Here is a starting point:
http://www.voip-info.org/wiki/view/Asterisk+dimensioning
Not really what you need, but still. When you figure out something -
add here :-)
>
> Has anyone put together a public list/wiki/info sheet on what the
> various maximums/rules of thumb are? Seems a better idea th
here is the dail plan I am using:
my extensions file:
[globals]
[ext-sip]
host=provider.sip.com
[default]
exten => bob,1,Dial(SIP/${exte...@ext-sip,20)
expected dialing plan:
when some one calls bob,
Asterisk should add b...@provider.sip.com and sent to the external world.
But that is not
Hi,
a few month ago, I tried to install zaptel for my Beronet BN8S0 pci card... I
gave up and took hfcmulti/lcr. Now dahdi (2.2.1.1) seems to support the card
and I'm very interested to get it to work.
But how to get rid of these annoying qozap driver?
bishop dahdi # lspci -v -nn -s 01:00.0
01:
Thanks a lot jim for the reply.
My issue is :
there is no numbers involved. I have soft clients.
when a user (bob) calls Alex,
he just opens his sip client and types in a...@pbx.com
so how can I write the translations for a case like that?
the examples you gave are when there are numbers..can u
On 4/27/2010 8:07 PM, Richard Kenner wrote:
[sip.broadvoice.com]
...
[broadvoice]
exten => 551234,1,Set(CDR(accountcode)="44")
and Asterisk is still giving me this error in the logs (while playing a
number does not exist sound clip):
[Apr 27 18:11:19] NOTICE[12179] ch
I am not sure what your problem is. You can have a numeric extension dial an
alphabetic sip user.
exten => 123,1,Dial(SIP/somename)
The soft phone registers to your box with whatever username you set up.
If your phone can dial alpha then you can have
exten => alpha,1,Dial(SIP/$(EXTEN})
--
J
The hidden number is no different from what I posted. This is inbound, I
pick up my cell phone, dial 551234, which then hits my * box, which
then the * box barfs that error.
On 4/27/2010 8:35 PM, Peder wrote:
Is this an inbound call to that number? Or are you calling out from that
number
Hi All,
pl help me with this basic question.
I have a users (soft clients) with usernames having Alphabetics.
I want to use Asterisk as my server.
How should I have the dial plans as there are no numbers involved .
so How can I make the configuration to work ( with numbers I can get this done
On Tue, 2010-04-27 at 11:01 -0400, John Novack wrote:
>
> Anita Hall wrote:
> > Hi
> >
> > Please check out this product
> >
> > http://www.sangoma.com/products/hardware_products/data_networking/a301.html
> >
> > Does it work on Asterisk or Freeswitch ?
> > Do Telcos provide an E3 connection ?
> >
Is this an inbound call to that number? Or are you calling out from that
number? I understand the need to obfuscate the numbers, but it says " Call
from '551234' to extension '551234'", so are you calling yourself?
Or did you just change both numbers to the same number. Maybe just change
> [sip.broadvoice.com]
...
> [broadvoice]
> exten => 551234,1,Set(CDR(accountcode)="44")
>
> and Asterisk is still giving me this error in the logs (while playing a
> number does not exist sound clip):
> [Apr 27 18:11:19] NOTICE[12179] chan_sip.c: Call from '551234' to
> exten
All,
I have been fighting with my dialplan for hours now, and google
searches talk lots but offer nothing in terms of explication for this. I
have my SIP peer set up and working with Broadvoice:
[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.
all you need to do is make the configurations mirror each other.
in the example below, all of the endpoints are SIP, but it doesn't matter if
you move the endpoints to another protocol, like Fxs:
on serverA
extesions.conf:
[phones]
include => localphones
include => to_serverB
[localphones]
e
> We are running Asterisk 1.6.2.7-rc1 and SfA without problem. What
> version are you running?
I'm using the current version from the 1.6.2 SVN branch, which is
called SVN-branch-1.6.2-r258676M. I'm glad to know that 1.6.2.7-rc1 works
because that's closer to what I have than 1.6.2.6.
--
_
We are running Asterisk 1.6.2.7-rc1 and SfA without problem. What version are
you running?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner
Sent: Tuesday, April 27, 2010 9:54 AM
To: asterisk-use
I am not sure what you are asking here.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 27, 2010, at 6:16 AM, Vasiliy G Tolstov wrote:
> В Втр, 27/04/2010 в 06:11 -0700, Jim Dickenson пишет:
>> In your sip.conf your permit line does not have an ip address to allow
I'm running Asterisk 1.6.2.1 with DAHDI 2.2.1 using a TDM410P. I have
callprogress=yes in chan_dahdi.conf because, from everything I've read, it
is needed when using call files over PSTN, which I DO use occasionally.
I know that callprogress=yes is "experimental" and causes some issues.
We've nev
Yers. You have 2.5 options:
Monitor, MixMonitor, (these make 1,5) and JACK_HOOK
On Tue, Apr 27, 2010 at 5:53 PM, Jonas Kellens wrote:
> Hello list,
>
> can a conversation be recorded without the caller or callee having to press
> some combination that is defined in features.conf ??
