On 7/10/2023 8:55 PM, Federico wrote:
I need to use app_macro, but it seems to be absent from asterisk 16.30.1
Is there a workaround?
It's disabled (not built) by default. You'll need to enable it using
menuselect[1], and load it in modules.conf
Note that app_macro has been removed now and wo
I need to use app_macro, but it seems to be absent from asterisk 16.30.1
Is there a workaround?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://c
Thank you Joshua
-Original Message-
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Friday, May 24, 2019 9:53 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Is there a way to make asterisk send a INVITE
in-dialog to re-establish the audio
On Fri, May 24
On Fri, May 24, 2019, at 9:47 AM, Dan Cropp wrote:
>
> We are working with an Avaya switch.
>
>
> We send them a REFER. If the transfer is successful, everything is
> great. If it fails (busy), they send an INVITE in-dialog with a media
> attribute of inactive. After that, they send a 486 bus
We are working with an Avaya switch.
We send them a REFER. If the transfer is successful, everything is great. If
it fails (busy), they send an INVITE in-dialog with a media attribute of
inactive. After that, they send a 486 busy.
The problem is Avaya basically put the call on hold so audio i
Yes: I never thought of using sudo to also forbid access some apps.
Using it for that is very smart !
Thank you for sharing it here.
I'll experiment with this and report here my findings.
Thanks again
2018-08-14 19:50 GMT+02:00 John Kiniston :
> I use sudo to limit this.
>
> Cmnd_Alias CAPTAGENT
I use sudo to limit this.
Cmnd_Alias CAPTAGENT = /sbin/service captagent stop, /sbin/service
captagent start, /sbin/service captagent restart
Cmnd_Alias ASTERISK = /sbin/service asterisk stop, /sbin/service asterisk
start, /sbin/service asterisk restart, /usr/sbin/rasterisk,
/usr/sbin/asterisk, /u
Hello,
Is there a way to let someone access to Asterisk CLI and type whatever
command (s)he likes but the shell command (the ones started by !) ?
Ideally, it could be an argument to rasterisk:
rasterisk --no-shell
When done, a session could be like this:
> pjsip show endpoints
...
> core reloa
Hello.
I have plain text password for endpoint with outbound registration
(someone else's server).
My aim is to write it in pjsip.conf.
md5 means that I know realm. I do not always know it.
Is where any way?
Dmitriy Serov.
--
_
Thanks.
Is there command is used for that?
I have checked the help show and there is no command like sip register
or sip unregister in the list.
Is it available on version 1.4?
On 11 March 2010 13:08, Kevin P. Fleming wrote:
> Frank Church wrote:
>> Is there a way for a client to tell a server
Hi
> There's actually not an UNREGISTER method in SIP.
> As Kevin stated, you send a REGISTER with a zero expiry to cancel a
> current registration.
Yes, of course you are right there, sorry for the confusion. I was
thinking about the resulting Asterisk CLI message:
Unregistered SIP 'peername
11 mar 2010 kl. 15.17 skrev Philipp von Klitzing:
>> Is there a way for a client to tell a server where it is registered to
>> remove the registration?
>
> Yes, it needs to send an UNREGISTER sip message.
>
There's actually not an UNREGISTER method in SIP.
As Kevin stated, you send a REGISTER w
> Is there a way for a client to tell a server where it is registered to
> remove the registration?
Yes, it needs to send an UNREGISTER sip message.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Frank Church wrote:
> Is there a way for a client to tell a server where it is registered to
> remove the registration?
Assuming you are talking about a SIP peer (since you didn't specify),
yes, the SIP peer can cancel the registration by sending an update to
the registration and setting the expir
Is there a way for a client to tell a server where it is registered to
remove the registration?
/voipfc
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
long time ago I added the "SIP_CODEC" variable that you can set from
within the dialplan, eg:
exten => s,1,Set(SIP_CODEC=alaw)
exten => s,n,Answer
exten => s,n,whatever
now if the remote side actually supports the chosen codec Asterisk
will try to use that one ...
there's no error reporting as fa
Hello,
I'd like to implement some public sip uri's that poeple can call into
and get an echo test. Is there a way to force a codec so that users
can test various codecs?
Something like:
echo-t...@example.com (negotiates whatever codec, is there a way to
figure out what codec was negotiate
Found it, I use the g flag in Dial command, that helps :)
Rennes
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rennes Neps
Sent: 1. oktoober 2009. a. 16:05
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Is there
Hei!
