Hi all,
I need to support this feature. When caller dial if the dial fail or no
answer from the
called number then play a music. So how to achieve that?
Thanks!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.c
Hi,
i have some spare (read: Boss get's a new one every few month ;)) Android
Phones laying around. Does someone know a way of using them as a mobile
gateway for asterisk? I could not find any SIP-Gateway in the Market, and i
don't think it's possible to use the GSM Audio directly with something l
Hi,
THIS IS IN DUBAI.
I am having PRI line with 100 DID's (00-99) and when we call to any landline
or mobile number then it shows us our board number or pilot number (i.e
4663000 means 00).. As i give all the extensions a particular DID, so people
from outside world can call them. The problem is t
On Monday 09 May 2011, mahesh katta wrote:
> Hi,
> THIS IS IN DUBAI.
>
> I am having PRI line with 100 DID's (00-99) and when we call to any
> landline or mobile number then it shows us our board number or pilot number
> (i.e 4663000 means 00)..
In the context through which outgoing calls are plac
On Fri, 2011-05-06 at 20:21 +0100, --[ UxBoD ]-- wrote:
> Are you not seeing issues with *8 call pick up then ?
Nope, I double checked it after seeing someone saying they had issues
with it and it is fine on the installation I have.
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161
Hi,
I have been away from the list for a bit so please forgive me if this has
already been covered.
>From what I understand if I have the t or T option in my dial string then the
>RTP must go through Asterisk since we need to know if any DTMF was pressed.
If both users and the server are not b
Jay,
AFAIK you can only use chan_datacard with a specific USB modem (I can not
recall the name at the moment).
I have tried it and it works well for both Audio and SMS messages.
You may want to try chan_mobile which works over blue tooth.
Regards,
Dovid
- Original Message -
From
John,
You want to do it only after it fails ?
If so you can do something like.
Exten => _X., 1, Dial(SIP/${EXTEN}@)
Exten => _X., 2, GotoIf($["${DIALSTATUS}" = "ANSWER"]?10)
Exten => _X., 4, MusicOnHold()
Exten => _X., 10, Hangup
- Original Message -
From: John Wu
To: Asterisk U
Alejandro,
What GUI are you using ? I don't think Asterisk comes with *30 to ban calls.
Regards,
Dovid
- Original Message -
From: "Alejandro Cabrera Obed"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, May 06, 2011 23:51
Subject: [asterisk-users] Black
Sir ,
this is not working
On Mon, May 9, 2011 at 1:52 PM, A J Stiles wrote:
> On Monday 09 May 2011, mahesh katta wrote:
> > Hi,
> > THIS IS IN DUBAI.
> >
> > I am having PRI line with 100 DID's (00-99) and when we call to any
> > landline or mobile number then it shows us our board number or p
Oh yeah - love your idea :-)
So just to clarify - I take it the Cisco phones (at least the 7940) are
supposed to be run with a tftp server available at all time - not only
during the initial configuration? Just making sure I didn't miss
something obvious in the documentation. Can somebody conf
On Mon, May 9, 2011 at 9:47 AM, Jay R. Worthington
wrote:
> gateway for asterisk? I could not find any SIP-Gateway in the Market, and i
Portech has made GSM and CDMA gateways for years - nothing that works
with your "old" Android phones, though.
http://www.portech.com.tw/p3-product1.asp?Cid=6
Sebastian Arcus wrote:
Cisco phones (at least the 7940) are supposed to be run with a tftp
server available at all time
That is my experience. But, if you're running tftp under Linux, then
it's probably spawned by xinetd and won't be running unless the service
is requested.
Doug
--
Ben F
>
> Are you not seeing issues with *8 call pick up then ?
> --
> Thanks, Phil
>
https://reviewboard.asterisk.org/r/1185/ helps with *8 pickup issues,
particulary when you have pickupsounds enabled.
