> Check out HAAST (High Availability ASTerisk) at [1]www.generationd.com
Bit out of my pricing. It must be possible to do it using downloadable
open-source.
Dan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digit
> How do you handle replicating voice mails?
I do that by putting the voicemails into MYSQL and replicating that.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live in
Hello,
Are there any guides to setting up high-availability asterisk platforms? Maybe
using Opensips.
I found this diagram, but i cant find any guides on how to go about setting it
up.
http://yfrog.com/5unetworkexampleg
Thanks
Dan
--
__
> I use realtime on 1.4 and 1.6 servers but always with rtcachefriends=yes in
> sip.conf
I already use that and it doesnt seem to re-register when a call comes in.
Only when the registration period expires, or the peer dials out.
--
__
I checked the bug reports and all I could find was similar issues with the
Asterisk 1.6 which (according to the reports) have been resolved.
I couldnt find anyone talking about 1.4 so I created a new issue and someone
closed the case and added this note:-
> This does not appear to be a bug, but
> My question is "if you are using realtime, why are you doing a sip reload?"
I said previously:-
> Let's say I add a new provider to my service and therefore have to add
> another "register=>" command into sip.conf, I'd have to issue a "sip reload"
> which would kill off all the realtime sip p
Can we not do pastebin any more?
I just received this:-
[PASTEBIN URL REMOVED] has been detected as suspicious URLs,and Quarantine to
user's spam folder has been taken on 9/20/2010 8:24:38 AM.
Message details:
Server: MADRID
Sender: d...@keshrcommunications.com;
Recipient: asterisk-users@lists.d
> Check the SIP debug and see what is going on.
> Leif.
Hi,
I checked the SIP debug.
As soon as I issue the RELOAD command, no SIP data gets transferred to the
phone.
Asterisk output: http://pastebin.com/FB675N16
Any ideas how I can do a SIP reload without losing the Sip Phones registration?
> Check the SIP debug and see what is going on. Alternatively you could turn
> off
the qualify option with qualify=no.
I'll take a look at the sip debug, but qualify needs to stay on, so thats not
an option.
--
_
-- Bandwidt
Is there any development work being done on the realtime addon? Theres been no
updates since April.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webi
> Have you checked the Issue Tracker
Not yet. I wanted to see if it's just me before searching through/raising a bug
report.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us f
> Noted - but if OP does a "reload" once a day, 120 seconds (2 minutes) out of
> 1 day (14400 seconds) is 99.17% uptime; "close enough" to 99.999 percent in
> most folks books. What percentage of businesses use their phones 24/7?
Even if its once a month, it's still too much in my book. No wonder
> As someone else said, the answer is
"don't do a 'reload'", do an "extensions reload" or whatever it is specific
to your changes.
You are correct. I'm just being lazy. But I'm just worried that some time in
the future, I'll have to reload the sip config, and therefore flush out all the
realtime
> A reload flushes the SIP registration database, so once you do a reload,
that phones reg is gone.
Finally an answer that seemed more realistic. But it doesnt explain why the
phones that are hard coded in the sip.conf file don't lose registration.
Any ideas?
Thanks
Dan
--
__
> That's not a bug. Only when the phone registers or performs some sort of
> action
> (such as placing a call, etc...) does Asterisk query the database. If your
> phones have a short re-registration time this becomes less of a problem.
How do you explain that as soon as I issue a "reload" comma
Hi,
I'm using the CallTime and a few other variables to name a recording so that I
can then take the wav file name and see when it was recorded, and what the
recording contains.
However, since ${CDR(start)} contains a space in part of the date, the filename
becomes corrupted when I use samba a
> By reload you mean "sip reload" or just any reload in general?
Reload in general.
It might be an issue only with the Polycom sip phones. Not been able to test
any others. I'll try a software phone tomorrow.
--
_
-- Bandwidth
Hi,
I think ive found a bug but need someone to double check.
Whenever I issue a "reload" in Asterisk, any realtime extensions stop receiving
calls.
I have to reboot the sip phones in order to get them to re-register.
Can anyone see if they have a similar problem?
Asterisk 1.4.32
Mysql realti
On 15/09/10 12:14, Rob Fugina wrote:
It is with deep sorrow and broken heart that am sending you this mail. Am in
deep need and my situation is lamentable. my family and I decide to come
visit Wales,United Kingdom for a short vacation. To our greatest dismay we
were attacked and ripped apart
I've managed to get 'info' working. Not sure why the others didnt want to work.
Thanks for your reply.
Dan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introducto
Is there any way to prevent the end user hearing the *1 key tones when the
touch recording is activated?
