k
-- Goto (ivr,s,4)
-- Executing BackGround("SIP/mvn-f877", "local/script8") in new
stack
Any suggestions would be appreciated.
Thank you
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
T
Copy and paste from my reply to a similar question a couple of weeks ago:
5000 sip registrations is quite a lot, but the more important thing is
the number of simultaneous calls.
If most of your calls is going to be SIP 2 SIP then I would suggest you
use openSER for the SIP registrations and
Sorry to do this but I sent a
couple of posts and I do not see them or any replies to them.
Could someone reply to this please.
Tx
Marnus
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093 patents,
Hi,
can somebody clear up the situation with SIP voicemail SUBSCRIBE and
realtime SIP peers in 1.4 for me.
From a lot of sketchy information about the new chan_sip in 1.4 I know
that it implements rfc compliant SUBSCRIBE behaviour (with the
subscribemwi=yes option?), but what about realtime
Your script will have to read the extra info from the msg.txt files
or it's realtime equivalent.
M
Now, maybe I'm stupid but how exactly
do I get details to it regarding all those VM variables that are
inserted when the email is normally sent out from voicemail. You know
the VM_NAME,
xecuting %s\n", arguments);
ast_safe_system(arguments);
}
}
M
RR wrote:
On 11/29/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
You can have your own external script to do
whatever you want when vm is
left
from voicemail.conf:
; If you need to
Anybody know of a SIP/IAX
softphone for Symbian Series 60? (Apart from the builtin Nokia one!)
Tx
M
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093 patents,
including the light bulb, phonogram and
Klaus Darilion wrote:
sillyant.com
Thanx, but AFAIK that can only be used with their service, not asterisk
or other SIP server.
M
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You can have your own external script to do whatever you want when vm is
left
from voicemail.conf:
; If you need to have an external program, i.e. /usr/bin/myapp
; called when a voicemail is left, delivered, or your voicemailbox
; is checked, uncomment this:
;externnotify=/usr/bin/myapp
M
RR
Details about externnotify and its arguments can be found here
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
M
RR wrote:
Hello all,
does anyone have a clever way of creating a customised email that goes
out as result of the voicemail notification. And I don't mean Editing
How many phones are in this group?
If only a couple, just put them all in the Dial statement (or a viarable).
Dial(Sip/phone1Sip/phone2Sip/phone3
or
PhoneGroup=Sip/phone1Sip/phone2Sip/phone3
and
Dial(${PhoneGroup})
M
nik600 wrote:
Hi
can i set up a group of SIP users and forward a
Have a look at the OpenSER and Asterisk part of
http://openser.org/dokuwiki/doku.php
and
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER
Arun Kumar wrote:
HI,
I'm not able to find some good doc or manual regarding Integration of
Asterisk with SER.
You need curl-devel just try
yum install curl-devel
Damon Estep wrote:
On version
1.2.12.1 running on FC4 with
curl.i386 installed the asterisk CURL function is not registered,
perhaps in
need something else (curl-devel.i386 ?)
Just use two different contexts for the two times of day (open/closed)
and use Playback to play the correct message before going direct into
voicemail without any prompt.
M
Wildheart wrote:
Hi,
I want to change my voicemail message based on the time of day. I would
like a message that
Supposing you have an extra column called 6second:
UPDATE cdr SET 6second=billsec+(6-mod(billsec,6) where 6second=0
if you want a decimal minutes column called billmin
UPDATE cdr SET billmin=round((billsec/60)+0.5),1) where billmin=0
Vicky wrote:
Thx and what would the sql query be
? . I
It is not easy (by a mile), but can be done.
have a look at http://www.dslreports.com/forum/remark,14450684
M
wrote:
Hello,
I nearly forgot about this mailing list! I accidentally bought a vonage
enabled PAP2 to use on my asterisk, however it's locked and I have no
access to the admin
Can someone tell me where I can DL a windows binary for sipXezphone.
Everything I find ultimately points me back to
http://www.sipfoundry.org/sipXezPhone/ which is broken.
Tx
M
--
Opportunity is missed by most people because it is
dressed in overalls and looks like work.
Thomas Alva Edison
Hi,
can two * boxes use the same realtime database?
I know they can in terms of connecting to the same db, but it is my
understanding that the peers are created realtime as and when it
registers, in other words even of the two boxes share the same db, the
peer will only exist on the one it
Yes, you can easily use asterisk for a call center, start looking here
http://www.voip-info.org/wiki/view/Asterisk+call+queues
M
Imed Imed wrote:
Hi,
I'm a novice in asterisk.
I'm just want to know if we can develop a Call centre
application on an asterisk ?
Is anybody using the Intrado
V9-1-1 service with asterisk?
Could you share some info, setup information if so.
Thank you
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 10
Colin,
for the record I think this post was exellent and deserves a
compliment. It is probably one of the best outlines of what is needed
for a professional system I have ever seen.
Marnus van Niekerk
Colin Anderson wrote:
I concur with your approach, but "Tier 1" means
to internal extensions on * but that is a DCS
programming issue.)
Anybody used a sirrix board like the before and any pointers before we
attempt this?
Thank you
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas A
Guys,
this may be a off question but where do I start to find all the
regulations to comply with to set up a residential voip phone service
in the US?
