On Wed, Nov 25, 2009 at 11:05:52AM +, Mike wrote:
On Wed, Nov 25, 2009 at 12:26:10PM +0200, Tzafrir Cohen wrote:
On Tue, Nov 24, 2009 at 11:03:16PM +, Mike wrote:
Folks,
I've got one of those GPO 1950's rotary dial phones that I'm trying to
get working in the UK. I've got
But then you create phonenumbers in enum, which doesn't exist as
pstn-numbers.
Not the idea behind enum.
On the other hand, if you owned 10 or 100 pstn-numbers in series, you
could get the last one or two digits delegated to your dns-server.
Why do I create numbers in enum which doesn't
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Hi all,
LDAP Account Manager (LAM) is a free WebGUI to manage LDAP directories.
The next release 2.9.0 will come with Asterisk support. Big thanks to
Pavel who donated the code.
LAM homepage: http://www.ldap-account-manager.org/
Now, I am searching
Leif Neland schrieb:
But if a pstn or cell call +431123456720 will it be connected to
+4311234567 ? Or will the call fail?
If so, +431123456720 is an invalid number.
Leif
That depends on the Dialplan coding.
A non-sip call comes in from the VoIP provider into the associated
context. The
C. Chad Wallace cwall...@lodgingcompany.com writes:
So, does anyone know of a way to detect whether a call from a SIP phone
is the first step of an attended transfer or an original call?
This is impossible. At that point the phone has done this:
1) Put the original caller on hold
2) Made a
- Original Message -
From: Norbert Zawodsky
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, November 26, 2009 10:46 AM
Subject: Re: [asterisk-users] Please some enlightment on ENUM !!
Leif Neland schrieb:
But if a pstn or cell call
- Original Message -
From: Norbert Zawodsky
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, November 26, 2009 10:46 AM
Subject: Re: [asterisk-users] Please some enlightment on ENUM !!
Leif Neland schrieb:
But if a pstn or cell call
Barry,
I'm using the Asterisk GUI. When defining a User extension (menu 'user')
the only option I have is Is Agent.
The SIP extension (11) is automatically created as an agent that needs
to log in.
In the advanced options I can manually edit queues.conf and change to
member=SIP/11.
This way of
Hi guys,
Do you maybe know for a fairly good quality IAX2/SIP hard phones in up to 40
USD?
Thanks!
Regards,
Blaz
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Hi guys,
Having a little problem.How can I know where queue is my agent login
from my CDR table?
Sorry my English's terrible.
Best regards,
Daniel Stefanus
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asterisk-users
Hello!
I am trying to come up with a way to read a digit *before* the call is
answered. My Asterisk version is 1.6.2.0-rc6
SIP early media works fine (I can receive and transmit audio before the
call is answered), but as soon as I start the read application, Asterisk
answers the call which is
Hi!
So, does anyone know of a way to detect whether a call from a SIP phone
is the first step of an attended transfer or an original call?
It could probably work if you put a SIP proxy in between (ref. Kamilio).
The only way to achieve what you want is to never allow a call to a
Hi All,
I understand that new TE412P interface card comes with VPMOCT128
echo-cancellation
module. I already have a server installed with asterisk-1.2.17,
zaptel-1.2.17.1 in production with TE412P card installed already and having
VPM450 echo cancellation module and it works. I need to purchase a
This is a very broad question. why don't you tell us something more
about tyour setup?
l.
2009/11/26 Daniel Stefanus shinichikud...@gmail.com
Hi guys,
Having a little problem.How can I know where queue is my agent login
from my CDR table?
Sorry my English's terrible.
Best regards,
Hi all,
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get it back. It
goes on hold just fine. But when I press the resume button, nothing
happends.
Anyone seen this befor? Any ideas on where to start to
Hi All,
I have a query regarding timing issues on a back-to-back asterisk setup we have
here in the UK. Let me explain.
We have a 2 port Digium TE220P card, one span is configured to connect to our
ISDN30 provider (British Telecom), the other span connects to our internal PBX.
Here's the
On Thursday 26 November 2009 05:47:47 Alexander Heinz wrote:
I am trying to come up with a way to read a digit *before* the call is
answered. My Asterisk version is 1.6.2.0-rc6
SIP early media works fine (I can receive and transmit audio before the
call is answered), but as soon as I start
Hi,
I am experiencing a weird issue with my MV-372.
Mobile1 Mobile2 are both registered to my asterisk server, I am able to
use them for outgoing call with no problem, but when I call the sims in my
gateway, they are routed to the right context/extension/priority, but as
soon as I hangup, the
Hi,
A server with no previous Digium card that just had a TE420B installed had
the CPU usage of it's first CPU go from ~0% at idle (when there are no calls
and nothing else really happening) to 20% at idle (again no calls).
Is this normal, or a sign of things that should be corrected?
Hi,
Happy Thanksgiving to those of us in the USA...
Been trying to debug an AGI (in C) on 1.4.26.2. I blind transfer a call to
this snippet of dialplan:
exten = 00,1,DeadAGI(pq.agi,50)
pq.agi then plays a prompt (which I hear just fine):
[Nov 26 02:42:47] VERBOSE[28721] logger.c:
Tzafrir Cohen wrote:
On Wed, Nov 25, 2009 at 11:05:52AM +, Mike wrote:
On Wed, Nov 25, 2009 at 12:26:10PM +0200, Tzafrir Cohen wrote:
On Tue, Nov 24, 2009 at 11:03:16PM +, Mike wrote:
Folks,
I've got one of those GPO 1950's rotary dial phones that I'm trying to
I've seen that behaviour on he MV-374. One possible solution (workaround)
is to prevent the gateway from registering itself and to declare each of
the channels explicitly in sip.conf via its associated IP + port.
--
exvito
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On Wed, Nov 25, 2009 at 11:41 PM, Shaun Clark shaun_cl...@hotmail.com wrote:
Can you add an agent dynamically to a queue with an external number, i.e.
cell phone as an extension? If so how? Thanks!
Maybe adding the channel Local/PSTN-number@context-that-dials-PSTN
to the queue as a member
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