/
Which is linked from:
http://www.vovida.org/applications/downloads/stun/
That's what I'm running.
Gordon
Thanks. Looking there, why would I need a "stun client" if the
device/softdevice already has ST
On Tuesday 03 April 2007 20:09, [EMAIL PROTECTED]
wrote:
> Charles Ulrich wrote:
> > I have an Asterisk system deployed at a customer's site. It is
> > connected to the outside world by a local SIP provider. When
> > someone calls in through the trunk to leave a voicemail, Asterisk
> > is not send
. . .
> http://sourceforge.net/projects/stun/
>
> Which is linked from:
>
>http://www.vovida.org/applications/downloads/stun/
>
> That's what I'm running.
>
> Gordon
Thanks. Looking there, why would I need a "stun client" if the
device/softdevice already has STUN support?
All I shoul
On Wed, 4 Apr 2007, Joe Acquisto wrote:
Gordon Henderson <[EMAIL PROTECTED]> Wrote: 4/4/2007 3:32 AM:
On Tue, 3 Apr 2007, Joe Acquisto wrote:
Is it possible to install a stun server on asterisk?
You can install a stun server on the same PC that asterisk is running
on.
No need for it to be p
Hello all,
We are setting up a gateway in which the SIP devices will be connected
dynamically using the Asterisk system.
We use the originate Manager API command from our code to call an IP as
(SIP/[EMAIL PROTECTED]). The call rings on the phone and goes through the
normal (default) context
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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Rolz wrote:
I was wondering if anyone knows how accurate the values are when you do a
"sip show peers" from the CLI.
My configuration is:
Asterisk box (192.168.1.102) -> gigabit switch <- PC running x-lite
(192.168.1.100)
the CLI reports 101 ms delay
however, ping is showing <1ms delay
Where
No I don't. So that will be my problem.
Thanks.
On 4/4/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
Hi
On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote:
> monk*CLI> zap show channels
> No such command 'zap show' (type 'help' for help)
>
> Does that mean I dont have ZAP support
New install, Asterisk, obviously, Baystack 450 swtiches, verizon T1, Digium 4
port T1 card
Some (few) users have had complaints from their clients that sound quality is
poor. I do not know if the calls were placed via asterisk, or received via
asterisk. If it matters.
I believe this is a "
Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite
softphones, for eval/testing. They do get registered, and can call each other,
but mostly get no audio, sometimes one way audio.
Suggestions/fixes?
joe a.
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Gordon Henderson <[EMAIL PROTECTED]> Wrote: 4/4/2007 3:32 AM:
> On Tue, 3 Apr 2007, Joe Acquisto wrote:
>
>> Is it possible to install a stun server on asterisk?
>
> You can install a stun server on the same PC that asterisk is running
> on.
> No need for it to be part of asterisk itself, it's
Farooq Ahmed wrote:
Hi All
Trying to install Digium B410P on Trixbox 2. After initializing card driver and asterisk i m getting
follow message asterisk shows no port.
Would be kind enough if somebody help me.
Regards
Farooq
#misdnportinfo
Port 1: TE-mode BRI S/T interface line (for phone lin
Hi, i need to find a job in Saudi Arabia related to the field of
VoIP/Asterisk. But i live in Pakistan, so anyone who can provide me the list
of companies working on VoIP/Asterisk related projects plz share.
Thanx in advance
--
Regards
Rizwan Hisham
Software Engineer
_
On Wed, Apr 04, 2007 at 09:55:33AM +0100, Chris Blunt wrote:
> Hi Tzafir / List.
>
>
>
> Thank you for your reply.
>
>
>
> I have run: make clean
>
> Configure
>
> Make
>
> Make install
>
>
>
> I get no compile errors, but still the same problems i
Hi all,
what kind of cable to connect TE110 and avaya between Straight cable or
crossover cable
thanks,
ti
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On Wed, Apr 04, 2007 at 06:50:50PM +0900, Xiu YuShen wrote:
> 2007/3/17, Tzafrir Cohen <[EMAIL PROTECTED]>:
>
> >http://bugs.digium.com/view.php?id=9303
> >
> >Please test.
>
> I tested this patch to asterisk-1.2.17 with zaptel-1.2.10 and redhat 9.
>
> However, to compile on my environment, 'first
Hi,
Could you please explain what your provider is expecting?
You should only have to provide your public IP address.
On 4/4/07, Il Neofita <[EMAIL PROTECTED]> wrote:
Hi,
I am using my asterisk server like a gateway and one provider ask me to
pass an extra field with the IP of the peer that i
Hi All
Trying to install Digium B410P on Trixbox 2. After initializing card driver and
asterisk i m getting
follow message asterisk shows no port.
Would be kind enough if somebody help me.
Regards
Farooq
#misdnportinfo
Port 1: TE-mode BRI S/T interface line (for phone lines)
-> Protocol: DSS1
Hello,
is there anyway or any plan to have the date/time stamp that's printed
in an outgoing voicemail notification email to NOT be the date/time of
the (*) machine but infact correspond to the timezone set for the
subscriber under the TZ variable?
I have the (*) machine set to UTC and when the
I tested this patch to asterisk-1.2.17 with zaptel-1.2.10 and redhat 9.
However, to compile on my environment, 'first' function was replaced
by the 'firstword' function.
Regards,
Xiu
--
Xiu YuShen
2007/3/17, Tzafrir Cohen <[EMAIL PROTECTED]>:
On Fri, Mar 16, 2007 at 11:28:43AM -0400, Sean Br
make clean
configure
make linux26
make install
perhaps
Bails
Chris Blunt wrote:
Hi Tzafir / List.
Thank you for your reply.
I have run: make clean
Configure
Make
Make install
I get no compile errors, but still the same problems if I try to
Hi Tzafir / List.
Thank you for your reply.
I have run: make clean
Configure
Make
Make install
I get no compile errors, but still the same problems if I try to insmod
zaptel
As you suggested I tried modinfo zaptel
Which resulted in: modinfo
Hi,
I am using my asterisk server like a gateway and one provider ask me to pass
an extra field with the IP of the peer that is using the connection,
probably to have more control on the authentication. I was wondering how I
can implement this.
Thank you
__
Richard Lyman wrote:
Action: Originate
Channel: Local/[EMAIL PROTECTED]
Context: dummy
Exten: 2 Priority: 1
In extensions.conf
[dummy]
Exten => _X,1,System(*some command*)
remember your permissions
OK, thank you!
--
Tomislav Parcina
[EMAIL PROTECTED]
__
On Tue, 3 Apr 2007, Joe Acquisto wrote:
Is it possible to install a stun server on asterisk?
You can install a stun server on the same PC that asterisk is running on.
No need for it to be part of asterisk itself, it's a totally separate
program and will exist happily on the same server.
Go
Anyone have an idea to re master centos,in other worlds I have an asterisk
on centos with all libraries and modules,how can I make it as an iso image
?
Regards
*
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