2009/8/4 Faheem faheem_...@yahoo.com
how to implement CLONED LINE Feature in asterisk
Hey, I want to implement Clone Line feature in asterisk. I am using
SPA-2100.
The feature should work in this way.
There are two ports in the SPA-2100 both are registered with asterisk with
same
On Tuesday 04 August 2009 01:29:52 am Pascal Maugeri wrote:
Hi
Is there anybody here who has tried to interface Asterisk with PMR446
system (http://en.wikipedia.org/wiki/PMR446) using the native EM interface
?
One way is the RoA (Radio over Asterisk) project
2009/8/5 Mike asterisk-us...@norgie.net
On Tue, Aug 04, 2009 at 03:35:22PM -0500, Doug Bailey wrote:
This code is designed to handle Message Waiting Indication (MWI) incoming
on FXO
line. This data could very well be embedded in your CID spill as part of
an
MDMF message that also
Question:
Naturally there are times when need to I reformat an extension in a context as
such:
;Reformat add CC1
exten = _NXXNXX,1,Goto(1${EXTEN},1)
-or-
;Reformat 011 with with +CC
exten = _011X. ,1,Goto(+${EXTEN:3},1)
It's a helpful trick, BUT there are times when I want to send
You could just store the initial value of ${EXTEN} in another channel
variable before doing any postprocessing on it. You can then refer to
the original received DNIS whenever you like elsewhere.
Karl Fife wrote:
Question:
Naturally there are times when need to I reformat an extension in a
On 04/08/09 23:57, Miguel Molina wrote:
Edwin Quijada escribió:
It depends about your traffic.
But myabe and I guess Core 2Duo 4Gb ram , Sata 160Gb
+_
It's pretty well known that asterisk is CPU intensive, not RAM
intensive. It think 4GB is much more than enough. BTW, if your asterisk
By placing OPENSIP in front of Asterisk, we can register multiple
accounts, and we can successfully make call for Outgoing only. But in
case of incoming it fails.
If two users are registered with asterisk or OpenSIP then the user that
is registered latest is considered to be valid, and he is
Karl Fife a écrit :
[...] there are times when I want to send the call to another context in its
original un-reformatted state. Naturally the ${EXTEN} variable has been
changed. It occurred to me to use CALLERID(DNID) as such:
exten =
Kayton Sapale a écrit :
Hi all,
HI alone :-)
Thanks to the previous replies that helped me with this before, but I
got side-tracked in the middle of trying to figure this out, so
apologies for posting the same issue. I use a Nokia e71, with an
asterisk server and am having an issue
Rob a écrit :
Hi all,
Hi
I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a
while and it works fine I just added CALL OUT ... I have no problem
with call setup ... the called party hears me ... but I can't hear them
again if the call comes INTO the server
Miguel Molina schrieb:
Klaus Darilion escribió:
Hi!
I just found out that Asterisk (1.4) does not write CDRs if the incoming
call was not forwarded but handled internally without answering the call.
E.g.:
[from_pstn]
exten = 997,1,Answer()
exten = 997,2,Playback(tt-weasels)
exten =
Hi All,
I want to make an SS7 connection using latest available Asterisk IPv6 build.
when I try to compile Asteriskv6-20080107 build with lib-ss7 1.0.2 version I
get an error saying SS7_TRANSPORT_ZAP is not found.
If I search for SS7_TRANSPORT_ZAP in libss7.h file, I donot find it.
FYI: I checked the sources and Asterisk does write CDRs only if the call
in answered locally or forwarded to an outgoing channel.
Thus, as workaround I wrapped the extensions behind Dial(Local/...)
regards
klaus
Klaus Darilion schrieb:
Hi!
I just found out that Asterisk (1.4) does not
Hi,
Vyatta Asterisk works fine here. We are using traffic shaping DynDNS and NAT.
Bye,
Patrick
On Tue, Aug 4, 2009 at 2:27 PM, Barry L. Klineblkl...@attglobal.net wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tarek Sawah wrote:
First of all it acts like a firewall and a router..
Hi list,
via the network of my VoIP-provider there is a possibility to send an
sms.
Now, I know that Asterisk can interact with an external
sms-module/gsm-gateway, but how does one let Asterisk send an sms via
the IP-network ?
Like the Dial()-application, is there a similar way of letting
Hello
I have few questions :
- what's the difference between a subscribe request et a register request ?
- in asterisk 1.6 allowguest=yes or no param does it work ? if yes, please
someone could explain how doest it work because I think i'm a little bit
confuse.
