To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM400 with 1 FXO
Klaverstyn, David C wrote:
My original post does have the contents of the file exactly.
In my /etc/asterisk/zapata.conf file I have
[trunkgroups]
[channels]
context=from-pstn
usecallerid=yes
app_ldap-1.0rc6 is not compiling on my CentOS 4.4 with Asterisk 1.4. I
have it working successfully using Asterisk 1.2. Can anyone give me any
hints?
make: *** [app_ldap.o] Error 1
___
--Bandwidth and Colocation provided by Easynews.com --
:36AM +1100, Klaverstyn, David C wrote:
app_ldap-1.0rc6 is not compiling on my CentOS 4.4 with Asterisk 1.4.
I
have it working successfully using Asterisk 1.2. Can anyone give me
any
hints?
make: *** [app_ldap.o] Error 1
The real error message is a bit above that line
Please help,
What am I doing wrong?
If the caller id name is empty then I want to set it to Unknown.
Set(CALLERID(name)=If($[${CALLERID(name)} = ]?Unknown)
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
Is there any reason why you could not do this?
exten = 123,1,Dial(SIP/123)
exten = 123,n,Goto(s-${DIALSTATUS},1)
exten = 123,n,HangUp
exten = s-NOANSWER,1,Voicemail(u123)
exten = s-NOANSWER,2,Hangup
exten = s-BUSY,1,Voicemail(b123)
exten = s-BUSY,2,Hangup
Or you could have
Hi All,
I seem to be having a problem with all my VSPs. When I am registering
with them I don't seem to be passing my port number. This problem
causes other users the inability to call my VoIP number with the VSP.
My VSP showed me what they are seeing.
I have changed my useragent to
Hi All,
I am starting to get complains from users that the call volume is very
low and people are having problems haring what is said. This is for
internal calls (between extensions) and over ZAP. The problem seems to
be with the caller and callee, no matter if it is an incoming or
outgoing
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Sound Volume Low : between extensions
andover ZAP.
Klaverstyn, David C wrote:
Hi All,
I am starting to get complains from users that the call volume is very
low and people are having problems haring what is said
Hi All,
I think I have missed something as I am resisted with 4 VSPs and I can
not dial in using any one of them using the corresponding VoIP numbers
assigned with the VSP. I can make outbound calls to another VoIP number
to the same provider.
The weird thing is that I have a DID with a
This is my code (that I copied form somewhere) for paging a group of
phones. By dialling 99 it will page phones 2101, 2102 and 2105.
Just include the context ext-paging in your dial plan and modify the
extension numbers and all should be good.
This works on Linksys Phones but should also
There seems to be something in Asterisk that disconnects the call at 1
hour.
At 59 minutes there is a beep then 1 minute later the call is dropped.
I have a basic install Asterisk Ver. 1.2.13. I have not specifically
said that calls are to be disconnected at a certain time (not that I
Does anyone know how to remotely reboot a PolyCom specifically 601
phone?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] X100P clone dial problems.
Well, my PSTN card has:
signalling-fxs_ks
and that works for me.
Klaverstyn, David C wrote:
Thanks for your help.
This is my file.
[channels]
language=au
context
Discussion
Subject: Re: [asterisk-users] X100P clone dial problems.
Klaverstyn, David C wrote:
I have since added fxs_ks=1
This is meaningless. Follow the example that I posted.
and channel = 1
This has not fixed the problem. I do notice a warning on the reload
I'm not sure if I have a configuration problem or not. I am unable to
dial out. When I try to dial in I can hear the phone ring on the
dialling phone but Asterisk does not register anything.
In zaptel.conf I have
loadzone = au
defaultzone=au
fxsks=1
In zapata.conf
language=au
Sent: Monday, 11 December 2006 4:47 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] X100P clone dial problems.
On Mon, Dec 11, 2006 at 04:18:20PM +1100, Klaverstyn, David C wrote:
I'm not sure if I have a configuration problem or not. I am unable to
dial out. When I try to dial
I have just installed Asterisk wit a TE110P card. I have configured 30
channels which seems to be recognised by staff and zap show channels.
I can make outbound calls with exceptional call quality but inbound
(receiving) calls the caller get a message saying Your call could not be
connected,
Actually when you download the firmware, there is no information
anywhere about configuring the directory. Sure there are sample cfg
files with the firmware but just the basic ones and it still does not
explain how to configure certain things.
From: [EMAIL
I have the value of xfersound = beep in my features.conf
file but when a call is transferred there is no beep noise. Can someone please
assist?
features.conf
xfersound =
beep
; to indicate an attended transfer is complete
___
We used the SPA-94x fro desktop phones and
the speaker phones on them a pretty good. We have a Snom 360 and the speaker
phone is lousy. I have just updated the firmware to the latest version and it
seems to be a better. It is not as good as the SPA as the Snom has background hiss
on
Can someone please help me with a problem that I seem to
have with this Polycom 601 phone. It will not see my TFTP server and
keeps saying Could not contact boot server, using existing
configuration. I have Linksys phones that use the TFTP server
without any problems but this Polycom will
I have a Snom 360 phone that will not work on an Asterisk
server but it will on another server. This phone has been working for over 4
months or so. I can not figure it out. This is the only Snom phone that I
have so I can check it against another one. The PBX that fails, fails with any
I have a Snom 360 phone that will not work on an Asterisk
server but it will on another server. This phone has been working for
over 4 months or so. I can not figure it out. This is the only Snom
phone that I have so I can check it against another one. The PBX that
fails, fails with any
Discussion
Subject: Re: [asterisk-users] Incorrect Ring tone. Getting a US tone
whenit should be AU tone
What phones are you using? It could be a phone level issue.
(my aastra has a setting for AU sounds..)
PaulH
On Mon, 2006-10-30 at 16:13 +1100, Klaverstyn, David C wrote:
For some reason Asterisk
For some reason Asterisk is producing a US ring tone when it should be an
Australian ring tone. I am using ztdummy and do not have any cards
installed. My configuration is as follows. I am using Trixbox
1.2.2. Can someone please guide me into the right direction?
zaptel.conf
loadzone
]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Monday, 30 October 2006 3:26 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Incorrect Ring tone. Getting a US tone
when itshould be AU tone
On Mon, Oct 30, 2006 at 04:13:12PM +1100, Klaverstyn, David C wrote:
For some
I am having a problem with an Asterisk server, in that when it
is receiving a call from another Asterisk server using an IAX2 trunk the phone
rings for 10 ms and then there is a hungup from asterisk and then the phone
rings again before another hangup.
The funny thing is that after I
For anyone interested the problem was we
needed to add a bindaddr= for the IP address of the cluster (virtual IP).
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C
Sent: Thursday, 26 October 2006
4:43 PM
To:
asterisk-users@lists.digium.com
Can this be done?
I call Asterisk using my mobile (cell), Asterisk then hangs
up on me so I am not charged for the call. Asterisk then calls my mobile
(cell) presenting me with a dial tone allowing me to make through the
PBX. It does this based on caller ID and only allowing certain
I dont mind Trixbox. The early
version did have bugs but 1.2.2 seems to be pretty good. I do prefer not to
use Trixbox and use plain Asterisk as you get more control. I have modified
conf files in Trixbox, just to have Trixbox overwrite my changes.
From:
[EMAIL PROTECTED]
101 - 130 of 130 matches
Mail list logo