Thanks
Dan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Hi,
It seems ive broken my settings and now, asterisk isnt detecting my DTMF tones.
What kind of diagnostics can I do to work this out?
I've set the extension in sip.conf to everything listed on this page but no
result. I've also played around with the settings on the phone with no help
either
RWC also wont show me how long I have to wait before I can restart.
And while RWC is executing, will idle lines be able to make a new call, or will
they have to wait till the restart is complete?
If they have to wait, then I prefer manually restarting asterisk when I see a
moment of quiet time.
-
> You should be doing "core show channels", even if all of your channels are
> indeed SIP.
Thats great thanks.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introd
Hi,
I'm trying to view a list of the active calls to see if I can restart Asterisk.
When I do 'sip show channels', I get a huge list like this (just a sample
pasted):-
92.110.7.210 (None) 198827f2469 00102/0 0x0 (nothing)No
Init: OPTIONS
92.110.7.210 (None)
Actually, ignore my last email. Forgot to reload.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk
> ${UNIQUEID} is going to be realtivly unique certnely in the short term
I dont understand something. When I do ${UNIQUEID}, I get something like this:-
"SIP/215.166.5.140-0bbf"
Is this correct? Its not a valid file name.
Thanks
Dan
--
Hi,
Im looking at using MixMonitor to record calls and I know that I need to set
the filename first.
However, with the number of calls coming in, hard coding the filename isnt an
option.
So I need to do something like this:-
MixMonitor(RANDOMNUMBER.wav)
But can't find a way to generate a ran
> I have our recordings written to a solid state drive rather than straight to
> storage disks then moved to long term storage to avoid this problem.
Not an option for me at the moment.
I'm running Asterisk on a cloud to reduce startup costs.
Once I reach around 1,000 extensions, I'll move over
How do you sort out the issue of having 2 wav files per call?
Also, when I press *1, asterisk thinks that both the caller and the callee have
pressed *1 and therefore it starts recording twice (therefore making 4 wav
files). Any idea what's going on there?
Heres the CLI output:-
-- Called
> 1) I want to create add *1 call recording and wanted to know whether the file
> is created during recording or only after? I want to syncronise the
> recorded files with my web server (on a different machine (Windows)) so I
> need a way of telling when the recorded call has ended before copyin
Hi,
1) I want to create add *1 call recording and wanted to know whether the
file is created during recording or only after? I want to syncronise the
recorded files with my web server (on a different machine (Windows)) so I need
a way of telling when the recorded call has ended before cop
> Yes, after I can pick it up from my phone (9133i), and it works. I had
> verbosity at 6 at the moment of testing. When he enters 701, only his phones
> displays "Failed", nothing in asterisk. I can pickup after that on mine.
Is there a dial plan on the phone that you need to alter?
--
__
> in asterisk 1.6.x there is a Dial option
Sorry, any solutions for Asterisk 1.4?
Thanks
Dan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar ev
Hi,
I'm currently programming an interface for my Asterisk service.
I've noticed an issue if someone sets up call forwarding on their sip phone.
Asterisk receives a 302 "Moved Temporarily" message, and forwards the call
successfully.
However, the CDR is not correct.
If I set up call forwarding
> So one shows as answered and the other doesn't?
Correct.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://ww
> I had a similar problem and as far as I know, the asterisk server doesn't
> know which of those numbers has answered your call.
> If anyone knows any different, I'd like to know as well!
Got it!
I created a context that contained this:-
[outgoing_context]
exten => _X.,1,Dial(SIP/${ext...@supp
Hello,
I've posted about this a few months back but I didn't understand the answer
properly and only just got round to sorting it out.
My question is, when I dial out to a few numbers at the same time, the CDR
lastdata for the call looks like this:-
SIP/07957543...@supplier&SIP/07957123...@sup
Looks like I have to go down that route.
Ok, thanks
Dan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.
> Is your main concern changes being made to the extensions.conf or
> someone having to manually make changes to the extensions.conf?
Someone having to manually change extensions.conf and then reload asterisk.
--
_
-- Bandwidth
> I think you asked this question earlier and there were good responses to it.
> There is nothing more to it than what people already suggested.
I think if you read my question properly, you'd see that I have one existing
context (CLIENT1_PHONES) and I want to INCLUDE a number of other contexts
Sorry, that was not me.
Dan
> I think you asked this question earlier and there were good responses to it.
> There is nothing more to it than what people already suggested.