- if I configure a sip terminal in
Yes ... as a matter of fact here is the sip.conf ... obviously private info
removed
[general]
register =
1747xxx:x...@proxy01.sipphone.com1747xxx%3ax...@proxy01.sipphone.com
port = 5060
bindaddr = 192.168.22.5
context = incoming
svrlookup=yes
;dtmfmode=inband
On Thu, Jul 30, 2009 at 8:50 PM, John Todd jt...@digium.com wrote:
I know many of you have been waiting for this for a while, so I'll
keep this short: The Skype for Asterisk Public Beta is now available
on the Digium store.
We are pleased to announce the open beta of Skype For Asterisk is
Hello,
well let me explain one part of your question, the host parameter. if
you want to restrict the access to one ip you can say it here.
host=192.168.2.13 means, that you can only use this account from
192.168.0.13, eg. for security reasons. i recommend so set it to
dynamic at the moment and
hi
just donwloaded the 1.6.1 branch and made configure install. so far so
good. after staerting asterisk with:
asterisk -cr
Could not load features.conf
== Registered application 'ParkedCall'
== Registered application 'Park'
== Manager registered action ParkedCalls
== Manager
I do not use Gizmo for inbound, only out. I have a register line that
looks like yours. In addition I have this:
[general]
context=nonesaid
allowguest=no
allowoverlap=yes
allowtransfer=yes
realm=my system's host name
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
maxexpiry=3600
minexpiry=60
On Wed, Aug 05, 2009 at 02:13:01PM +0800, D Tucny wrote:
The problem is that your mailbox line was below channel=1, as such, it
applied to the next channel, channel=3 not channel=1...
d
Nice one. Thanks for spotting that.
Mike.
signature.asc
Description: Digital signature
tom schrieb:
hi
just donwloaded the 1.6.1 branch and made configure install. so far
so good. after staerting asterisk with:
asterisk -cr
Could not load features.conf
== Registered application 'ParkedCall'
== Registered application 'Park'
== Manager registered action
;-)
thx
On Wed, Aug 5, 2009 at 10:56 AM, Christian Victor
christ...@victormedia.dewrote:
tom schrieb:
hi
just donwloaded the 1.6.1 branch and made configure install. so far
so good. after staerting asterisk with:
asterisk -cr
Could not load features.conf
== Registered
Thanks Daniel. It looks like I didn't paste everything into the email,
but not sure if this will make a difference:
What I saw in debug with the device that does not work:
Found peer '104'
What I saw in debug with a device that does work:
Found peer '103' Found RTP audio format 96 Found RTP
When I try that number I get a message on the device:
Connection time-out
I get the same message for other local numbers also.
Message: 13
Date: Tue, 4 Aug 2009 16:22:11 -0500
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] Calling issue for non-extension numbers
To:
Kayton Sapale a écrit :
Thanks Daniel. It looks like I didn't paste everything into the
email, but not sure if this will make a difference:
No need to send agian the same datas, I cutted non relevant part in my
answer.
From your other mail I'm sure that your problem is dialplan related.
I have in my sip.conf the following
[jon.moore]
type=friend
mailbox=8100,8150
In voicemail.conf, both mailboxes are defined.
On my Aastra 480i phone, I only see the first mailbox
listed. I've verified this, by changing mailbox= to
reverse the order, and I then see 8150 when I go to
On Wed, 2009-08-05 at 13:12 -0500, Jon Moore wrote:
I have in my sip.conf the following
[jon.moore]
type=friend
mailbox=8100,8150
In voicemail.conf, both mailboxes are defined.
Have you tried 81008150 (using an ampersand instead of a comma)?
--
Jared Smith
Training Manager
Hi all,
For a while now I've been using Asterisk together with HFC-PCI cards
(Cologne chipset) for Euro-ISDN BRI support. However, I do not
consider this to be the most reliable solution and believe that the
most stubborn problems have always been software related.
If my clients are
On Wed, 2009-08-05 at 14:32 +0200, harry R wrote:
- what's the difference between a subscribe request et a register
request ?
A subscription in the SIP protocol is saying Hey, I'd like to be
notified when something happens. This is most often used when a phone
wants to subscribe to the state
my problem is this. I have google forward the call to gizmo5. I have this line
in my sip file :
register = user:passw...@proxy01.sipphone.com
I believe this lines connects asterisk with gizmo5 so when it gets a call from
Google, asterisk will answer it?
At the end of my sip file i have this
On Wed, Aug 5, 2009 at 1:47 PM, Jared Smithjsm...@digium.com wrote:
Have you tried 81008150 (using an ampersand instead of a comma)?
Just changed it. Reloaded asterisk and restarted the phone. Same behavior
as before. Well, only a single mailbox shows up anyways.
-jonathan
Jonathan Moore wrote:
Just changed it. Reloaded asterisk and restarted the phone. Same behavior
as before. Well, only a single mailbox shows up anyways.
