No, it will not have "and" if You just put and.ulaw. You should
correct file say.conf - there are rules how to read numbers, and You
should add "and" there if You want to hear it.
2012/7/23 נפתלי מאיר :
> It`s not will to be: ; "one - thousand - two - hundred - and - thirty - four
> ??
>
> I put a
If you properly link users.conf to sip.conf you can use it it there too.
2012/6/12 Rabary :
> Hi list,
>
> I want to use rtptimeout function on asterisk 1.4 but any docs I read, it is
> said that I need to configure it in sip.conf file,
> But can I use rtptimeout in users.conf file or do I need to
Hi
As far as I know "Alert-Info" is as far as vendor specific extension
to SIP used by CISCO VOIP-gateways only. Didn't noticed any other
vendors to support that. Software clients neither. So such trick is
only usable in conjunction with CISCO.
Anyway, wait another answer, probably somebody knows m
2012/5/3 JIMMY GATHAGE :
> I am using a SIP trunk to make outgoing calls. Outgoing calls are
> going through okay. I am using the AMI to Originate a call. The
> channel is not returning any event when the phone on the PSTN is
> ringing. How can i detect the phone ringing on the SIP channel?
It is
2012/4/18 Matthew Jordan :
> I imagine that this is the case, as ASTERISK-19601 noted that
> when this situation occurs, the NOTICE message indicates that
> there is a failure to match the extension, as opposed to a failure
> to match an allowed domain.
Yes, it was hell to detect real error cause
2012/4/17 Danny Nicholas :
> Maybe it needs to be _4001020?
>
Not, it doesn't. Actually I have traced this incoming call step by
step. Real reason it refuses - wrong domain. But why it wrong - have
not any idea.
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-- Bandwidth
Hi
Have an asterisk. Setup a couple of friends.
Sip.conf - http://pastebin.com/zUgiYbBi
Trying to make incoming call, and have such error(cli output)
http://pastebin.com/zFfgYcNR
NOTICE[4994]: chan_sip.c:23316 handle_request_invite: Call from
'RMT20' (192.168.8.1:5062) to extension '4001020' reje
2012/2/20 M Takahashi :
> Is anybody running multiprocess of Asterisk on a server ? Does it work well?
> My configuration is too complicated. I know Asterisk on a virtual machine
> works well. but OS overhead is considerable. that is why I want to divide a
> process.
Running 3 instances of Aste
2012/2/4 Steve Edwards :
> does DO-SOME-ACTION-BASED-ON-PREVIOUS-CALL-RETURN-CODE involve some
> interaction with Asterisk?
Yes, DO-SOME-ACTION-BASED-ON-PREVIOUS-CALL-RETURN-CODE interacts with Asterisk.
> Does the entire ${PROFILE_MUSIC} file need to be played or
> does it need to be interrupted
2012/2/3 Danny Nicholas :
> In my PERL AGI I use
> - print STDOUT "SET MUSIC ON HOLD DEFAULT\n";
> - print STDOUT "SET MUSIC ON HOLD OFF\n";
>
> Ignore the "-" - stupid outlook needs them.
There is one problem: I have not any MOH class, and cannot pre-create
it. File I will play does not exists un
2012/2/3 Danny Nicholas :
>
> Fork or shell task 1
> Fork or shell task 2
>
What exactly commands I should invoke in AGI instead of
and ?
STREAM FILE returns only after file ends, this is not what I want.
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2012/2/3 Steve Edwards :
> Do your processing in an AGI. Before you get into your 'longtime task'
> create another thread to play your file.
>
> When you finish your 'longtime task' join the background thread.
Yes, part of task is executed in AGI. But, I still do not understand
how I can do someth
Hi
I have a task. While serving incoming call I should do some longtime
task(consumes more than few tens of seconds). So I decided to turn on
background music in order to entertain caller. `core show
applications` showed me 3 potential candidates: Background, Playback
and StartMusicOnHold. Unfortu
Hi
I have an application. It connects to Asterisk via AMI, and when user
decides it begins asynchronous origination to some device. But very
often user decides to break origination and make another call. How can
I achieve that? As much as I see, Asterisk doesn't returns any ID of
dial process and
2011/12/26 sean darcy :
> So how do I get * to listen to two different ports?
sip.conf
section [general]
bindport=whatever-port-you-want
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New to Asterisk? J
2011/12/6 Danny Nicholas :
> You don't state your Asterisk version, but this sounds like a task for
> chan_skinny perhaps? Or it might just be as simple as hitting an RTP range.
