in advance,
Aisling.
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The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should
Hi,
I have an issue which is kind of a catch 22 situation. I had outgoing calls
to my new PSTN provider working perfectly. Then I started focussing on
incoming calls. It seems that I can solve an error which gets my incoming
calls working but that in turns means my outgoing calls don't work. -
dtmf issue?
set
dtmfmode=rfc2833 in sip.confand try again.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aisling
Sent: Wednesday, February 01, 2006 11:03 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] XLite
dtmf issue?
Hi,
Im wondering if anyone has
appears vm-incorrect and I get an Unable
to read password message on the asterisk console. Has anyone experienced
issues with XLite dtmf?
Many thanks,
Aisling.
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The above electronic mail transmission
I see this timeout
error.
Thanks,
Aisling.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 11 January 2006 01:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Can't
= 2,1,Goto(mainconfmenu,s,1)
Many Thanks,
Aisling.
-Original Message-
From: Aisling [mailto:[EMAIL PROTECTED]
Sent: 11 January 2006 10:14
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main
Menu
Hi Kokmeng
add column delete varchar(3)
NOT NULL default no;
I get a message telling me that I have an error in my MySQL
syntax..Is this because the delete word I s a reserved word
and if so is this something others have experienced?
Many thanks,
Aisling.
---Legal Disclaimer
, but no rule 't' in context
'incomingpstn'
I used the 'Goto' as Iqbal suggested instead of includes...
Has anyone ever experienced this kind of behaviour before?
Many thanks,
Aisling.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 09 January
.
Many thanks,
Aisling.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Giovanni Miano
Sent: 05 January 2006 21:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls
Is Exist InternalExtension
and move to menu option 1 (another sound file) it wont
let me interrupt and I eventually get the error Timeout but no rule t
in context default.
Does anyone have any ides where the problem might be?
Many thanks,
Aisling.
---Legal Disclaimer
thanks,
Aisling.
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alyed Tzompa
Sent: 04 January 2006 00:28
To:
asterisk-users@lists.digium.com
Subject: re: [Asterisk-Users]
confusion about contexts
I'm a bit confused on how you get
your calls
should be applied.
I have included the relevant parts of my sip.conf and extensions.conf
below. I would appreciate any advice as to why these issues are
occurring.
Many thanks,
Aisling.
;sip.conf
[general]
bindport=5064
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
srvlookup=yes
anyone have any idea as to how I would solve this?
Hope someone can shed light on this,
Many thanks,
Aisling.
-Original Message-
From: Aisling
[mailto:[EMAIL PROTECTED]
Sent: 07 September 2005 13:54
To:
'asterisk-users@lists.digium.com'
Subject: Eeven Stranger - Asterisk
BT100 Password
missing
something important How do I get it?
Many Thanks.
-Original Message-
From: Aisling
[mailto:[EMAIL PROTECTED]]
Sent: 06 September 2005 18:09
To:
'asterisk-users@lists.digium.com'
Subject: Asterisk BT100 Password Issue
Hi,
I am getting the following error when I attempt
- Asterisk BT100 Password
Issue
I always get an unable to read password error if I hang up without
entering a password when prompted. Maybe is this what you are doing?
Even if you hear nothing, it is probably still expecting a password to
be entered.
On 9/7/05, Aisling [EMAIL PROTECTED] wrote
: [Asterisk-Users]
Asterisk won't listen on another port
try bindport=5062 and
bind the IP address too
bindaddr=IP_ADDRESS
On 9/5/05, Aisling [EMAIL PROTECTED] wrote:
Hello,
Hope somebody can help me Asterisk is behaving very
oddly and I'm totally stumped! I have SER
, Voicemail (u2092)
exten = 2092,
102, Voicemail (b2092)
exten = 2092,
103, Hangup
exten = ,
1, VoicemailMain(${CALLERIDNUM})
;voicemail.conf
[general]
format=wav
[from-sip]
2092 = 2092, 2092, emailaddress
Has anyone any inkling as to what the cause could be?
Many thanks,
Aisling
asterisk. A '4' was sent back to my phone.Strange that it
wasn't a 404 message, just a 4. Anyhow, when I removed the secret=1234
line from the sip.conf, the error still remains:
vm_authenticate: unable to read password.
Any further ideas?
Many thanks,
Aisling
-Original Message-
From: Alvin
that
sip.conf? Does anyone have other suggestions for what could be making Asterisk
behave so oddly?
Many thanks,
Aisling.
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Hello,
I have this already in sip.conf.
;sip.conf
[general]
context=default
port=5062
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no
autocreatepeer=yes
I have done sip reload and also restarted
asterisk with stop now and asterisk vvvgc.
Unfortunately Asterisk still does
Hello,
I have SER and Asterisk running on the same box. I want SER
to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have
configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will
purely act as a voicemail server at the moment. However I cannot get
/src/asterisk
Make: *** [depend]
Error 1
Has anyone come across this or does anyone
know a way of solving this?
Many thanks
-Original Message-
From: Aisling
[mailto:[EMAIL PROTECTED]
Sent: 26 August 2005 15:44
To: 'asterisk-users@lists.digium.com'
Subject: cvs update error?
Hi
I'm using suse linux.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Bockman
Sent: 29 August 2005 16:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FW: cvs update error?
I am trying to update Asterisk
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Cotton
Sent: 29 August 2005 17:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FW: cvs update error?
On Mon, 2005-08-29 at 14:04 +0100, Aisling wrote:
Hi,
I am trying
Hi,
Im
experiencing a problem with playing back my voicemail. (Failed
to write frame). It has been indicated in the archives that this is problem
can be solved by updating asterisk from the cvs. I
did make update in the /usr/src//asterisk
directory to resolve this. However I got a
on this?
Many Thanks,
Aisling.
---Legal Disclaimer---
The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should
Any more ideas on my below mail? If a user is registered with SER and
leaves a voicemail message with asterisk (by using rewritehostport
etc in ser.cfg), then how is the user supposed to listen to the
message afterwards? Is there any other way other than the MWI method??
Thnaksm
Aisling
a ridiculous question but how can I listen to my
message afterwards? I have read about a solution by Java Rockx using
sipsak for sending mwi sip notify messages to the phone but is there
a simpler way which I am blindly ignoring??
Thank you in advance,
Aisling.
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