Thank you, Joshua!
--
sergio.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
On Sat, Jun 12, 2021 at 11:54 PM sergio wrote:
> I have pjsip endpoint with callerid= context=localpeers which
> looks follow:
>
> [localpeers]
> exten => _.@_./sergio,1,Dial(Local/${EXTEN}@somecontext)
> exten =>_.@_.,1,NoOp()
>
> And this works fine:
>
>== Setting global variable
I have pjsip endpoint with callerid= context=localpeers which
looks follow:
[localpeers]
exten => _.@_./sergio,1,Dial(Local/${EXTEN}@somecontext)
exten =>_.@_.,1,NoOp()
And this works fine:
== Setting global variable 'SIPDOMAIN' to 'DOMAIN'
-- Executing [EXTEN@localpeers:1]
On Thursday 07 March 2013, Luis H. Forchesatto wrote:
Greetings.
I got an extension on my Elastix who cannot pick calls on the other
extensions, but It can transfer his calls to the other extensions. When
this extension tries to pickup a call pressing *8 it simply does not pick
it up.
I think I found the problem. Better looking the sip_additional.conf file I
noticed that a few extensions didnt had a callgroup and pickgroup
configured, even with the interface appointing otherwise.
I manually configured this options and reloader asterisk and now I'm gonna
test the extensions and
Yes, it worked :D
Thankyou guys for the help.
2013/3/8 Luis H. Forchesatto luisforchesa...@gmail.com
I think I found the problem. Better looking the sip_additional.conf file I
noticed that a few extensions didnt had a callgroup and pickgroup
configured, even with the interface appointing
On Friday 08 March 2013, Luis H. Forchesatto wrote:
Yes, it worked :D
Thankyou guys for the help.
Glad it worked for you.
Just be very careful if you change anything via the GUI in future, because it
might undo any changes you made manually -- especially if you didn't get the
format of
Greetings.
I got an extension on my Elastix who cannot pick calls on the other
extensions, but It can transfer his calls to the other extensions. When
this extension tries to pickup a call pressing *8 it simply does not pick
it up. Transfering calls works just fine so dtmf may be not the
If I was in your shoes, I'll check in the elastix mailing list... Asterisk
itself can't be blamed.
Leandro
I am typing from my mobile phone...
Il giorno 07/mar/2013 19:06, Luis H. Forchesatto
luisforchesa...@gmail.com ha scritto:
Greetings.
I got an extension on my Elastix who cannot pick
On 8/03/2013, at 7:46 AM, Leandro Dardini ldard...@gmail.com wrote:
If I was in your shoes, I'll check in the elastix mailing list... Asterisk
itself can't be blamed.
Leandro
I am typing from my mobile phone...
Il giorno 07/mar/2013 19:06, Luis H. Forchesatto
do you have only ONE phone, that can´t pickup, or is this a general problem?
is pickup configured (feature.conf) AND enabled ?
regards,
yves
Am 07.03.2013 19:05, schrieb Luis H. Forchesatto:
Greetings.
I got an extension on my Elastix who cannot pick calls on the other
extensions, but It
Its only ONE phone who doesnt pickup calls.
2013/3/7 Yves A. yves...@gmx.de
do you have only ONE phone, that can´t pickup, or is this a general
problem?
is pickup configured (feature.conf) AND enabled ?
regards,
yves
Am 07.03.2013 19:05, schrieb Luis H. Forchesatto:
Greetings.
I
is it the same type and make of phone than one of the working ones?
- compare (dtmf) settings, firmware release etc.
- check call-group and pickup group... is the non working extension
configured there?
regards,
yves
Am 07.03.2013 20:28, schrieb Luis H. Forchesatto:
Its only ONE phone who
Yes, both are configured in the same ata (linksys pap2) and the
configuration options are the same. Call group and pick group are the same
for both too.
2013/3/7 Yves A. yves...@gmx.de
is it the same type and make of phone than one of the working ones?
- compare (dtmf) settings, firmware
mmh... should work... (i think you checked double and applied any
changes, right..?
sometimes deleting the extension and configuring a new one can fulfil
wonders...)
I have no further tip... maybe elastix support or forum can help... if
you are familiar with
cli output and sip debugging...