Here's my problem. I have an Asterisk with SS7 and SIP trunks. Asterisk
version is 1.6. I'm setting up a custom CDR fields and I was wondering
is there a way to know who initiated a hangup? Asterisk must be aware of
that info somehow, cause in queue_log, that info is present
(completecalle
Eric Chamberlain schrieb:
> Is there a way to override the fromdomain specified in the sip.conf
> and instead set the value from the dialplan?
>
> If we use:
>
> Set(CALLERID(num)[EMAIL PROTECTED]
>
> The SIP From header turns into:
>
> [EMAIL PROTECTED]@10.10.10.10
Maybe you could abuse S
Is there a way to override the fromdomain specified in the sip.conf
and instead set the value from the dialplan?
If we use:
Set(CALLERID(num)[EMAIL PROTECTED]
The SIP From header turns into:
[EMAIL PROTECTED]@10.10.10.10
We want [EMAIL PROTECTED], and we can't have an entry in sip.conf for
Brent Davidson wrote:
Babcock, Michael Alex wrote:
hey;
i'm at best western and am curious is there a way i could find out if
our best western, with out asking, is using asterisk?
oh and petsmart i think is using asterisk they have alason voice for
there main voicem enu.
mike
thanks fo
Steve Totaro wrote:
>
> My only wish is that Linux had a facility like XP to bridge NICs
> without running all sorts of commands for brctl. Just a GUI like XP.
> Last time I setup a bridge in Linux, I had to change many kernel
> options and rebuild the entire kernel to get bridging working
>
At 9:37 AM -0700 2008/10/13, Eric Chamberlain wrote:
>On Oct 11, 2008, at 8:28 PM, Rob Hillis wrote:
>
>> Eric Chamberlain wrote:
Is there a particular reason you /can't/ register? It would seem
that
registration would provide the functionality you require, even if
you're
On Mon, Oct 13, 2008 at 12:37 PM, Eric Chamberlain <[EMAIL PROTECTED]> wrote:
>
>
> We're developing the client and don't have control over the server,
> which may or may not be Asterisk. Adding extra extensions isn't
> possible.
>
> Can OPTION packets be used to verify authentication?
>
> --
> Er
On Oct 11, 2008, at 8:28 PM, Rob Hillis wrote:
> Eric Chamberlain wrote:
>>> Is there a particular reason you /can't/ register? It would seem
>>> that
>>> registration would provide the functionality you require, even if
>>> you're
>>> only making outbound calls.
>>>
>> In the case of a server
Eric Chamberlain wrote:
>> Is there a particular reason you /can't/ register? It would seem that
>> registration would provide the functionality you require, even if
>> you're
>> only making outbound calls.
>>
> In the case of a server like Asterisk, wouldn't sending a register
> disrupt
On Oct 11, 2008, at 1:41 PM, Rob Hillis wrote:
> Eric Chamberlain wrote:
>> I should have clarified, we're only making outbound calls, not
>> inbound, so there is no registration.
>>
>
> Is there a particular reason you /can't/ register? It would seem that
> registration would provide the functi
Eric Chamberlain wrote:
> I should have clarified, we're only making outbound calls, not
> inbound, so there is no registration.
>
Is there a particular reason you /can't/ register? It would seem that
registration would provide the functionality you require, even if you're
only making outb
then this is a error from me, thanks
- Original Message -
From: "Eric Chamberlain" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, October 11, 2008 6:03 PM
Subject: Re: [asterisk-users]
> - Original Message -
> From: "Eric Chamberlain" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Saturday, October 11, 2008 5:20 PM
> Subject: [asterisk-users] Is there a way to test SIP credentials
&g
, October 11, 2008 5:20 PM
Subject: [asterisk-users] Is there a way to test SIP credentials
withoutmaking a call?
> Is there a SIP packet that a SIP client can send to Asterisk to
> confirm that the credentials entered by the user are correct, without
> placing a call?
>
> We
Is there a SIP packet that a SIP client can send to Asterisk to
confirm that the credentials entered by the user are correct, without
placing a call?
We'd like to test the credentials when the user enters them, rather
than wait until they try to make their first call.
--
Eric Chamberlain
Steve Totaro wrote:
My only wish is that Linux had a facility like XP to bridge NICs without
running all sorts of commands for brctl. Just a GUI like XP. Last time I
setup a bridge in Linux, I had to change many kernel options and rebuild the
entire kernel to get bridging working properly. Wi
On Fri, Oct 10, 2008 at 07:33:45PM -0700, Eric Fort wrote:
> nmap for scanning and identification. cross platform and even a nice gui
> for windows.