Alec
--
_
-- Bandwidth a
Thanks to all for reply,
I have already put 1.8 in production. Actually we are using basic
function so I hope we are good and fingurs cross.
--
Sent from my iPhone
On May 9, 2011, at 7:18 AM, Alec Davis wrote:
Are you not seeing issues with *8 call pick up then ?
--
Thanks, Phil
https
On 05/09/2011 12:02 PM, Doug Lytle wrote:
Sebastian Arcus wrote:
Cisco phones (at least the 7940) are supposed to be run with a tftp
server available at all time
That is my experience. But, if you're running tftp under Linux, then
it's probably spawned by xinetd and won't be running unless t
Hi,
I would be curious to play with an Android phone with Wifi-only capability.
My plan is to install Bria on it and see if it could be used within a couple
of WiFi access points, as a high-end wireless phone.
Which handset would you recommend ?
Regards
--
___
Chan_datacard can selected model of huawei usb modem for voice and sms.
Chan_mobile on the other hand use bluetooth connection for voice. I have not
tried sms. For mobile phone, it seeMs nokia is quite good.
--- Sent with System SEVEN - the new generation of mobile messaging
-original message-
On 05/09/2011 08:47 AM, Jay R. Worthington wrote:
Hi,
i have some spare (read: Boss get's a new one every few month ;))
Android Phones laying around. Does someone know a way of using them as a
mobile gateway for asterisk? I could not find any SIP-Gateway in the
Market, and i don't think it's p
Lots of Android handsets support wifi, like my G2, aka HTC DesireZ.
Michael Graves
mgraves mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype mjgraves
> Original Message
> Subject: [asterisk-users] OT - Which Android handset with Wifi-only ?
> From: Ol
Dear
I have a small pbx with asterisk 1.6.2.16.
I have a funny problem, there is exactly 40sec between dial execution and
sending first invite packet on sip.
do you have any idea where the problem is ?
Best regards
--
Pezhman Lali
--
__
Dear Dovis, I'm using Elastix and the dialplan comes with this line:
*30,1,Goto(app-blacklist-add,s,1)
Any idea ??? Thanks a lot.
2011/5/9 Dovid Bender :
> Alejandro,
>
> What GUI are you using ? I don't think Asterisk comes with *30 to ban calls.
>
> Regards,
>
> Dovid
>
> - Original Messag
On Mon, May 9, 2011 at 2:20 PM, wrote:
> Lots of Android handsets support wifi, like my G2, aka HTC DesireZ.
Wouldn't ANY modern one have wifi? That would be odd if it didn't, would it not?
:r
--
_
-- Bandwidth and Colocation
Try the Elastix forums.
- Original Message -
From: "Alejandro Cabrera Obed"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, May 09, 2011 15:35
Subject: Re: [asterisk-users] Blacklist with *30
Dear Dovis, I'm using Elastix and the dialplan comes with thi
Hi,
Has anyone ever tried getting the Audio of ustream (ustream.tv) in to Asterisk
for MOH ?
Regards,
Dovid
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introduc
I have tested the following dialplan and it could be used as a
starting point. What you have to resolve is how to generate different
MeetMe conference room - in the example we have only one room = 1234
If you prefix the dialled extension with 1 => you will have a "lovely
chat". With 2 -> "cursing
Hi !
We curently have a centos 5 / asterisk 1.4 server that we have some DTMF
problems with. It has a Sangoma A104d card and only port one is used to
connect to the PSTN. Port 2 is conencted via a cross-over cable to a RAS for
modem access and port 3 is connected for data communication via PPP
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Jason Parker
> Sent: 06 May 2011 20:01
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Cannot install dahdi-li
Updated dialplan: fix a typo when using MOH variable and now you have
truly dynamic conference rooms.
Have fun,
Ioan.
+
exten => _[12]XXX,1,Set(__MM=${EPOCH})
exten => _1XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,love,1))
exten => _2XXX,n,Dial(SIP/${EXTEN:1},,G
Make sure the firmware on the card is latest. I had a problem, not like your,
and flashing the card to the latest firmware resolved it.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 9, 2011, at 6:11 AM, Nicolas Ross wrote:
> Hi !