Hello,
I want to manage hints in a different way, putting all the hints in the
same context and trying to recognize the subscribing peer, but I can't find
any variable set about the calling peer. Peers need to be authenticated to
be able to subscribe to the hint, but I am not able to access any of
We have a extension pad on our Yealink phone for the receptionist. With our
old non-voip PBX system, the receptionist could pickup a specific extension
by pressing the corresponding key. Is this possible with Asterisk too?
I have configured Asterisk to pickup a specific extension with *59exten.
Hi, I'm using AMI to get the extension status but always get -1 i.e. extension
not found. #!/usr/bin/php -q
?phpinclude_once (phpagi-2.14/phpagi.php);
include_once (/phpagi-2.14/phpagi-asmanager.php);
$agi = new AGI();
$as = new AGI_AsteriskManager();
$exten =
Why don't you use AMI? There's are phpami project if you google.
Sent from my iPhone
On May 25, 2012, at 1:51 AM, Kamlesh Kumar kamlesh_...@hotmail.com wrote:
Hi,
I'm using AMI to get the extension status but always get -1 i.e. extension
not found.
#!/usr/bin/php -q
?php
sir,
is there any idea for this whenever 667and668 extension will dial isd call
before connect agent will dial password like ..
Best Regards,
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75
Thanks for reply.
but I need same context. because i am using dialer when I change the context
meetme is not configuring at login with dialer.
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75
sir,
is there any idea for this whenever 667and668 extension will dial isd call
before connect agent will dial password like ..
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali
Hi all,
I have n no. of extensions in my dialer. from 456 to 556 extensions. I was
created 2 other extensions 667 and 668 I need to allow only STD calls to
go from this extensions.
These all extensions are same context . I need to define the STD dialplan
for only this 2 extensions. how I can ?
On Wed, Oct 27, 2010 at 3:43 PM, Jose P. Espinal j...@slackware-es.comwrote:
Hello List,
A few days ago I installed ViciDial on a server, and while looking to
the default 'extensions.conf' file, I saw this line:
exten = _010*010*010*015*.,1,Dial(${TRUNKTESTast}/${EXTEN:16},55,oT)
Can
Thanks Leif,
Forgot I could do a db lookup for the ddi.
Dan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Hi,
When a VOIP user dials an external number, the calls are routed through our SIP
provider.
Is there a simple way to check whether the DDI exists locally before dialling
out to the sip provider?
Something like GotoIfExists(5551...@incoming_calls)
Currently, I'm paying for all calls,
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Monday, October 25, 2010 3:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extension Exists
Hi,
When a VOIP
On 10-10-25 04:21 PM, Dan Journo wrote:
Hi,
When a VOIP user dials an external number, the calls are routed through
our SIP provider.
Is there a simple way to check whether the DDI exists locally before
dialling out to the sip provider?
Something like GotoIfExists(5551...@incoming_calls)
Hi,
I use asterisk with sip3000 device with sip-aho connected to PSTN and
sip-ahi connected to a phone.
When call arrives from PSTN, the *phone continues ringing even after caller
hanged up*.
The dialplan contains the following lines:
[from-pstn]
...
exten = 99,n,Dial(SIP/sip-ahi,30,g)
exten =
Have you tried removing option 'g' from your Dial command?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-20 7:45 AM, Arie Skliarouk sklia...@gmail.com wrote:
Hi,
I use asterisk with sip3000 device with sip-aho connected to PSTN and
sip-ahi connected to a phone.
When call arrives from
Hi,
On Mon, Sep 20, 2010 at 16:39, Zeeshan Zakaria zisha...@gmail.com wrote:
Have you tried removing option 'g' from your Dial command?
Of course, with the same result.
--
Arie
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-20 7:45 AM, Arie Skliarouk sklia...@gmail.com wrote:
Do you mean spa3000 or sip3000? I remember having same problem with spa3000
and the problem was somewhere in the settings of spa3000 that wouldn't stop
ringing the phone. I don't remember the details at this moment as it was
long time ago, but this much I can tell that it is a config issue with
On Mon, Sep 20, 2010 at 7:31 AM, Arie Skliarouk sklia...@gmail.com wrote:
When call arrives from PSTN, the phone continues ringing even after caller
hanged up.