--
_
-- Bandwidth and Colocation Provided by http://w
Hi,
I think I already know the answer to this question, but is there any way to do
the following using realtime? Or do I have to create a full dialplan for each
client without using includes?
[client1_phones]
include => client1_internal
include => client1_outgoing_calls
include => test_calls
in
> Un-top-posting...
Sorry, been a while since I posted and forgot that Outlook top posts by design.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory web
ilto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: 17 August 2010 15:10
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Create File Directory
Hello,
I have to following dial plan.
exten => 5551234,1,Answer()
exten =>
5551234,n,Read(ACCOUNTNUMBER|/var/lib
Hello,
I have to following dial plan.
exten => 5551234,1,Answer()
exten =>
5551234,n,Read(ACCOUNTNUMBER|/var/lib/asterisk/clientsounds/company/recordingsystem/welcome_accountnumberplease)
exten => 5551234,n,Read(MENUNUMBER|/var/lib/asterisk/clientsounds/ company
/recordingsystem/menu_number_ple
Hi,
I'd like to be able to create contexts in real-time when I add new clients to
my asterisk box.
Currently, I have to create a blank context in extensions.conf and add:-
switch => Realtime/@
Is there any way to avoid the step of creating the blank context and simply
include all the entries
Hi,
I have the same setup as you.
I didn't bother mapping any ports.
I just enabled nat and keepalive.
Here is a screenshot of the config on the phones. For some reason, on some
phones I had to turn the NAT Mapping Enabled to Off otherwise call transfer
didn't work.
http://www.postimage.org/i
Hi,
I'm going abroad shortly and want to be able to dial into asterisk and get it
to call me back so that I can make an outgoing call through my voip provider,
rather than paying crazy international rates.
Can anyone point me in the right direction with regards to the dialplan?
Im using Asteri
> do i need to have an smtp server somewhere. i tried directly from my
> dialplan but no joy! i know you know that i am not a star with this
> but any help would be cool
Hi,
You should check, but I think some carriers don't have email to sms as part of
their service, in which case I use an sm
> - you could also consider the M() option to Dial together with the CDR
> userfield for logging whatever channel variable make sense to you
I'll see if I can sort it out with that.
> - have you looked at the destination channel in the CDR?
The destination channel says:-
SIP/sipprovider-00
>> I am diverting an incoming call to a mobile phone and a landline using the
>> following:-
>>
>> exten =>
>> 020300,3,Dial(SIP/44208...@sipprovider&SIP/4470...@sipprovider,120,r)
>>
>> For billing purposes, i need to be able to work out whether the diverted
>> call was answered by the
Hi,
I am diverting an incoming call to a mobile phone and a landline using the
following:-
exten =>
020300,3,Dial(SIP/44208...@sipprovider&SIP/4470...@sipprovider,120,r)
For billing purposes, i need to be able to work out whether the diverted call
was answered by the mobile or whe
14:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls Dropping
On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote:
> How can i log a continuous ping test to a file and include the date
> and time of each ping?
Try this:
#!/bin/
> my advise check your internet connection on the remote location and keep a
> ping from that network to your server running all the time to check for time
> outs.
How can i log a continuous ping test to a file and include the date and time of
each ping?
I've found this bash code but it only lo
Hi,
I'm having a major problem with random calls dropping. After spending weeks
trying to figure it out, i've finally spotted the issue but don't know how to
resolve it.
I run a sip server that's hosted in a data centre. It has a public IP address
with no nat involved. My provider also has a p
On Thu, 29 Apr 2010, David Backeberg wrote:
> I'm considering a situation where I buy about twenty ATA devices.
>
> I've played with the Linksys / Cisco PAP2T, and got that working fine
> with some inbound and outbound faxing. The web GUI was okay. I'm
> seeing prices around $45 to $50 for this th
Look at option A(x) on this page:-
A(x): Play an announcement (x.gsm) to the called party.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Dial(SIP/11,mA(soundfile))
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mickael MO
cro-screen]
exten => s,1,Wait(0.2)
exten => s,n,Read(ACCEPT|priv-instruct-custom|1)
exten => s,n,GotoIf($[LEN(${ACCEPT}) < 1 ] ?no)
exten => s,n,GotoIf($[${ACCEPT} = 1 ] ?yes:no)
exten => s,n(no),Set(MACRO_RESULT=CONTINUE)
exten => s,n(yes),Wait(0.1)
___
Hello,
I use the following macro to screen calls when they come in.
Priv-instruct-custom says "press 1 to accept, press 2 to reject"
However, when no input is made (or the call goes to my mobile's voicemail and
therefore no input is made), the result is that the ACCEPT variable is not set
and
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail Remote Access
Dan Journo wrote:
>
> Hi,
>
> Any ideas?
