Hello,
I am running Asterisk 1.4 with the RPID patch (8824) in order to display
the name of the *called* person. It works on Cisco and Polycom phones, but
not on SNOM. The local SNOM dealer claims that he saw it working, but he
cannot give me more details. Has anyone managed to make it working?
Kare,
Thanks much appreciated. It connected as soon as I created a SIP
account. However I must try and figure out as how to get this box use
IAX2.
The vendors are not very helpful.
Thanks for your help.
Regards
Manoj
From: asterisk-users-boun...@lists.digi
Gordon Henderson wrote:
> On Sat, 23 May 2009, Dunc wrote:
Hi Gordon, thanks for your reply, I was hoping to find someone who was
involved in the thread before.
See inline comments
>
>> Hi everyone,
>>
>> I just found this thread, which is amazing as I'm on my first go with
>> asterisk and
>> Hi Tiago,
>>
>> I have an OpenVox A400P11, it shows up like this...
>>
>> eddie ~ # lsdahdi
>> ### Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
>> 1 FXSFXSKS (EC: MG2) RED
>> 2 FXOFXOLS (EC: MG2)
>> Use of uninitialized value in string eq at
>> /u
2009/5/24 Manoj Panicker - FOES
> Kare,
> Thanks much appreciated. It connected as soon as I created a SIP
> account. However I must try and figure out as how to get this box use IAX2.
>
Are you sure the FritzBox actually supports IAX2? As far as I remember it
does not.
Christian
_
On Sun, 24 May 2009, Dunc wrote:
>> However I have 2 clients with Teleworst lines and I've never been able to
>> make caller ID work on them, even though teleworst insist they are
>> providing caller ID..
>
> Well that's a shame because I specifically went for NTL so that I
> haven't got BT to con
On Sun, 24 May 2009, Dunc wrote:
> extensions.conf
> ---
> And the same for extensions.conf..
>
> eddie asterisk # cat /etc/asterisk/extensions_dunc.conf
> [from-sip]
> exten => 07875123456,1,Dial(DAHDI/1/07875123456,20)
>
> [from-internal]
> exten => 07875123456,1,Dial(DAHDI/1/078
On Sun, May 24, 2009 at 12:36:26PM +0100, Dunc wrote:
> >> Hi Tiago,
> >>
> >> I have an OpenVox A400P11, it shows up like this...
> >>
> >> eddie ~ # lsdahdi
> >> ### Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
> >> 1 FXSFXSKS (EC: MG2) RED
> >> 2 FXOFXO
Hi ,
Any idea as how to divert the Incoming PSTN calls on the FritzBox to
one of the Numbers in the Asterisk domian? and vice versa.
I want ot use the FritzBox as the bridge between the PSTN and Astrisk
Thanks
Manoj
From: Manoj Panicker - FOES
Sent: 24 Ma
Hi,
I have a system with the following configuration:
- Asterisk 1.6.0.3-rc1
- DAHDI Linux 2.1.0.4
- DAHDI Tools 2.1.0.2
- wanpipe-3.3.15
- Sangoma A108E card with ISDN PRI interfaces
Calls arrive over the PRI interface and users need to supply DTMF digits
before asterisk decides what to do with
What's the bets way to verify T.38 is being used on both incoming an
outgoing transaction?
On Fri, May 22, 2009 at 8:08 PM, Lee Howard wrote:
> Todd S wrote:
> > Our call path is Sip trunk from MAX TNT - Asterisk - T1 - Adtran
> > endpoint converting sip trunk to copper line for house wiring.
>
Tzafrir Cohen wrote:
> On Sun, May 24, 2009 at 12:36:26PM +0100, Dunc wrote:
Hi Tiago,
I have an OpenVox A400P11, it shows up like this...
eddie ~ # lsdahdi
### Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
1 FXSFXSKS (EC: MG2) RED
On Sun, May 24, 2009 at 03:00:21PM +0100, Dunc wrote:
> Tzafrir Cohen wrote:
> > On Sun, May 24, 2009 at 12:36:26PM +0100, Dunc wrote:
> >> ; Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
> >> ;;; line="1 WCTDM/4/0"
> >> signalling=fxs_ks
> >
> > signalling=fxo_ks
> >
> >> callerid
Gordon Henderson wrote:
> On Sun, 24 May 2009, Dunc wrote:
>
>> extensions.conf
>> ---
>> And the same for extensions.conf..
>>
>> eddie asterisk # cat /etc/asterisk/extensions_dunc.conf
>> [from-sip]
>> exten => 07875123456,1,Dial(DAHDI/1/07875123456,20)
>>
>> [from-internal]
>> e
sean darcy wrote:
> I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 over this weekend.