I suspect a bug [1] but without a SIP debug, I cannot be sure.
[1] https://reviewboard.asterisk.org/r/870/
--
Paul Belanger | dCAP
You are right, the device is called Sipura SPA-3000. The settings are
factory-set, I haven't changed anything beside of SIP registration with the
asterisk.
How can I enable SIP debug?
--
Arie
On Mon, Sep 20, 2010 at 18:01, Paul Belanger
paul.belan...@polybeacon.comwrote:
On Mon, Sep 20,
Hi,
I've noticed that if a phone goes UNREACHABLE while it is Ringing,
when the phone comes back, Asterisk will not clear the channel that
was created, so it still thinks it is in the Ringing state.
The only way to clear this is to do a soft hangup on the SIP channel
or to restart Asterisk.
Ben Schorr wrote:
Is there some reason why I keep getting this same message from “cool
dude” over and over and over? And under different subject lines?
I assumed that he was sending it to every post, trying to get a
response. Sorta spammy in my opinion.
Doug
--
Ben Franklin quote:
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can
make call outside and exten 2006 to 2010 can not make call outside. heres my
dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
...@rolandschorr.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cool dude
Sent: Friday, February 12, 2010 21:24
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] extension not found
hi friend need ur help in dial plan
Hello,
I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know
how can i monitor the extension status?
when i wrote sip show peers on asterisk
Extension Domain port Status
111/111(Unspecified)D 0
ahmed magdy wrote:
Hello,
I am new in Asterisk Community, i am working on Asterisk 1.6, i need
to know how can i monitor the extension status?
when i wrote sip show peers on asterisk
Extension Domain port Status
111/111
11 jan 2010 kl. 12.25 skrev ahmed magdy:
Hello,
I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know
how can i monitor the extension status?
when i wrote sip show peers on asterisk
Extension Domain port Status
111/111
There are a couple of ways you could see that,
One would be by having a service .NET connected to the manager interface
and watching for activity on the phone, this way you could tell if the
phone is busy or not.
[If phone has more than one line then set call-limit=1]
Is this for routing
Hi,
We have SPA921 handsets which apparently support Paging, however i can't
find any information on configuring Asterisk to make a page call.
Does anyone have any information on Paging?
Many thanks
Dan Journo
___
-- Bandwidth and
Check on voip-info.org
-Original Message-From: d...@keshercommunications.comSent: Wed, 14 Oct 2009 22:24:41 +0100To: asterisk-users@lists.digium.comSubject: [asterisk-users] Extension Paging
Hi,
We have SPA921 handsets which apparently support Paging, however i can’t find any
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Extension Paging
Check on voip-info.org
-Original Message-
From: d...@keshercommunications.com
Sent: Wed, 14 Oct 2009 22:24:41 +0100
To: asterisk-users@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Godbout
Sent: 14 October 2009 22:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Extension Paging
Check on voip-info.org
-Original Message-
From: d
So what I'm gathering is this I have to map each extension to a button,
whether physical on a 560m or 536m or virtual using the soft buttons on
the phone. What I was hoping for was something like the Directory app
http://voip-pbx/aastra/directory.php that came with the phone's firmware
that shows
2009/6/30 Carlos Chavez cur...@telecomabmex.com
On Tue, 2009-06-30 at 16:17 -0400, Jeremy Winder wrote:
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is
On Wed, Jul 1, 2009 at 1:10 AM, Olivieroza-4...@myamail.com wrote:
The 57i phone has 6 soft buttons which can show the status of at
least
16 phones (if you do not want to use the rest of the soft buttons which
would give you another 16).
Are you sure of that ?
How can you set more
2009/7/1 Jonathan Moore supermegat...@gmail.com
On Wed, Jul 1, 2009 at 1:10 AM, Olivieroza-4...@myamail.com wrote:
The 57i phone has 6 soft buttons which can show the status of at
least
16 phones (if you do not want to use the rest of the soft buttons which
would give you another
On Wed, Jul 1, 2009 at 4:40 PM, Olivieroza-4...@myamail.com wrote:
True but how can a single light be blinking because extension 1001 is
receiving a call and at the same time, be turned on because extension 1002
is on call ?