>
>
I'd be helpful to see the console output.
Doug
--
_
-- Bandw
Hi,
I'm trying to set up remote voicemail pickup. I've created the following
dialplan, but when I press *, I am not sent to voicemailmain. The unavailable
message just continues to play as normal.
exten => 234555,1,Set(MAILBOXID=1)
exten => 234555,n,Set(MAILBOXCONTEXT=company3)
exten =>
em to copy
a blank audio file into place before calling Dial so that Asterisk
thinks the caller has already recorded their name.
On 3/17/2010 1:42 PM, Dan Journo wrote:
> Thats similar to how I want it to work, however I dont want the caller to
> have to give their name (even the first tim
Thats similar to how I want it to work, however I dont want the caller to have
to give their name (even the first time they call)
Is there any way of using the p option of the dial command, but totally remove
the caller name recording feature?
Thanks
Dan
-Original Message-
From: asteri
Hi,
I would like to develop a dialplan that allows the callee to reject the call
like this:-
1) Call comes in and receives a greeting and get put into a queue.
2) A second call is placed to the member of staff (SIP phone or mobile phone)
3) The member of staff answers the call and is presented w
terisk Redundancy
Just do something like Dial(SIP/asteriskbox1&asteriskbox2/{$EXTEN})
On Sun, Mar 7, 2010 at 1:46 PM, Dan Journo
mailto:d...@keshercommunications.com>> wrote:
Hi,
Sorry, I replied to the wrong email.
Heres the question
If I set up two servers for load balancing and
Sun, Feb 14, 2010 at 11:42 AM, Dan Journo
wrote:
> Hello,
>
>
>
> My host just had a faulty power supply and therefore, my Asterisk server was
> down for 7 hours.
>
> It was a Sunday so no one was making calls, however if it happened during
> the week, I'd have proble
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: 06 March 2010 20:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Denial of Service Attack
That solution works fine for the polycom phones because you can set
Dan Journo wrote:
> I currently have a dedicated server with a hosting provider for my voip and
> the provider is currently experiencing a DOS attack. I have been looking at
> purchasing a number of servers and creating my own VOIP setup with
> redundancy built in.
>
> However, ho
Hi,
I currently have a dedicated server with a hosting provider for my voip and the
provider is currently experiencing a DOS attack.
I have been looking at purchasing a number of servers and creating my own VOIP
setup with redundancy built in.
However, how I can design the system to ensure serv
I agree that better hardware is needed.
I'm looking into buying my own servers and getting a rack in a data centre.
I'll impliment a redundancy solution at the same time.
Thanks for the links.
Dan Journo
Kesher Communications Ltd
-Original Message-
From: Steve Totaro
Sent: 1
Hello,
My host just had a faulty power supply and therefore, my Asterisk server was
down for 7 hours.
It was a Sunday so no one was making calls, however if it happened during the
week, I'd have problems.
I was trying to find a whitepaper or advice on how to set up two Asterisk
servers to prov
I was recommended Polycom phones. I tested some. And now, I LOVE them.
Look at the Polycom IP321.
It's a great phone with provisioning and two lines. Dont know about G729, but
I'd be surprised if it didn't support it.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun..
Set up the CDR.
http://www.voip-info.org/wiki/view/Asterisk+cdr+csv
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: 09 February 2010 10:10
To: Asterisk Users Mailing List - Non-Commercial
I've never seen that in Outlook. What client do you use?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francesco
Peeters
Sent: 07 January 2010 18:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Go to this address for information on how to remove yourself:-
http://lists.digium.com/mailman/listinfo/asterisk-users
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Dean
Sent: 07 January 2010 15:50
To: A
t: 30 December 2009 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Parked Call Ringback
Dan Journo wrote:
>
> Hello,
>
> I have enabled call parking and it works great.
>
> However, when the "hold time" hits the "parkingt
Hello,
I have enabled call parking and it works great.
However, when the "hold time" hits the "parkingtime", the extension that
parked the call is called back.
The problem is, if that extension does not pickup the returning call, the
call gets dropped.
Is it possible to get Asterisk to
I recommend you follow the detailed install guide in this book and install all
the required support programs etc.
http://downloads.oreilly.com/books/9780596510480.pdf
Thank you for contacting Kesher Communications Ltd.
IT Maintenance Clients can now receive a fas
Do you have any error logs? What output do you get when you try "make install"
with the asterisk package?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
] Calls Dropping
The info you need is here
http://www.voip-info.org/wiki/view/Asterisk+config+logger.conf
Ish
Dan Journo wrote:
>
> Hello,
>
>
>
> We have a problem that calls seem to be dropping for no reason.