>
> We curently have a centos 5
Hi,
have you tried to manage all with dialplane ?
just an example:
[incoming]
**exten => s,1,Dial (SIP/your_called_party,20)
exten => s,n, Playback(music_message)
.
In the first step the call is redirect to the configured called party
and if without answer (busy, not logged, not answered
Make sure the firmware on the card is latest. I had a problem, not like
your, and flashing the card to the latest firmware resolved it.
I did the upgrade, I will make another test when appropriate.
I will also upgrade my curent card, I am curent at version 25, wich dates <
2007, it might solve
Hi,
Apologies if this is a duplicate - been having mail server issues and I don't
think I managed to send it when I tried this morning.
It seems there is no .conf syntax highlighting script available for gedit. I'm
thinking of putting one together myself, but don't want to reinvent the wheel.
On 05/09/2011 10:32 AM, Naomi Rosenberg wrote:
So I'm just enquiring if anyone knows of one that already exists
that i've missed.
Not to inflame editor-related passions, but vim does quite a good job.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta
On Mon, 2011-05-09 at 15:32 +0100, Naomi Rosenberg wrote:
> Hi,
>
> Apologies if this is a duplicate - been having mail server issues and I don't
> think I managed to send it when I tried this morning.
>
> It seems there is no .conf syntax highlighting script available for gedit.
> I'm thinking
On Mon, May 9, 2011 at 7:07 AM, Sebastian Arcus wrote:
>
>
> On 05/09/2011 12:02 PM, Doug Lytle wrote:
>
>> Sebastian Arcus wrote:
>>
>>> Cisco phones (at least the 7940) are supposed to be run with a tftp
>>> server available at all time
>>>
>>
>> That is my experience. But, if you're running tf
On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali wrote:
> Dear
> I have a small pbx with asterisk 1.6.2.16.
> I have a funny problem, there is exactly 40sec between dial execution and
> sending first invite packet on sip.
> do you have any idea where the problem is ?
>
Check the dial timeout on your
Hello
Do you set your callerid in the context outgoing?
[outgoing]
exten => _X.,1,Set(CALLERID(num)=4663000)
exten => _X.,n,Dial(..
On Mon, May 9, 2011 at 4:45 AM, mahesh katta wrote:
> Sir ,
>
> this is not working
>
>
> On Mon, May 9, 2011 at 1:52 PM, A J Stiles
> wrote:
>
>> On
On 5/9/11 6:02 AM, "Doug Lytle" wrote:
>Sebastian Arcus wrote:
>> Cisco phones (at least the 7940) are supposed to be run with a tftp
>> server available at all time
>
>That is my experience. But, if you're running tftp under Linux, then
>it's probably spawned by xinetd and won't be running un
no worries, i'm not as passionate as some!
It just happens gedit is the one I've gravitated towards, despite what I'm sure
are good reasons to use something more hardcore. And I also fancy the project
of writing the highlighting script, it would be a nice little job for me and
I'm sure there
Thanks for the input. Long ago the CDR showed "asterisk" as the CLID but it
doesn't anymore so I am puzzled now how to even stop taking calls because my
CLID is now blank and I can't refuse any call with no CLID.
*WARNING[11002] chan_dahdi.c: CallerID returned with error on channel
'DAHDI/2-1'*
H
Dear All
Can anyone let me know where i can free sound file whcih i can use for
system monitoring alrams.
Regards
Amit--
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a li
On 05/09/2011 03:40 PM, Warren Selby wrote:
Thanks for the reply. No, I run tftpd directly from rc.local script
(on Slackware). That's fine - I just wanted to know I wasn't doing
something wrong. If everybody else is in the same boat - I'll just
be along for the ride then :D
On Mon, May 9, 2011 at 10:48 AM, Sebastian Arcus wrote:
> That's strange. Mine get stuck on the booting phase, looking for the tftp
> server, if they can't find it there. Even if I change the dhcpd option not
> to pass out any tftp server. Any ideas what did you configure differently?