Maybe typing on Next button would alternatively show extension 1001
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is the inability to see
who is on the phone. Can someone point in the right direction to setup
an XML app on the
On Tue, Jun 30, 2009 at 4:17 PM, Jeremy Winder jwin...@logicalsi.comwrote:
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is the inability to see
who is on
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jeremy Winder wrote:
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is the inability to see
who is on the
On Tue, Jun 30, 2009 at 3:17 PM, Jeremy Winderjwin...@logicalsi.com wrote:
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is the inability to see
who is on
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Barry L. Kline wrote:
If you'd like a more generalized approach you can install an Openfile
server and use the Asterisk plugin. That'll give you an internal IM
server which will show the status you seek.
Sorry, not 'openfile' but 'openfire'.
On Tue, 2009-06-30 at 16:17 -0400, Jeremy Winder wrote:
I'm in the process of converting our current hybrid key system to
Asterisk and Aastra 57i phones. One of the features that seems to be a
show stopper for almost everyone in the office is the inability to see
who is on the phone. Can
You maybe using wrong username. If the user is defined in sip, you should be
able to register using the correct username and password. Also, see if
asterisk is listening on a defferent sip port instead of default 5060. If
its different use that port.
On Wed, Sep 24, 2008 at 3:32 AM, michel freiha
Hi all,
I have the below extension defined under sip.conf:
[2203]
type=friend
username=2203
secret=123456
host=192.168.0.164
mailbox=2203
context=intern
canreinvite=yes
dtmfmode=rfc2833
When trying to register from a softphone installed on a PC behind a nat with
IP=192.168.0.164, I got 503
Make host=dynamic.
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 2104-7000
- michel freiha
michel freiha a écrit :
Hi all,
Hi
I have the below extension defined under sip.conf:
[2203]
type=friend
username=2203
secret=123456
host=192.168.0.164
mailbox=2203
context=intern
canreinvite=yes
dtmfmode=rfc2833
When trying to register from a softphone installed on a PC behind a
michel freiha wrote:
[2203]
type=friend
username=2203
secret=123456
host=192.168.0.164 http://192.168.0.164
mailbox=2203
context=intern
canreinvite=yes
dtmfmode=rfc2833
You've forgotten nat=yes. You'll also want to specify a context on your
mailbox line. (i.e. [EMAIL PROTECTED])
Doug
With host=dynamic it's working fine...I need to force the user to use his
extension from one IP address and not from different IP addresses
Regards
On Tue, Sep 23, 2008 at 3:40 PM, Vinícius Fontes [EMAIL PROTECTED]wrote:
Make host=dynamic.
Atenciosamente,
Vinícius Fontes
Núcleo de
On Sep 23, 2008, at 8:40 AM, Vinícius Fontes wrote:
Make host=dynamic.
Also, set nat=yes
Hi all,
I have the below extension defined under sip.conf:
[2203]
type=friend
username=2203
secret=123456
host= 192.168.0.164
mailbox=2203
context=intern
canreinvite=yes
dtmfmode=rfc2833
Fred
If I make host=dynamic, then the customer will be able to register on my
asterisk server from any IP address...What I need is to force the User to
register on asterisk from a specific IP address like 192.168.0.164...How
this could be done?
Regards
On Tue, Sep 23, 2008 at 3:52 PM, Fred Posner
The user won't need to register at all, registration is only good if
the ip address changes. Much simpler that way. Just put host=the ip
address you want
on Tuesday 09/23/2008 michel freiha([EMAIL PROTECTED]) wrote
If I make host=dynamic, then the customer will be able to register on my
Is there a way to register to asterisk only from a specific IP address,
which mean the customer can use his extension only from one IP address?
Regards
On Tue, Sep 23, 2008 at 3:49 PM, Administrator TOOTAI [EMAIL PROTECTED]wrote:
michel freiha a écrit :
Hi all,
Hi
I have the below
On Sep 23, 2008, at 9:03 AM, michel freiha wrote:
Is there a way to register to asterisk only from a specific IP
address, which mean the customer can use his extension only from one
IP address?
host=192.168.0.164
Yes, use the external IP that the client is sending you instead of the
]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Extension registration
If I make host=dynamic, then the customer will be able to register on my
asterisk server from any IP address...What I need is to force the User to
register on asterisk from a specific IP address like
Fred,
The context should stay friend or i should change it to another thing?