>
> Like in qu
Jonas Kellens wrote:
> Hello list,
>
> can a conversation be recorded without the caller or callee having to
> press some combination that is defined in features.conf ??
>
> Like in queues.conf you have the ability to record a conversation with
> MixMonitor when the caller is connected to an agen
On 04/27/2010 10:41 PM, Anita Hall wrote:
> Hi
>
> Please check out this product
>
> http://www.sangoma.com/products/hardware_products/data_networking/a301.html
>
> Does it work on Asterisk or Freeswitch ?
> Do Telcos provide an E3 connection ?
>
> One of our customers had an inquiry for terminatin
- "Anita Hall" wrote:
> Hi
>
> Please check out this product
>
> http://www.sangoma.com/products/hardware_products/data_networking/a301.html
>
> Does it work on Asterisk or Freeswitch ?
> Do Telcos provide an E3 connection ?
>
> One of our customers had an inquiry for terminating 600
Anita Hall wrote:
> Hi
>
> Please check out this product
>
> http://www.sangoma.com/products/hardware_products/data_networking/a301.html
>
> Does it work on Asterisk or Freeswitch ?
> Do Telcos provide an E3 connection ?
>
> One of our customers had an inquiry for terminating 6000 calls
> simult
Hello list,
can a conversation be recorded without the caller or callee having to
press some combination that is defined in features.conf ??
Like in queues.conf you have the ability to record a conversation with
MixMonitor when the caller is connected to an agent/member of the queue.
Can th
Hi
Please check out this product
http://www.sangoma.com/products/hardware_products/data_networking/a301.html
Does it work on Asterisk or Freeswitch ?
Do Telcos provide an E3 connection ?
One of our customers had an inquiry for terminating 6000 calls
simultaneously. I want to do some homework be
Richard Kenner wrote:
> Is there an issue with running it with the latest from the 1.6.2 branch?
> I did an svn update and make install and now when somebody comes in via
> Skype, I get an infinite loop of:
>
> [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed:
> Invalid
Is there an issue with running it with the latest from the 1.6.2 branch?
I did an svn update and make install and now when somebody comes in via
Skype, I get an infinite loop of:
[Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed:
Invalid argument
[Apr 27 09:53:29] WARNING
В Втр, 27/04/2010 в 06:11 -0700, Jim Dickenson пишет:
> In your sip.conf your permit line does not have an ip address to allow the
> register from so the call is coming in as a guest and that is likely using
> context default.
>
> Set the permit line to either the ip address of the phone or the
Simply place the SIP Extension of the GSM gateway in another context
context=from-gsm
and in your extensions.conf use something like this
[from-gsm]
exten= => _X.,1,Goto(whatever IVR you want)
> Date: Mon, 26 Apr 2010 17:23:40 -0300
> From: aco1...
In your sip.conf your permit line does not have an ip address to allow the
register from so the call is coming in as a guest and that is likely using
context default.
Set the permit line to either the ip address of the phone or the network the
phone is on.
permit=192.168.1.0/255.255.255.0 with
This is probably a good idea, BUT it is likely that the dialed phone will
never ring (Perhaps that is the desired effect); In my experience it takes
Zap/DAHDI about 2-7 seconds to generate the first ring of a call.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:a
1. Why aren't you using IAX instead of SIP/FXS?
2. If you can connect from A-B using SIP, the process should be
reversible unless B just sees A as a phone and not a peer/server.
3. to make an FXS connection, you're going to have to introduce
Zaptel/DAHDI (you don't state what level o
Hi!
I need some help
Well i have this cenario:
1 ip04 running asterisk [A]
1 pc running asterisk [B]
I nedd to make calls from A to B, and B to A. Via sip
The A-B calls are working. Now I need to configure the dial plan to call B-A
either to sip numbers and Fxs.
Anyone can help me?
--
___
Hello. I'm new with asterisk. Can you help me in this:
I have cisco sip phone (601) connected to asterisk server, and 1 client
number (500).
I want to dial from 601 to 500.
But get error in cli console:
[Apr 27 15:30:15] NOTICE[9650]: chan_sip.c:20059 handle_request_invite:
Call from '601' to exte
Steve Gladden wrote:
> So that explains why it won't compile eh?
> And wow Kevin...
> I'm curious how much work would it be and would it be worth it?
> I've always imagined RT kernels would be excellent for asterisk.
> I've also wondered why it appears not to have been done 'out there'
> Or discus
Hello,
For a couple of hardphones which do not have Message Waiting Indicator, I'm
wondering what could be the most efficient and reliable way to notify users
a message is waiting.
Though messages could be sent as email attachment, I'm thinking I should
mimic cellphones behaviour like this:
5 min
when i install asterisk addon ,i got error here
chan_ooh323.c:1934: error: dereferencing pointer to incomplete type
chan_ooh323.c:1935: error: dereferencing pointer to incomplete type
chan_ooh323.c:1937: error: dereferencing pointer to incomplete type
chan_ooh323.c:1938: error: dereferencing pointe
another idea you could test is to use a very short Timeout in your Dial command.
like Dial(ZAP/012345678,1) - will dial and exit after 1 sec with
DIALSTATUS set accordingly
HTH,
Ioan
--
_
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