What nmap does is called "fingerprinting". it mostly uses the fact that
when faced with normal behaviours, most stacks behave the same. But when
fac
I will look into that when I get my Acer Aspire One running FC8, it came
with windows XP and I got the 1gig ram, 120gig HD.
I am following threads on howto but nobody has a definitive guide yet, that
allows the embedded webcam and the NIC to work properly.
Maybe (probably) my USB Alpha AWUS036H w
nmap for scanning and identification. cross platform and even a nice gui
for windows.
Eric
On Fri, Oct 10, 2008 at 3:20 PM, Steve Totaro <
[EMAIL PROTECTED]> wrote:
>
>
> On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson <
> [EMAIL PROTECTED]> wrote:
>
>> Babcock, Michael Alex wrote:
>> > hey;
>>
steve;
thanks a lot
mike
On Oct 10, 2008, at 2:20 PM, Steve Totaro wrote:
On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson <[EMAIL PROTECTED]
> wrote:
Babcock, Michael Alex wrote:
> hey;
> i'm at best western and am curious is there a way i could find out
if
> our best western, with out as
no i'm a guest at the bestwestern
On Oct 10, 2008, at 1:55 PM, Brent Davidson wrote:
> Babcock, Michael Alex wrote:
>> hey;
>> i'm at best western and am curious is there a way i could find out if
>> our best western, with out asking, is using asterisk?
>> oh and petsmart i think is using asterisk
On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson <[EMAIL PROTECTED]
> wrote:
> Babcock, Michael Alex wrote:
> > hey;
> > i'm at best western and am curious is there a way i could find out if
> > our best western, with out asking, is using asterisk?
> > oh and petsmart i think is using asterisk they
Babcock, Michael Alex wrote:
> hey;
> i'm at best western and am curious is there a way i could find out if
> our best western, with out asking, is using asterisk?
> oh and petsmart i think is using asterisk they have alason voice for
> there main voicem enu.
> mike
>
>
> thanks for reading
> S
hey;
i'm at best western and am curious is there a way i could find out if
our best western, with out asking, is using asterisk?
oh and petsmart i think is using asterisk they have alason voice for
there main voicem enu.
mike
thanks for reading
Systems administrator and owner of http://gwhost
Here's the use case: call comes in, extension match is made on caller
ID and dialed number, dial plan dials a number and connects the two
call legs.
Is there a way to get the Call-ID from the SIP header of the outbound
call leg and store it in the CDR?
--
Eric Chamberlain
__
"Philippe Sultan" <[EMAIL PROTECTED]> writes:
> Well, if someone steals the md5secret (HA1) for a given username and
> realm, he can use it to authenticate to the SIP proxy or B2BUA that
> serves the target user.
This is unavoidable with password-based systems.
Either you transfer the password u
On 20 Aug 2008, at 18:00, Eric Chamberlain wrote:
> We are exploring using Asterisk for a project and we are looking for a
> way to encrypt/decrypt the peer passwords stored in the realtime
> database (postrges).
>
> Ideally, we want to use a public key to encrypt the passwords before
> they go i
On Wed, Aug 20, 2008 at 02:10:02PM -0700, Eric Chamberlain wrote:
>
> On Aug 20, 2008, at 10:19 AM, Tzafrir Cohen wrote:
>
> > On Wed, Aug 20, 2008 at 10:00:55AM -0700, Eric Chamberlain wrote:
> >> We are exploring using Asterisk for a project and we are looking
> >> for a
> >> way to encrypt/d
Well, if someone steals the md5secret (HA1) for a given username and
realm, he can use it to authenticate to the SIP proxy or B2BUA that
serves the target user.
On both sides (SIP client and proxy or B2BUA), the values to be
compared are the computed results of MD5(HA1:nonce:HA2), where :
HA1 = MD
Hey Eric,
That I really have no experience with. Never really played with security
modules. Although someone more experienced should be able to chime in.
Eric Chamberlain wrote:
> On Aug 20, 2008, at 12:34 PM, Igor Hernandez wrote:
>
>> Hey SIP,
>>
>> I understand what you're saying but keeping
On Aug 20, 2008, at 12:34 PM, Igor Hernandez wrote:
> Hey SIP,
>
> I understand what you're saying but keeping the key in memory
> permanently doesn't protect you for very long, it just makes the
> attacker waste a bit more time scanning the memory to get at the key.