Add the @context on each of the mailboxes:
mailbox=8...@yourcontext,8...@yourcontext
Doug
--
Ben Franklin quote:
Those who
On Wed, Aug 5, 2009 at 2:55 PM, Doug Lytlesupp...@drdos.info wrote:
Jonathan Moore wrote:
Just changed it. Reloaded asterisk and restarted the phone. Same behavior
as before. Well, only a single mailbox shows up anyways.
Add the @context on each of the mailboxes:
Just to update on my troubles.
I noticed the MWI light wasn't coming on when I received a new message, so
I removed the mailbox= from sip.conf and added just a single mailbox. Now,
my sip.conf looks as follows..
[jon.moore]
type=friend
mailbox=8...@default
And I do get the message
I might even be willing to try out a more expensive PRI card if I knew
it also supported BRI: just as long as I would no longer have to worry
about the software support for it -- for both Asterisk 1.4 and 1.6.
Thanks,
Jaap
You can use Sangoma Media Gateway along with Asterisk (
I'd suggest using different user names and getting asterisk to handle the
cleverness... And, well, doing it this way is pretty simple, straight
forward, basic asterisk functionality...
Trying to get two different instances registered as the same user, is, as
you've found out, not going to be
Klaus Darilion wrote:
FYI: I checked the sources and Asterisk does write CDRs only if the call
in answered locally or forwarded to an outgoing channel.
Thus, as workaround I wrapped the extensions behind Dial(Local/...)
regards
klaus
Klaus Darilion schrieb:
Hi!
I just found out
On Wednesday 05 August 2009 15:13:47 Jonathan Moore wrote:
Just to update on my troubles.
I noticed the MWI light wasn't coming on when I received a new message, so
I removed the mailbox= from sip.conf and added just a single mailbox. Now,
my sip.conf looks as follows..
[jon.moore]
On Wed, Aug 5, 2009 at 4:05 PM, Tilghman
Leshertilgh...@mail.jeffandtilghman.com wrote:
Are you using plaintext storage, ODBC storage, or IMAP storage for your
voicemail messages?
Plain storage. My voicemail.conf is just about the same as the sample
config that's installed, with
the expection
Greetings again List.
I'm facing a strange case with one of the productive Asterisk servers..
i have 3 providers sending traffic to the call center where agents pickup the
calls.
calls come into the server Queue Agents
Last October .. an undersea cable got disconnected placing Egypt and the
We use Patton BRI gateways. No problems so far.
If possible, we prefer to keep telephony interfaces out of Asterisk box.
Regards
Jorge
Jaap Winius wrote:
Hi all,
For a while now I've been using Asterisk together with HFC-PCI cards
(Cologne chipset) for Euro-ISDN BRI support. However, I do
On Wed, Aug 5, 2009 at 5:28 PM, Jorge Mendoza mend...@tcc.com.pe wrote:
We use Patton BRI gateways. No problems so far.
If possible, we prefer to keep telephony interfaces out of Asterisk box.
Regards
Jorge
Just for the record, Sangoma Media Gateway does exactly that, leave all your
PSTN
Quoting Jorge Mendoza mend...@tcc.com.pe:
We use Patton BRI gateways. No problems so far.
If possible, we prefer to keep telephony interfaces out of Asterisk box.
What a great idea! I'm going to remember that. Unfortunately, I
believe that would be of no use if you also wanted to use your
At 11:24 PM on 31 Jul 2009, Emrah wrote:
Doug,
Thanks for the suggestion.
I know there are plenty of workarounds there, I am not asking how to
do it because I know how to do it too.
What I am saying is that it could be an embedded feature in the
Voicemail application, like the recent
Please any one help for this problem.
-- Forwarded message --
From: velusamy velu velu.techni...@gmail.com
Date: Mon, Aug 3, 2009 at 10:22 AM
Subject: User Authentication in sip.conf
To: asterisk-users@lists.digium.com
Dear all,
I want to setup the incoming calls, that
Jaap Winius wrote:
Quoting Jorge Mendoza mend...@tcc.com.pe:
We use Patton BRI gateways. No problems so far.
If possible, we prefer to keep telephony interfaces out of Asterisk box.
What a great idea! I'm going to remember that. Unfortunately, I
believe that would be of no use if you
On Fri, Jul 24, 2009 at 08:28:48AM -0500, Danny Nicholas wrote:
Here's how I think your dialplan should look:
exten = 101,1,Ringing
exten = 101,2,Answer()
exten = 101,3,Dial(SIP/quentin,10)
exten = 101,n,VoiceMail(1...@default,u)
exten = 101,n,Playback(vm-goodbye)
exten =
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