>
> Sebasti
- Original Message -
> On Fri, 2011-05-06 at 20:21 +0100, --[ UxBoD ]-- wrote:
> > Are you not seeing issues with *8 call pick up then ?
>
> Nope, I double checked it after seeing someone saying they had issues
> with it and it is fine on the installation I have.
>
Which release are you
On 05/09/2011 04:50 PM, Warren Selby wrote:
On Mon, May 9, 2011 at 10:48 AM, Sebastian Arcus mailto:s...@open-t.co.uk>> wrote:
That's strange. Mine get stuck on the booting phase, looking for the
tftp server, if they can't find it there. Even if I change the dhcpd
option not to pas
> Which release are you running as this is still open
> https://issues.asterisk.org/view.php?id=18654
> --
> Thanks, Phil
I am using current SVN branch 1.8 and We aren't using above call pickup
features.
> _
> -- Bandwidth
On Monday 09 May 2011, Cassius Smith wrote:
> On 5/9/11 6:02 AM, "Doug Lytle" wrote:
> >Sebastian Arcus wrote:
> >> Cisco phones (at least the 7940) are supposed to be run with a tftp
> >> server available at all time
> >
> >That is my experience. But, if you're running tftp under Linux, then
> >
Hi,
It seems there is no .conf syntax highlighting script available for gedit. I'm
thinking of putting one together myself, but don't want to reinvent the wheel.
So I'm just enquiring if anyone knows of one that already exists that i've
missed.
Thanks
Naomi Rosenberg
www.servicesforasterisk
2011/5/9 randulo
> On Mon, May 9, 2011 at 2:20 PM, wrote:
> > Lots of Android handsets support wifi, like my G2, aka HTC DesireZ.
>
> Wouldn't ANY modern one have wifi? That would be odd if it didn't, would it
> not?
>
>
Yes, of course, all dual-mode phones support WiFi but :
1. I'm not certain
On 05/10/2011 12:55 AM, Olivier wrote:
2011/5/9 randulo mailto:rand...@randulo.com>>
On Mon, May 9, 2011 at 2:20 PM, mailto:mgra...@mstvp.com>> wrote:
> Lots of Android handsets support wifi, like my G2, aka HTC DesireZ.
Wouldn't ANY modern one have wifi? That would be odd if it di
Hey guys!
I have issue between iax vs iax2 following is my setup
asterisk-1.2 <--IAX>Asterisk-1.8
I am able to call from 1.8 to 1.2 over iax but no from asterisk 1.2 to 1.8
--
Hi,
I am using Asterisk 1.4.17 for my C4 routing but I am experiencing a high pdd
of around 20 seconds. Could you please help me to reduce it and what could be
the reason. Thanks
Abid Saleem --
On Mon, 9 May 2011, satish patel wrote:
Hey guys!
I have issue between iax vs iax2 following is my setup
asterisk-1.2 <--IAX>Asterisk-1.8
I am able to call from 1.8 to 1.2 over iax but no from asterisk 1.2 to 1.8
Might you be missing
requirecalltoken=no
Awesome!
root@:~# cat /etc/asterisk/iax.conf | grep requirecalltoken
; By setting 'requirecalltoken=no', call token validation becomes optional for
; that peer/user. By setting 'requirecalltoken=auto', call token validation
; can require it from this peer. So, requirecalltoken is internal
Hi,
It looks to me that the 401 unauth packets aren't getting back to the phones.
Which suggests a network/router/nat issue rather than anything wrong with the
asterisk or phone configuration.
Cheers,
Paul.