Regards
On Tue, Sep 23, 2008 at 4:59 PM, Fred Posner [EMAIL PROTECTED] wrote:
On Sep 23, 2008, at 9:03 AM, michel freiha wrote:
Is there a way to register to asterisk only from a specific IP address,
which
Fred,
The context should stay friend or i should change it to another thing?
Regards
This would depend on what you want that user to be able to do...
Here's a good source to learn the differences:
http://www.voip-info.org/wiki/view/Asterisk+sip+type
Fred Posner
[EMAIL PROTECTED]
Hi all,
I need please the exact extension definition under extensions.conf that
accepts any call coming from an appropriate username and Ip address...This
mean that the authentication should be done on username and IP address
Regards
___
-- Bandwidth
On Tue, 23 Sep 2008, michel freiha wrote:
I need please the exact extension definition under extensions.conf that
accepts any call coming from an appropriate username and Ip address...This
mean that the authentication should be done on username and IP address
Guessing based on the information
This is done in sip.conf, iax.conf, etc, not in extensions.conf. By the
time a call gets to extensions.conf it must already be authenticated.
Assume the username is robertdobbs and the ip is 209.17.71.61
In sip.conf you would have something like this:
[robertdobbs]
deny=0.0.0.0/0
Hello Eric,
i didwhat you asked me to do but i'm getting Notfound sip message when
trying to register
regrads
On Tue, Sep 23, 2008 at 9:56 PM, Eric ManxPower Wieling [EMAIL
PROTECTED]wrote:
This is done in sip.conf, iax.conf, etc, not in extensions.conf. By the
time a call gets to
Dear All,
I have the following scenario...When a customer dial 111 number a beep
message will iplay in order to record and playback his voice...Else he'll be
routed to another call flow as you can see in the context below:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ?
Hi Michel,
Am Freitag, den 12.09.2008, 17:41 +0300 schrieb michel freiha:
Dear All,
I have the following scenario...When a customer dial 111 number a beep
message will iplay in order to record and playback his voice...Else
he'll be routed to another call flow as you can see in the context
Thanks :-D change the context to default and everithing works fine.
I assigned the sip context because that was the context on the example.
Thanks :-)
Nomar
Alex Balashov wrote:
Nomar Mora wrote:
Alex Balashov wrote:
Do you have dial plan routes for internal extension calls?
Dear Randulo,
Thanks for your suggention.
Now i am able to communicate between 2 computers.
Regards,
Baskar
--- randulo [EMAIL PROTECTED] wrote:
On Mon, May 19, 2008 at 8:44 AM, bas karan
[EMAIL PROTECTED] wrote:
[May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879
handle_request_invite: Call
Good day:
I recently install asterisk-now. Setup a pair of SipXpert 160 phones and
all fine. Later I try to setup some Octtels VoIP Gateways SP4220. I
config the proxy setings like this:
http://www.fundacitetachira.gob.ve/settings/Settings.png
the sip.conf entrys are like the asterisk manual
Do you have dial plan routes for internal extension calls?
Nomar Mora wrote:
Good day:
I recently install asterisk-now. Setup a pair of SipXpert 160 phones and
all fine. Later I try to setup some Octtels VoIP Gateways SP4220. I
config the proxy setings like this:
Alex Balashov wrote:
Do you have dial plan routes for internal extension calls?
Do you mean if I have configured the extension.conf? Yes, I config the
extensions on the extension.conf file
otherwise, no I have not.
Thanks in Advance
Nomar
--
2008 Año del satélite Simón Bolívar
Nomar Mora wrote:
Alex Balashov wrote:
Do you have dial plan routes for internal extension calls?
Do you mean if I have configured the extension.conf? Yes, I config the
extensions on the extension.conf file
otherwise, no I have not.
Thanks in Advance
Nomar
In the 'sip' context?
Dear Friends,
This is Baskar from Chennai, trying to configure
asterisk. Now I planned to start with communication
between 2 systems using soft phones.