>
>
>
> Is there any way to write a debug log to dis
Hello,
We have a problem that calls seem to be dropping for no reason.
Is there any way to write a debug log to disk so that I can check it as soon as
a call is lost?
It happens randomly once or twice a day to different callers.
Many thanks
Dan
___
-
I had this problem.
I was told by someone on this list to put this into the General section of
sip.conf
session-timers=refuse
Seems to have worked.
Dan
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of
: [asterisk-users] Call Limits
On Sun, 2009-12-06 at 08:49 -0500, Dan Journo wrote:
> I’m trying to figure out how to limit the number of concurrent calls a
> client can make.
I prefer to use the GROUP() and GROUP_COUNT() dialplan functions to
enforce arbitrary call limits in Asterisk
--
Jared
tus" but it's not in any of the menus..
Many Thanks
Dan Journo
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I agree it would be easier. However that would mean allowing them to make up to
12 concurrent calls.
I've been asked to reconfigure Asterisk so that we can bill the client per
outgoing channel.
For example, if the client has 2 outgoing channels, and 10 extensions, they
should only be able to mak
one of the 2 other calls
to end.
I thought that maybe one way would be to duplicate the outbound sip settings
and label them "outbound_client_1" and then use call-limit within that.
Has anyone got any experience of this?
Thanks
D
Thanks for your replies.
Am I correct that if I use "session-timers=refuse", asterisk will never
disconnect a call?
That could be quite expensive if a call gets lost.
Any idea why the line I am using in the dialplan isn't working?
Thanks
Dan
Dan Journo wrote:
>
> Hi,
>
Hi,
I have a problem with incoming calls. They all seem to be ending after 600
seconds (10 minutes).
I've added:-
exten => _X.,2,Set(TIMEOUT(absolute)=18000)
However the calls seem to still be ending after 600 seconds.
I've checked the debug and verbose and the line above is being executed.
Is t
Sorry for the repetition.
I didn't see the other responses.
-Original Message-
From: Thomas Kenyon
Sent: 02 December 2009 07:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about g729
Tilghman Lesher wrote:
> On Tuesday 01 Decembe
However, I've read somewhere that passthrough doesnt require a license. Which
means that if your sip clients can transmit in g729 and your voip provider can
receive in g729, your asterisk server won't need to do any encoding and
therefore doesn't need any licenses. It is simply passing the data
You pay per channel. Which I believe to mean, if you have 10 sip clients but
only 2 clients make calls at the same time, you only need 2 licenses.
You only need to purchase 10 licenses, if all 10 clients will be making calls
at the same time.
I hope that makes sense. Maybe someone can explain it
/view.php?id=14426 - link to the issue
Hope that helps.
Dan Journo
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ABBAS SHAKEEL
Sent: 25 November 2009 09:59
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users
Usually, hackers give out the details they discover so that thousands of
people use the stolen details, and therefore its impossible to detect
which user is the actual hacker.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Sorry for causing this war. It's just, if everyone sent private
messages:-
a) there would be no point of the mailing list
b) our mailboxes would fill up in minutes, leaving no space of our
business emails.
Dan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:aste
r you.
-Original Message-
From: aster...@opensourcesolution.in [mailto:aster...@opensourcesolution.in]
Sent: 13 November 2009 09:18
To: Dan Journo
Subject: hi Dan
Hi dan,
sorry for sending u personal mail. i am a beginner in asterisk, i had
configured a minimum dial plan in which i had mad
CallWaitingRing to ring the phone anyway.
I do this with Polycom 501s.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, November 12, 2009 6:24 AM
To: Asterisk Users Mailing List
Am I correct in saying that the without allowguest=no anyone can connect and
make calls through the default context?
If allowguest is set to no, how can I ensure that incoming calls can still be
received from our DDI supplier?
Many Thanks
Dan
-Original Message-
From: asterisk-users-bou
Hello,
I have Asterisk set up with 6 extensions. When a call comes in, I use a
Dial command to call all the extensions together until someone picks up.
The problem is, when there is an incoming call and an extension is in
use, if the extension puts down the phone while the incoming call is
Hello, I am having a problem with getting call transfer to work.
This is what is happening:-
1) External call comes in on SIP from a DDI provider
2) The call is answered by extension 204
3) Then extension 204 presses the Xfer button and the call is
placed on hold
4) E
101 - 200 of 272 matches
Mail list logo