>
> In other words, if the ke
On Aug 20, 2008, at 10:19 AM, Tzafrir Cohen wrote:
> On Wed, Aug 20, 2008 at 10:00:55AM -0700, Eric Chamberlain wrote:
>> We are exploring using Asterisk for a project and we are looking
>> for a
>> way to encrypt/decrypt the peer passwords stored in the realtime
>> database (postrges).
>>
>> I
I understand the advantage of md5 hashing, its been the standard for
years for day to day user auths. What we were discussing was the merits
of the proposed public key scheme for this application, where the
private key would always need to be available therefore not giving any
real security.
Regar
Igor Hernandez wrote:
> I was thinking the same thing I believe Tzafrir just alluded to. If the
> passwords are encrypted in the DB with a public key then...asterisk
> needs to have the private key stored somewhere to be able to decrypt the
> values to authenticate the user. In this way there is no
I've never used it, but check out the md5 one-way encryption of passwords:
http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+md5secret
http://books.google.com/books?id=vAT8Mfvp8GsC&pg=PA225&lpg=PA225&dq=asterisk+md5+secret&source=web&ots=1mUADiyRkP&sig=FJSBgcWMY3K0zoilVvgNvibJE4A&hl=en&sa
Hey SIP,
I understand what you're saying but keeping the key in memory
permanently doesn't protect you for very long, it just makes the
attacker waste a bit more time scanning the memory to get at the key.
In other words, if the key is available to asterisk it will be available
to anyone else in
Igor Hernandez wrote:
> I was thinking the same thing I believe Tzafrir just alluded to. If the
> passwords are encrypted in the DB with a public key then...asterisk
> needs to have the private key stored somewhere to be able to decrypt the
> values to authenticate the user. In this way there is no
Igor Hernandez wrote:
> I was thinking the same thing I believe Tzafrir just alluded to. If the
> passwords are encrypted in the DB with a public key then...asterisk
> needs to have the private key stored somewhere to be able to decrypt the
> values to authenticate the user. In this way there is no
On Wed, Aug 20, 2008 at 02:20:50PM -0400, SIP wrote:
> Tzafrir Cohen wrote:
> > On Wed, Aug 20, 2008 at 10:00:55AM -0700, Eric Chamberlain wrote:
> >
> >> We are exploring using Asterisk for a project and we are looking for a
> >> way to encrypt/decrypt the peer passwords stored in the realtim
I was thinking the same thing I believe Tzafrir just alluded to. If the
passwords are encrypted in the DB with a public key then...asterisk
needs to have the private key stored somewhere to be able to decrypt the
values to authenticate the user. In this way there is nothing preventing
whoever intru
Tzafrir Cohen wrote:
> On Wed, Aug 20, 2008 at 10:00:55AM -0700, Eric Chamberlain wrote:
>
>> We are exploring using Asterisk for a project and we are looking for a
>> way to encrypt/decrypt the peer passwords stored in the realtime
>> database (postrges).
>>
>> Ideally, we want to use a pub
On Wed, Aug 20, 2008 at 10:00:55AM -0700, Eric Chamberlain wrote:
> We are exploring using Asterisk for a project and we are looking for a
> way to encrypt/decrypt the peer passwords stored in the realtime
> database (postrges).
>
> Ideally, we want to use a public key to encrypt the passwords
We are exploring using Asterisk for a project and we are looking for a
way to encrypt/decrypt the peer passwords stored in the realtime
database (postrges).
Ideally, we want to use a public key to encrypt the passwords before
they go into the database and have Asterisk use a private key to
On Tue, May 13, 2008 at 4:22 AM, Grey Man <[EMAIL PROTECTED]> wrote:
>
> On Mon, May 12, 2008 at 9:44 PM, Sanjay Rajdev
> <[EMAIL PROTECTED]> wrote:
> > Hello All,
> >
> > Is there a way to have Manager Bridge Channel to the specified extension
> > without the channel being connected.
> >
>
On Mon, May 12, 2008 at 9:44 PM, Sanjay Rajdev
<[EMAIL PROTECTED]> wrote:
> Hello All,
>
> Is there a way to have Manager Bridge Channel to the specified extension
> without the channel being connected.
>
> In the current scenario the channel only bridges once the call get
> connected, it does not
Hello All,
Is there a way to have Manager Bridge Channel to the specified extension
without the channel being connected.
In the current scenario the channel only bridges once the call get connected,
it does not bridge when any service provider (telco) message is played. I want
to record all
Andreas Bayer wrote:
> is there a way to turn of SIP METHOD OPTIONS in asterisk?