>
> I'm getting:
>
I asked the same thing just over a week ago and didn't get a response.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
a
I can really only speak to questions 2;
The other 2 pins in your cable are for your second phone line - if you
have one that is. You should be able to plug in a 4 wire cable into any
socket in the house and get access to Both lines.
On Sat, 2009-05-23 at 16:02 +0100, Dunc wrote:
> Hi everyone,
On Sun, 24 May 2009, Mark Phillips wrote:
> I can really only speak to questions 2;
>
> The other 2 pins in your cable are for your second phone line - if you
> have one that is. You should be able to plug in a 4 wire cable into any
> socket in the house and get access to Both lines.
> Mark Phill
Can I run two instances of asterisk sharing a single te412p ?
I want to be able to have several asterisk servers (for testing various
scenarios) running on one server. I was wondering if these asterisk
processes could share a zaptel/dahdi card nicely.
Julian
__
On 24/05/09 19:21, Gordon Henderson wrote:
> Here's an example:
>
> http://www.wppltd.demon.co.uk/WPP/Wiring/UK_telephone/uk_telephone.html
>
> In these enlightened days, it's normal to not use the 3rd wire internally
> on extension as it has been known to degrade an ADSL signal.
I think this las
On Sun, May 24, 2009 at 08:01:53PM +0100, Julian Lyndon-Smith wrote:
> Can I run two instances of asterisk sharing a single te412p ?
>
> I want to be able to have several asterisk servers (for testing various
> scenarios) running on one server. I was wondering if these asterisk
> processes could
Yes, you can share it as long as you designate say the first 2 ports
to the first asterisk instance and the other 2 to another one.
Martin
On Sun, May 24, 2009 at 2:01 PM, Julian Lyndon-Smith wrote:
> Can I run two instances of asterisk sharing a single te412p ?
>
> I want to be able to have sev
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Tzafrir, Danny.
El jueves 21 de mayo del 2009 a las 06:55:14 -0300,
Tzafrir Cohen escribió:
>> > Mmmm... but I believe that it had done already in that order. In fact, I
>> > reviewed the existence of the module and it was in the directory. For th
Is there any method in Asterisk to enable the updating process
into another SQL database through entering IVR options during the call.
Thanks a lot.
_
Invite your mail contacts to join your friends list with Windows Live Spaces.
It
Torintino T schrieb:
> Is there any method in Asterisk to enable the updating process
> into another SQL database through entering IVR options during the call.
Depending on what you are trying to do there are various solutions:
Channel Event Logging (CEL) - http://www.asterisk.org/node/48358
AGI
S
On Sun, May 24, 2009 at 04:53:41PM -0300, Daniel Bareiro wrote:
> # cat /proc/dahdi/*
> Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
>
>1 WCTDM/4/0 RED
>2 WCTDM/4/1
>3 WCTDM/4/2
>4 WCTDM/4/3
>
>
> After to run dahdi_cfg -vvv:
>
> #
Todd S writes:
> What's the bets way to verify T.38 is being used on both incoming an
> outgoing transaction?
I don't know about "best", but there's sip set debug ip whatever, or my
personal favourite:
tcpdump -s0 -vnieth0 udp
You can easily tell if t38 is used because the UDP traffic will (al
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Tzafrir.
El domingo 24 de mayo del 2009 a las 17:33:36 -0300,
Tzafrir Cohen escribió:
>> # cat /proc/dahdi/*
>> Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
>>
>>1 WCTDM/4/0 RED
>>2 WCTDM/4/1
>>3
Alex Samad wrote:
> Hi
>
> I have an account with mynetphone (australia), which gives me two voip
> (sip) accounts, which i used to have connected to a spa9000.
>
> this is behind a firewall, so on the spa9000 I would listen on another
> port apart from 5060. so on the firewall 5060 would go to vo
Todd S wrote:
> What's the bets way to verify T.38 is being used on both incoming an
> outgoing transaction?
3 to 1 in favour of not working. ;)
PaulH
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing lis
On Mon, May 25, 2009 at 09:29:54AM +1000, Paul Hales wrote:
> Alex Samad wrote:
> > Hi
> >
> > I have an account with mynetphone (australia), which gives me two voip
> > (sip) accounts, which i used to have connected to a spa9000.
> >
> > this is behind a firewall, so on the spa9000 I would listen
Dear All,
Please help as I new to Asterisk. I want know something what is Trunk what
it will do. And I want to create a Dialplan like bellow:
1. It ask to dial a extension.
2. User will dial a extension.
3. User will be routed to that extension.
I also want to connect my asterisk phone
Farooq Hussain wrote:
> Dear All,
>
> Please help as I new to Asterisk. I want know something what is Trunk
> what it will do. And I want to create a Dialplan like bellow:
>
>1. It ask to dial a extension.
>2. User will dial a extension.
>3. User will be routed to that extension.
>
> I
32 matches
Mail list logo