On 8 May 2011, at 01:59, GNUbie wrote:
> Hello all,
>
> I have installed the .deb
Hi All,
new to the list. Wondering if anyone has / knows of, a good rate importer
tool that can be used to standardize and normalize the ratesheets / rate
decks etc. obtained from various carriers so they can be analysed and
imported into a DB or be saved as a CSV or something?
Thanks so much in
I know most billing software sell this as a monthly service. You get cd-rom
every month where they have collected the published tarrif tables filed with
the FCC. You load it on the software to analyze call costs. I'm guessing this
is a lot of labor hours/manual work thus they charge for provi
On Mon, May 9, 2011 at 3:05 PM, Jason Aarons (AM) <
jason.aar...@dimensiondata.com> wrote:
> I know most billing software sell this as a monthly service. You get
> cd-rom every month where they have collected the published tarrif tables
> filed with the FCC. You load it on the software to analyz
For those of that are fans of stackoverflow.com, and stackexchange.com,
there's an effort to define a telephony stackexchange site. It's still in
the definition phase. What it needs to move forwards is more votes on
on/off topic questions, and perhaps some better questions to vote for or
agains
Dear, finally I implement the functionality code *94 in order to
access the blacklist menu from my own extension and put another
extension in the black list of Asterisk.
But after blacklisting a given extension, when I call from that
extension to my own extension the call always rings, it is not d
Anyone have some recommended equipment for alerting people to calls in a noisy
environment?
I have Polycom IP550 phones set up in some really noisy environments - our mine
hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now.
They're analog devices, attached to Linksy
On Mon, May 9, 2011 at 4:40 PM, Justin Sherrill
wrote:
> Anyone have some recommended equipment for alerting people to calls in a
> noisy environment?
>
> I have Polycom IP550 phones set up in some really noisy environments - our
> mine hoists - and they tend to drown out the ringers. I'm using
Hi,
I can't figure out a way of achieving what I want to do with the
voicemail feature. I thought I'd ask here to see if there are any
creative solutions that I have not considered.
What I want to do is have a message that says "Press 1 for Dick, or 2
for Jane". Then, depending on which numb
I'm not sure why but my call is being ended when I SendDTMF(*).
I'm using agi to originate a call and set the context,extension,priority to
test,1,1 respectively. I've got the following in my extensions.conf:
[test]
exten => 1,1,Answer();
same =>n,Wait(5);
same =>n,Verbose(1, Sending *);
On Mon, May 09, 2011 at 03:00:19PM -0600, John Marvin wrote:
>However, I want to record what is "said" during that time and send it
>to a third voicemail box once the caller hangs up without having
>pressed 1 or 2.
You could use Monitor to record the whole call, then use an AGI to do
something wit
On 5/9/2011 3:08 PM, Roger Burton West wrote:
You could use Monitor to record the whole call, then use an AGI to do
something with it on hangup if the other conditions haven't been
satisfied...?
I understand how to do the first part, and I at least understand that I
could do something fancy
Hi Phil,
Happily running with the following here:
dom0: Debian Lenny Xen 3.2-1 2.6.26-2-xen-amd64
domU: Asterisk 1.4 Debian Lenny 2.6.26-2-xen-amd64
domU: Asterisk 1.6 Debian Squeeze 2.6.32-5-amd64 (which is a Xen-aware
kernel)
domU: Asterisk 1.8 Debian Squeeze 2.6.32-5-amd64 (which is a Xen-awar
On 10/05/11 2:32 AM, Naomi Rosenberg wrote:
Hi,
Apologies if this is a duplicate - been having mail server issues and I don't
think I managed to send it when I tried this morning.
It seems there is no .conf syntax highlighting script available for gedit. I'm
thinking of putting one together m
Hi,
> new to the list. Wondering if anyone has / knows of, a good rate importer
> tool that can be used to standardize and normalize the ratesheets / rate
> decks etc. obtained from various carriers so they can be analysed and
> imported into a DB or be saved as a CSV or something?