>
> I have a sip pbx which ignore Sip Option Messages from a unknown user.
> Asterisk send Option Messages to peers with From: [EMAIL PROTECTED] The sip
> server expects From: [EMAIL PROTECTED] server domain].
>
Hi,
is there a way to turn of SIP METHOD OPTIONS in asterisk?
I have a sip pbx which ignore Sip Option Messages from a unknown user.
Asterisk send Option Messages to peers with From: [EMAIL PROTECTED] The sip
server expects From: [EMAIL PROTECTED] server domain].
So i have to turn off Options
Hi,
Does anyone knows if I can collect DTMF digits (inband) during a bridged
call (E1 to E1),?
I see the DTMF tones on the debug file but does not activate the dialplan.
My problem is, I need to signal from the second E1 to bridge the call
to another E1 (a third one), if I use the tra
Hi all.
Today I have tried to connect to the AMP with http://myserverip but I can
not connect to the AMP (it sends me out of my network).
What would be happening?.
The last thing I did is to try to change the digital receptionist manually.
Is there a way to re-install the amp?
Thanks
___
Split the contexts up even more. Keep in mind the SIP users you setup can all
start in a different context and you can have your incoming zap calls start in a
different context.
Ie, make a context that includes ability to dial internally and outside. Then
make another context that just inclu
Alvaro:
I dont think such a thing exists for a simple reason. I you think
things "right", you will be able to end with a good combination of
contexts.
In your case, having:
[internal]
include => invalid
exten => _XX,1,Dial(SIP/12345${EXTEN}) ; Short alias for internal lines
exten => _12345XX,1,
I've defined my dialplan as showed below. My internal lines are numbered
as 12345XX, and internal users can call another by the entire 7-digits
extension, or by just last 2 digits.
[invalid]
exten => _X.,1,Playback(pbx-invalid)
exten => _X.,2,Hangup()
[internal]
include => invalid
exten => _XX
On 5/1/06, Obelix <[EMAIL PROTECTED]> wrote:
Is there a way to monitor a call for DTMF tones an trigger some actions based on
those DTMF tones?
I am interested in any arbitrary DTMF tones, not those related to the usual PBX
functions like call transfer, music on hold, call diversion etc
take
Is there a way to monitor a call for DTMF tones an trigger some actions based on
those DTMF tones?
I am interested in any arbitrary DTMF tones, not those related to the usual PBX
functions like call transfer, music on hold, call diversion etc
/Obelix
_
Is there a way to monitor the DTMF tones on a channel?
I have a prepaid application working in asterisk. When the user dials a call and
wants to cancel the call before it is answered, there is now way to do it
without hanging up and redialling the access number.
Is there way to monitor a sequen
Hi,
Is there a way to see what agents belong to which group. In the CLI I see
references to my Agent group ie Agent/@1 but I have no way of verifying that
all users are in the right group.
Thanks
___
--Bandwidth and Colocation sponsored by Easynews.com
Using contexts, and making sure which device is coming in to where.
On 8/19/05, Angus Comber <[EMAIL PROTECTED]> wrote:
> Hello
>
> If callerid is not available on an external line, how can you tell if call
> is incoming or outgoing?
>
> Angus
>
>
>
Hello
If callerid is not available on an external line, how can you tell if call
is incoming or outgoing?
Angus
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
AIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Is there a way to get inserted into an LEC's
CLIDB?
NeuStar also offers CNAM db services, but VeriSign pays you for your cnam
listings as they receive reciprocal compensatio
NeuStar also offers CNAM db services, but VeriSign pays you for your cnam
listings as they receive reciprocal compensation for their databases,
probably charging rbocs, clecs etc per query.. I'm not sure about NeuStar
or how they handle this, but I'm almost positive that they provide cnam
updat
Tom Samplonius wrote:
I had be using a group of two PRIs for more than a year on a Nortel
PBX. After I started testing with Asterisk through a AS5300 gateway,
I quickly noticed that I could present any calling number.
Yes, we all know we can do that (and do it every day). The poster's
question
n
> > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's
> > CLIDB?
> >
> >
> > Does anyone know if there's a service out there to -- for a fee --
> > inject our DID into the LEC's CLI database so a called party gets our
> &g
Verisign, CNAM
http://www.verisign.com/products-services/communications-services/intelligent-database-services/cnam-calling-name-database/page_001662.html
Look there.