Asterisk >=1.8
No, I want manually connect to asterisk via ssh console. I.e. like:
ssh user@ast-host /sbin/rasterisk -
Hello
I have machine running a couple of instances of asterisk. Each
instance create own control pipe (asterisk.ctl). How I can remotely
connect into asterisk which own pipe I know?
I know I can do it if path to pipe specified in asterisk.conf, but I
have not any asterisk.conf accessible, only co
After origination successfully complete and channel will be created
you probably should link ActionID and channel name.
Origination action will be next:
Action: Originate
Channel: Local/1@internal
Exten: 384087
Context: SIP-UA-00128
Priority: 1
CallerID: 601
ActionID: FFA02C6A03
Variable: Actio
I shall contact when(and if) decision will be made. But such decision
cannot be made basing only on this paragraph, because it does not
describes anything. There are no description of licensing procedure,
nor pricing, nor liability, rights or freedoms(at least in general
approximation) of sides. So
Greetings
I have found next paragraph in Licence file(source root)
"Digium, Inc. (formerly Linux Support Services) holds copyright
and/or sufficient licenses to all components of the Asterisk
package, and therefore can grant, at its sole discretion, the ability
for companies, individuals, or organ
You need simple dialplan of four steps:
same =>n,Set(conf_name=conf-${RAND(1,1000)})
same =>n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x)
same =>n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x)
same =>n,MeetMe(${conf_name},dFI1xAC)
same =>n,Noop(do post conference stuff)
2011/10/31 Thana
2011/10/25 Tarek Sawah :
> Hello,
> Is L6 a remote device? is there any firewall residing between the server and
> UA?
>
>
> Tarek Sawah
>
> Information Technology Adviser
>
> Integrated Digital Systems
>
> CCNP, MCSE, RHCE, TELECOM
>
> USA: +1 386 492 9993
>
L6 is account of DLINK DVG7022S VoIP
Hello
Always returns 401 Unauthorized, because of
[Oct 25 11:59:48] NOTICE[2501] chan_sip.c: Correct auth, but based on
stale nonce received from '"L6"
;tag=31b9dc9e-684902'
L6 is realtime device of type FRIEND (DLINK DVG7022S)
Reviewed SIP conversation - no results.
SIP debug
<--- SIP read fro
Hello
Have a setup of asterisk with realtime SIP devices.
Trying to organise monitoring of my SIP devices. Once device
registered, its state becomes NOT_INUSE (result of
DEVICE_STATE(SIP/device) function).
Simulating of device breakage - powerdown it.
Waiting for a while (minute or two), retrievin
2011/10/12 Kevin P. Fleming :
> then Asterisk *could* stop sending audio towards the device connected to that
> channel.
> then Asterisk *could* send it a message telling it to not bother sending any
> audio.
I think in any case Asterisk must not halt any audio data stream. It
is task of applica
Unfortunately I don't know behaviour of Progress() function, so cannot
make any conclusions. As far as I traced it back to tech-
implementation, this call does not changes any state of channel. But I
analysed only sip and dahdi drivers. Neither it plays any indication
tones.
2011/10/5 Sammy Govind
Yes, something like that, but
hold"-state should not answer channel. answer command will be given
explicitly. or call can be transfered to Dial command, etc.
2011/10/5 Sammy Govind :
> Can you please explain what you are trying to do? What I've perceived from
> this thread is that you want to put
wrote additional module to make * work as I
require).
2011/10/5 Nasir Iqbal :
> What about waiting in "queues"?
> Nasir Iqbal
>
> ICT Innovations
> http://www.ictinnovations.com/
>
>
>
> On Wed, Oct 5, 2011 at 1:35 PM, Yaroslav Panych wrote:
>>
>>
Hello, everyone
Here part of my dialplan context
[globals]
CMD_NOOP=0
CMD_DOSTUFF1=1
CMD_DOSTUFF2=2
CMD_DOSTUFF3=2
[blah-context]
same => n,Set(COMMAND=${CMD_NOOP})
same => n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
same =>
n(COMMAND_SWITCH),GoToIf($["${COMMAND}"="${CMD_DOSTUFF1}"]?LBL_DO
Hello, everyone
Here part of my dialplan context
[globals]
CMD_NOOP=0
CMD_DOSTUFF1=1
CMD_DOSTUFF2=2
CMD_DOSTUFF3=2
[blah-context]
same => n,Set(COMMAND=${CMD_NOOP})
same => n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
same =>
n(COMMAND_SWITCH),GoToIf($["${COMMAND}"="${CMD_DOSTUFF1}"]?LBL_DO_S
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