I'm using a2bi
On Mon, May 9, 2011 at 7:58 PM, Markus wrote:
> Hi,
>
> > new to the list. Wondering if anyone has / knows of, a good rate importer
> > tool that can be used to standardize and normalize the ratesheets / rate
> > decks etc. obtained from various carriers so they can be analysed and
> > imported i
Has anyone else noticed this?
v/r,
Me
On Fri, May 6, 2011 at 12:11 PM, Louis Carreiro wrote:
> Has anyone else noticed that QueueCallerAbandon is not showing up in the
> AMI after the 1.8.3.3? Am I missing something? I'm getting what seems like
> everything else but QueueCallerAbandon.
>
> v/r
Hi,
I'm hoping someone has a suggestion for us.
We have an ITSP that sends inbound traffic to us. Unannounced to us last
week they started alternately sending traffic from two IP addresses, instead
of the one we knew about. Some calls would pass, and others would be dumped
as unauthenti
You need to give the two sip.conf peers different names (in square
brackets).
On 05/09/2011 09:12 PM, Claude Hayn wrote:
Hi,
I'm hoping someone has a suggestion for us.
We have an ITSP that sends inbound traffic to us. Unannounced to us
last week they started alternately sending traffic from
hi:
our current connection is below:
sip phone<--->asterisk<>alcatel PBX<>PSTN
asterisk and alcatel PBX is connected via E1 isdn-pri.
when I use sip phone to dial outside PSTN world:
1. with 1.4 it is fine.
2. with 1.6.2, I need to set prematuremedia=no is sip.conf.
Apply this patch https://issues.asterisk.org/view.php?id=18868
--
Sent from my iPhone
On May 9, 2011, at 9:57 PM, d tbsky wrote:
hi:
our current connection is below:
sip phone<--->asterisk<>alcatel PBX<>PSTN
asterisk and alcatel PBX is connected via E1 isdn-pri.
when I
hi:
thanks a lot for your quick reply. I saw that patch and think that
it was already included in 1.8.3.
now I know it will be included in 1.8.5.
I will try it and thanks again for your kindly help!!
2011/5/10 Satish Patel :
> Apply this patch https://issues.asterisk.org/view.php?id=18868
>
Thanks a lot loan. Will try it today.
Cheers
On Mon, May 9, 2011 at 6:25 PM, Ioan Indreias wrote:
> Updated dialplan: fix a typo when using MOH variable and now you have
> truly dynamic conference rooms.
>
> Have fun,
> Ioan.
>
> +
> exten => _[12]XXX,1,S
2011/5/9 Olivier
>
> 2011/5/9 randulo
>
>> On Mon, May 9, 2011 at 2:20 PM, wrote:
>> > Lots of Android handsets support wifi, like my G2, aka HTC DesireZ.
>>
>> Wouldn't ANY modern one have wifi? That would be odd if it didn't, would
>> it not?
>>
>>
> Yes, of course, all dual-mode phones supp
sir,
Below configuration i wase made in server . but this is not working.
exten => _90X,1,NoOp(${CALLERID(num)})
exten => _90X/5001,2,Set(CALLERID(name)=44578999)
exten => _90X,3,AGI(agi://127.0.0.1:4577/call_log)
exten => _90X/5001,4,Set(CALLERID(num)=44578999)
e
Hi Mahesh,
I have solutions but its paid, i can provide setting if you are interested
on counsultancy
Cheers
Dhaval
On Tue, May 10, 2011 at 10:45 AM, mahesh katta wrote:
> sir,
>
> Below configuration i wase made in server . but this is not working.
>
>
> exten => _90X,1,NoOp(${CALLERI
Hi Daval,
how much you require for this .
On Tue, May 10, 2011 at 12:00 PM, DHAVAL INDRODIYA wrote:
> Hi Mahesh,
>
> I have solutions but its paid, i can provide setting if you are
> interested on counsultancy
>
> Cheers
> Dhaval
>
>
> On Tue, May 10, 2011 at 10:45 AM, mahesh katta
> wrote:
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