The tech stuff I am fine with but the legal is where I need some
pointers.
Also I am not looking for a flame war just pointers on the legal
You could implement this very easily yourself.
Just write a small webpage that saves the user's find-me/follow-me
extension to a text file somewhere (or a database of course)
Then write a small agi, that checks for the file (or db value) and sets
a variable to jump to that extension
M
Has
Assuming the 456 is the ATA number and the outside number is always 10
digits.
exten = _456.,1,Dial(SIP/${EXTEN:0:3}/${EXTEN:-10},tT)
but then it might as well be
exten = _456.,1,Dial(SIP/456/${EXTEN:-10},tT)
; to dial outside thru GSM gateway
exten =
Inside the brackets for the D
flag
exten = _456.,1,Dial(SIP/${EXTEN:0:3},30,tTD(w${EXTEN:3}))
M
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) = 17.36
Thus an average of 17-18 simultaneous calls at any given time but
obviously it will peak much higher than that.
Tx
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093 patents,
Hi,
I am looking for a good affordable USA toll free DID provider for
asterisk.
Thank you
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093 patents,
including the
Also have a look at .call
files.
You web app can just create a .call file and then move it to the right
location and asterisk will place the call
No manager interface needed.
Marnus van Niekerk
"Opportunity is missed by most people because it is
dressed in overalls and looks like
.
Can anybody point me in the right direction please?
Thank you
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093 patents,
including the light bulb, phonogram and motio
Hi,
I am looking for a simple php agi script that locates a speeddial
number in a MySQL database and then dials that number.
ie.
exten = 01,1,Noop(speeddial 01)
exten = 01,2,Agi(do database lookup on 01 and finds $NUMBERTODIAL)
exten = 01,3,Goto($NUMBERTODIAL,1)
Anybody know if something
will be
appreciated.
Thank you
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093 patents,
including the light bulb, phonogram and motio
Thanx, but for the record and
archive purposes this did not work in 1.2.7.1 but it does work with
1.2.4.
Marnus van Niekerk
tom wrote:
Marnus van Niekerk wrote:
Hi,
I am currently running several * boxes on 1.0.9 with HFC chipset ISDN
modems using i4l's hisax driver
changes as possible.
Thank you
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093 patents,
including the light bulb, phonogram and motio
including my sirrix.conf and the output of some of the * Srx
commands below. Any pointers would be appreciated.
Thank you
Marnus van Niekerk
--
; global settings
[Global]
internationalprefix = 09
nationalprefix = 0
;crypto_app = /root/cvs/sirrix-pci/crypto/crypt
; external link
[out]
mode = TE
ptp
Hi,
I have a strange problem on some isdn modem channels. (* 1.0.9 /
chan_modem / 2xHFC-S cards).
Everything works fine but when the 2nd (and 3rd etc..) call comes in
and * answers and there is about a 1/2 second of sound from the
previous call (ivr) before the sound from the new call is
Thanx, but that would mean
that the system can't use NTP for timesync.
M
Paradise Dove wrote:
this is a new bug which is submitted: http://bugs.digium.com/view.php?id=6349
change your system date to an older value. everything will work again.
paradise dove
On 1/25/06, Marnus van Niekerk
I have two older Advanctage
Century Telecomms P104S phones (Sold with name ViG) that do only H323.
I am hoping I can find the firmware to ungrade these to SIP somewhere.
Tx
M
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva
I have two older Advantage
Century Telecomms ACT-P104S phones (Sold with name ViG) that do only
H323.
I am hoping I can find the firmware to upgrade these to SIP somewhere.
Tx
M
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas
the first one is easier...
A
Marnus van Niekerk schrieb:
Thanx, but that would mean that the system
can't use NTP for timesync.
M
Paradise Dove wrote:
this is a new bug which is submitted:
http://bugs.digium.com/view.php?id=6349
change your
Hi,
I set up a small system over the last couple of days and all was fine.
(* 1.2.2 - Fedora Core 1
System has 2xTDM400P with 7xFXO and 1xFXS, a couple of SIP phones
(snom320 and IP300), fax machine on the FXS channel and an IAX2 trunk
through a local provider.
All worked fine until this
this
connector to a patch panel or R11 jacks.
Also most of the docs do not mention this card - am I correct in
assuming that I have to load the wct24xxp kernel module? and which
channels will it be for two quad fxo modules 1-8 or 13-20?
Tx
Marnus van Niekerk
Thanx but why 17-24, what is the logic behind that?
M
BJ Weschke wrote:
There's an adapter on VoIPSupply that has the cable that plugs into
these cards and breaks it out into RJ11 jacks for you. Otherwise, you
can just get a telco cat3 AMP cable and a 66 block.
Yes, you need to load
Can someone give me an idea
of the processor power I will need for 1 x TDM240 with 2xquad FXO's and
8 sip phones/ATA's on a quite 100Mbit LAN.
The machine we have available of hand is a P4 1GHz with 768MB RAM.
Tx
Marnus van Niekerk
--
"Opportunity is missed by most people be
]
[mailto:[EMAIL PROTECTED]] On Behalf Of Marnus
van Niekerk
Sent: Thursday, January 19, 2006 8:23 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Processor Size
Can someone give me an
idea of the processor power I will need for 1 x TDM240 with 2xquad
FXO's
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