Hi Guys,
asterisk drive me crazy!
Now I have tried to use FreePBX but it require MySQL which I can not
install du to a conflict with PostgreSQL.
Does someone know, how to configure FreePBX to use PostgreSQL?
Or does someone know another Asterisk Web-Frontend, without Database?
It is realy
> >
> > exten => 123,1,Verbose(1,Test)
> > exten => 123,n,Set(CONNECTEDLINE(number,i)="555-555-")
> > exten => 123,n,Set(rclidname="TestingB <123-444->")
>
> This line is just setting an ordinary channel variable.
> What do you think is supposed to use this value?
>
> > exten => 123,n,Se
On 7/18/2012 2:27 AM, Jeremy Kister wrote:
I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10.
.. ok, if the system weren't Solaris - let's say it was Debian Linux,
what would be on the list of things to check for ?
--
Jeremy Kister
http://jeremy.kister.net./
--
_
Why would you NOT want the connectedline info sent immediately?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, July 18, 2012 12:24 PM
To: Asterisk Users Mailing List - Non-Comm
>> exten => 124,n,Set(CONNECTEDLINE(all,i)="Name <555-555->") instead of a
>> separate name and number priority.
An example of my line is:
Set(CONNECTEDLINE(all)="${cid.name}" <${ARG1}>)
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
> > I m trying to set my system to set a caller id using the diaplan when
> > calling an internal extension. In other words, when I dial Joe Smith s
> > extension I want my own phone to show Joe Smith 555 . I have sort of
> > managed that in the sense that my phone shows Joe Smith s caller id
> >
On 07/18/2012 10:51 AM, Eric Wieling wrote:
Thank you. While you are at it, ask them to document where the audio / data from "
fax set g711cap| t38cap on" is saved to. 8-)
That is documented in the CLI help for the commands themselves; the
capture files are placed into subdirectories of the
> I’m trying to set my system to set a caller id using the diaplan when
> calling an internal extension. In other words, when I dial Joe
> Smith’s extension I want my own phone to show “Joe Smith 555”. I
> have sort of managed that in the sense that my phone shows Joe
> Smith’s caller id based on h
Remove the ",i" to start with. Do you have the various rpid related options in
sip.conf set?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, July 18, 2012 12:08 PM
To: 'Asterisk Users Ma
Hi,
I'm trying to set my system to set a caller id using the diaplan when
calling an internal extension. In other words, when I dial Joe Smith's
extension I want my own phone to show "Joe Smith 555". I have sort of
managed that in the sense that my phone shows Joe Smith's caller id based on
hi
Thank you. While you are at it, ask them to document where the audio / data
from " fax set g711cap| t38cap on" is saved to. 8-)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Wednesday
On 07/18/2012 10:06 AM, Eric Wieling wrote:
We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13
The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is
supported, but when I enable it I get the message "res_fax_digium.c:1624
dgm_fax_new: V.3
We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13
The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf
indicate v34 is supported, but when I enable it I get the message
"res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored." Is
v34 on
On 07/18/2012 06:30 AM, Alejandro Recarey wrote:
Hi all, and thanks for taking the time to read this.
I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:
[fax]
exten => _X.,1,Se
On Thu, 2012-04-19 at 12:20 +0100, Ishfaq Malik wrote:
> Hi
>
> I'm having a problem with the entirety of a call being recorded in the
> following scenario
> I'm using asterisk 1.8.7.0
>
> Person A (asterisk peer) calls Person B (not on asterisk, real world
> number via a SIP trunk)
> Mixmonitor
On Wed, 2012-07-18 at 09:16 -0500, Matthew Jordan wrote:
>
> - Original Message -
> > From: "Ishfaq Malik"
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >
> > Sent: Wednesday, July 18, 2012 3:13:13 AM
> > Subject: Re: [asterisk-users] Inconsistency in CDR between N
On 07/18/2012 09:43 PM, Matthew Jordan wrote:
- Original Message -
From: "Alejandro Recarey"
To: "Asterisk Users Mailing List"
Sent: Wednesday, July 18, 2012 6:30:26 AM
Subject: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway
Hi all, and thanks for taking the time to read t
- Original Message -
> From: "Ishfaq Malik"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Wednesday, July 18, 2012 3:13:13 AM
> Subject: Re: [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY
> calls
>
> On Wed, 2012-07-11 at 15:08 +0100, Ish
- Original Message -
> From: "Alejandro Recarey"
> To: "Asterisk Users Mailing List"
> Sent: Wednesday, July 18, 2012 6:30:26 AM
> Subject: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway
>
> Hi all, and thanks for taking the time to read this.
>
> I am trying to configure A
You can use asterisk 1.6+ and libss7 for this functionality. Any
Digium or Sangoma card working ok on this setup. Currently i am using
it on both of them.
On Wed, Jul 18, 2012 at 5:14 PM, Ashish Agarwal wrote:
> Hello,
>
> Can someone give me an understanding about E1 with ISUP on CCS 7 signallin
you need to either use chan_ss7 or libss7.
Also look for mailing list archives of asterisk-ss7
Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.
Hello,
Can someone give me an understanding about E1 with ISUP on CCS 7
signalling? Is it possible with asterisk + digium card and how
Regards,
Ashish
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Ne
I forgot to ask:
Do I have to load "res_fax" or "app_fax" to use the T38 gateway capability?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar ever
Unfortunately not, I already tried different forms callerid(num). Always
the same error.
I came across this entry in asterisk changelogs - maybe an update of
asterisk will help.
* Asterisk 1.4.36-rc1 Released.
2010-08-20 16:46 + [r283048-283123] Richard Mudgett
* channels/chan_dah
Hi all, and thanks for taking the time to read this.
I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:
[fax]
exten => _X.,1,Set(FAXOPT(t38gateway)=yes,20)
exten => _X.,n,Dial(SI
Mebe your operator doesnt like the CallerID(num) set as NULL just remove
the callerid(num) statement and let the standard callerId get set by
network.
Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400
Hi List!
I have a Problem with Telecom Hungary, if I set a callforwarding on the
Snom, to an external number (mobile).
Versions: Asterisk version 1.4.35, libpri 1.4.11.4, dahdi 2.6.0, snom-7.7.30
When I call the Snom (Extension 68), it responds with "302 moved
temporarily", and Asterisk try to di
Hey Ioan,
thanks for your answer.
It helped a little bit but I have no idea what exactly could work wrong.
My new situation:
*CLI> originate SIP/123456789101112 application MusicOnHold
> == Using SIP RTP CoS mark 5
> -- Got SIP response 482 "Loop Detected" back from 192.168.0.102:5060
>
On Wed, 2012-07-11 at 15:08 +0100, Ishfaq Malik wrote:
> Hi
>
> I'm using asterisk 1.8.7
>
> My dialplan for an inbound number is along the lines of
>
> [default]
> exten => idenfier,1,Goto(specific-context,s,1)
>
> [specific-context]
> exten => s,1,NoOp()
> exten => s,2,Dial(SIP/some-extenion,
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