-m
-
"Yeah, we rocked the vote all right. Those little
bastards betrayed us agai
We offer that service for our termination customers,
however we can only provide it for (206) area code
numbers. So what we find is people who don't care as
much about the number and more about their callerid
lookup such as businesses and call centers opt to
utilize it. We can even change the name
Robert Goodyear wrote:
Does anyone know if there's a service out there to -- for a fee --
inject our DID into the LEC's CLI database so a called party gets
our associated name?
No, only if the LEC servicing the number offers it to you. It is the
responsibility of the operator running the switc
Robert Goodyear wrote:
Does anyone know if there's a service out there to -- for a fee --
inject our DID into the LEC's CLI database so a called party gets our
associated name?
No, only if the LEC servicing the number offers it to you. It is the
responsibility of the operator running the switch
Robert Goodyear wrote:
Does anyone know if there's a service out there to -- for a fee --
inject our DID into the LEC's CLI database so a called party gets our
associated name?
No, only if the LEC servicing the number offers it to you. It is the
responsibility of the operator running the switch
> -Original Message-
> From: Robert Goodyear [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, March 22, 2005 1:21 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's
> CLIDB?
>
>
Does anyone know if there's a service out there to -- for a fee --
inject our DID into the LEC's CLI database so a called party gets our
associated name?
/rg
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailma
how about using chanisavail via manager api
On Thu, 2005-03-03 at 16:21, Paco Perez wrote:
Hello:
I would like to know if there's a way to request free chanels from remote
asterisk servers ?
My idea is to make an agi returning a dial to inter-asterisk connected servers
when there's not en
Hello:
I would like to know if there's a way to request free chanels from remote
asterisk servers ?
My idea is to make an agi returning a dial to inter-asterisk connected servers
when there's not enought chanels on local server, maybe like a ping to all of
them or maybe requesting to a central
Hi!
Is there a way to avoid being "at the middle" of communications between
two SIP endpoints? So that we can avoid loosing bandwidth with it?
Is there a way to "forward" the authentication to a IAX provider and
"transfer" the call to it, avoiding using my own bandwidth?
I've tested it with SE
On Tue, 23 Nov 2004 17:14:30 -0700, Chris Modesitt <[EMAIL PROTECTED]> wrote:
>
>
>
> Is there a way to check if an extension exists?
>
>
>
> This is the problem I ran into, I have exceeded the number of extensions you
> can attempt to match in one pass (1500+ Extensions).
The solution is
Is there a way to check if an extension exists?
This is the problem I ran into, I have exceeded the number
of extensions you can attempt to match in one pass (1500+ Extensions). I am
hoping that someone has discovered a clever way of checking if an extension
exists in a particular conte
I think that has to be done on a device by device basis.
In come cases the configuration change is done on the client
(software/device).
If it's a card that goes in the linux box that is running asterisk you
can set that in the config file for the device.
zapata.conf for Zaptel cards.
phondev.co
On Mon, 2004-11-01 at 19:56 -0600, John Lange wrote:
> I would like to completely disable call waiting.
>
> Does Asterisk have an option for that?
While there are many very smart individuals on this list, I don't think
there is many mind readers nor hackers who will log into your system to
explai
I would like to completely disable call waiting.
Does Asterisk have an option for that?
Thanks,
--
John Lange
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How can i acheive this feature?
Regards...
Girish
From: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Is there a way to transfer a call from CLI
Date: Mon, 26 Jan 2004 19:36:44 -0500 (EST)
Does anyone know of a way to tr
I'd be interested in the patch as well
On Thu, 2004-01-22 at 13:51, Bill Hamel wrote:
> Hi Chris,
>
> This sounds what I am looking for, many thanks !
>
> Also, I do not see an attachment, the patch that is :)
>
> I dont know if the list strips attachments, perhaps send it to my email address
>
Nope, I'm an idiot. Here's the patch :P
On Thu, 22 Jan 2004, Bill Hamel waxed:
> Hi Chris,
>
> This sounds what I am looking for, many thanks !
>
> Also, I do not see an attachment, the patch that is :)
>
> I dont know if the list strips attachments, perhaps send it to my email address
> [EMA
Hi Chris,
This sounds what I am looking for, many thanks !
Also, I do not see an attachment, the patch that is :)
I dont know if the list strips attachments, perhaps send it to my email address
[EMAIL PROTECTED]
Thanks again,
-bh
Quoting "C. Maj" <[EMAIL PROTECTED]>:
> I attached a patch I've
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