When I tried to call the other computer I am getting
the following error message on asterisk terminal,
Connected to Asterisk 1.4.18 currently
On Mon, May 19, 2008 at 8:44 AM, bas karan [EMAIL PROTECTED] wrote:
[May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879
handle_request_invite: Call from 'Phone3' to extension
'5' rejected because extension not found.
-- Registered SIP 'Phone3' at 192.168.1.101 port
Extension.conf enteries
Dear Randulo,
Thanks for your replay.
I am new to this concept, Could you explain me little
bit extra please?
Thanks Regards,
Baskar
--- randulo [EMAIL PROTECTED] wrote:
On Mon, May 19, 2008 at 8:44 AM, bas karan
[EMAIL PROTECTED] wrote:
[May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879
On Mon, May 19, 2008 at 11:36 AM, bas karan [EMAIL PROTECTED] wrote:
I am new to this concept, Could you explain me little
bit extra please?
You will need to put extensions in contexts. The context is a
fundamental concept of the dialplan. All extensions are inside a
context. In your sip.conf
Hi,
I'm having a little difficulty with my extensions setup.
What I'm trying to do is to have a PBX where I can call in to check
mail and call-out using the attached mobile or SIP phones.
If someone I know calls then they can be forwarded to me.
if it is someone I don't know then just ring the
To you extensions.conf gurus, I'd like some help on having a button/feature to
turn on/off system wide call forwarding.
I need the phone system to forward calls received, after the feature is
activated, to an answering service.
Calls received are on a PRI. I need all DIDs forwarded once the
Hi.
I'm trying to develop a module that emulates
the Cisco Extension Mobility feature from CallManager
(the ability to log in to a phone and temporarily
acquire the extension, soft key programming, and all
other settings for that user profile)
with Asterisk 1.4 and Cisco 79xx phones
(some with
Hi list,
Is there a limit on the length of an extension? I have an 18 byte long
extension, when issuing goto, Asterisk comes back with invalid
extension on the console. Anyone had this experience before?
___
--Bandwidth and Colocation Provided by
I am assuming you mean 18 digits long. it shouldnt be a problem you
mind posting your configs?
On 10/3/07, Wai Wu [EMAIL PROTECTED] wrote:
Hi list,
Is there a limit on the length of an extension? I have an 18 byte long
extension, when issuing goto, Asterisk comes back with invalid
extension
Hi,
I have spend allot of time searching a solution: We have different SIP
accounts that our Asterisk registers to, for example:
[general]
port=5060
disable=all
allow=[...]
srvlookup=yes
pedantic=no
context=start
language=de
register = 0123456789:[EMAIL PROTECTED]/someExtension
Problem 1:
Hi,
I have spend allot of time searching a solution: We have different SIP
accounts that our Asterisk registers to, for example:
[general]
port=5060
disable=all
allow=[...]
srvlookup=yes
pedantic=no
context=start
language=de
register = 0123456789:[EMAIL PROTECTED]/someExtension
Problem 1:
Jan 3 08:05:03 NOTICE[66269]: app_dial.c:1056 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
I end up getting this when I call from 2000 to 2001.
2000, 2002, and 2001 all exist in sip.conf and I connect using them.
I have all three setup to use the
Vulpes Velox wrote:
Jan 3 08:05:03 NOTICE[66269]: app_dial.c:1056 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
I end up getting this when I call from 2000 to 2001.
2000, 2002, and 2001 all exist in sip.conf and I connect using them.
I have all
I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card
connected to a POTS line and a phone set (physical extension). I've got
all incoming calls launching directly into an AGI script. I'd like to do
the same for the physical extension. In other words, when picking up the
hand
Time Bandit wrote:
I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card
connected to a POTS line and a phone set (physical extension). I've got
all incoming calls launching directly into an AGI script. I'd like to do
the same for the physical extension. In other words, when
I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card
connected to a POTS line and a phone set (physical extension). I've got
all incoming calls launching directly into an AGI script. I'd like to do
the same for the physical extension. In other words, when picking up the
hand
Here is my Extensions.conf file (Default Context). When an
individual calling in dials the extension, the response time seems
very slow. It doesn't immediately go to the next step, but hangs out
for a few seconds (silence)... Suggestions?
Thanks in advance... /pj
[default]
exten =
1 - 100 of 340 matches
Mail list logo
