Hello,
Does Asterisk-GUI 2.0 compatible with Asterisk 1.6.1.1 ?
Sincerely,
Tseveen
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asteri
On 7/30/09, Steve Totaro wrote:
> The first time is always free :)
>
> On Thu, Jul 30, 2009 at 1:50 PM, John Todd wrote:
>
>>
>> I know many of you have been waiting for this for a while, so I'll
>> keep this short: The Skype for Asterisk Public Beta is now available
>> on the Digium store.
>>
>
On Thu, Jul 30, 2009 at 1:19 AM, hadi motamedi wrote:
> Thank you very much for your reply . But please be informed that our current
> line-outgoing route is being configured as the followings (in
> extensions.conf):
Set(TIMEOUT(digit)=timeout)
There's definitely more to your dialplan than the sa
Howdy,
Just installed a new switch in a new location (Ubuntu, 2.6.24-24 kernel,
zaptel 1.4.12.1 built from source, libpri-1.4.10.1 built from source,
asterisk 1.4.26 built from source, wanpipe 3.5.4 built from source,
Sangoma A104d with firmware that is probably a year old).
I plugged in an R
The following pastebin shows the inbound call, inbound INFO containing the
Remote-Party-ID string, and the SIP acknowledgement of the INFO. Asterisk
does not send the data from the Remote-Party-ID string on to the phone, nor
does it set the CALLERID(name) variable after receiving the message.
htt
Hi All,
I'm trying to test asterisk voicemail on recording my own unavailable
message, busy message or temporary message. I was looking at the console
and saw this message:
app_voicemail store_file Memory map failed
Then i looked at /var/spool/asterisk/ there were no recorded
greetings. w
The first time is always free :)
On Thu, Jul 30, 2009 at 1:50 PM, John Todd wrote:
>
> I know many of you have been waiting for this for a while, so I'll
> keep this short: The Skype for Asterisk Public Beta is now available
> on the Digium store.
>
> We are pleased to announce the open beta of
I have problems with it...
[Jul 30 14:34:17] NOTICE[30529]: core.cpp:1153 skype_cp_handler: Found license
'XX' providing 1 concurrent calls
[Jul 30 14:34:17] NOTICE[30529]: core.cpp:1000 display_host: Skype For Asterisk
Host-ID: X
[Jul 30 14:34:17] NOTICE[30529]: core.cpp:1320 sfa_st
I know many of you have been waiting for this for a while, so I'll
keep this short: The Skype for Asterisk Public Beta is now available
on the Digium store.
We are pleased to announce the open beta of Skype For Asterisk is
ready to begin and we look forward to you participation. To obtain
Alexandre Rodrigues escribió:
> Hello all,
>
> I am quite new in asterisk and I am trying to create a dialplan that
> executes the following steps:
>
> 1. A SIP friend dials 102 extension.
>
> 2. Asterisk PBX responds with some beeps.
>
> 3. The sip friend hangs up the phone.
>
> 4. Asterisk PBX ca
Lyle wrote:
>I had this issue with Teliax. Basically with SIP, Teliax could not (or
>the protocol won't let you) set your outbound caller ID via Asterisk.
>Caller ID is set on a per account basis with Teliax when using SIP(IAX
>was not working well for me with Teliax). So I have two outbound pa
Jeff LaCoursiere wrote:
>You don't have to send the traffic back to broadvoice for outbound if
>you
>don't want or need to. Perhaps you can send the home traffic to
>Broadvoice and pick another carrier to send your other outbound traffic
>to, perhaps one that won't be so picky about your ou
Hello all,
I am quite new in asterisk and I am trying to create a dialplan that
executes the following steps:
1. A SIP friend dials 102 extension.
2. Asterisk PBX responds with some beeps.
3. The sip friend hangs up the phone.
4. Asterisk PBX calls back to the sip friend after 30 seconds with
>>
>> [peer]
>> defaultip=xxx.xxx.xxx.xxx
>> host=xxx.xxx.xxx.xxx
>> deny=0.0.0.0/0.0.0.0
>> allow=xxx.xxx.xxx.0/255.255.255.0 < read what you've put!!! The
'allow' should be 'permit' as Jared already told you (and he should know
what he's talking about).
>> insecure=port,invite
>>
On Thu, 30 Jul 2009, Paulo Santos wrote:
> Hello everyone,
>
> I'm having a hard time configuring my router to forward asterisk traffic
> correctly. I have the following ports being forwarded to asterisk:
>
> 5060, 1-2
>
> Now, I can register the accounts when outside the network and I can
On Thu, 2009-07-30 at 16:19 +0100, Paulo Santos wrote:
> Hello everyone,
>
> I'm having a hard time configuring my router to forward asterisk traffic
> correctly. I have the following ports being forwarded to asterisk:
>
> 5060, 1-2
>
> Now, I can register the accounts when outside the
hi,
the -g option is right.
make sure that the system allows core files (ulimit -a).
Regards
--
Marcus
De: Gustavo A Gonzalez
Para: asterisk-users@lists.digium.com
Enviadas: Quinta-feira, 30 de Julho de 2009 11:17:50
Assunto: Re: [asterisk-users] Asterisk
Hello everyone,
I'm having a hard time configuring my router to forward asterisk traffic
correctly. I have the following ports being forwarded to asterisk:
5060, 1-2
Now, I can register the accounts when outside the network and I can call
every extension that is inside the network. The
Steve Edwards wrote:
> On Wed, 29 Jul 2009, Myles Wakeham wrote:
>
>
>> I have setup an Asterisk system for my home & home office.
>>
>
> [snip]
>
>
>> The cost of all these lines with analog carriers was getting ridiculous,
>> so I'm moving over to a SIP carrier. I created one account
Thanks Tzafrir for your answer. Because I had some problems running
safe_asterisk script to restart asterisk automatically in our callcenter ,
I've developed a simple script that runs from a schedule task and check if
asterisk is running each minute. This is not the best solution yet but it
works
Hello,
>
> For people having experienced Asterisk Business Edition, please I need some
> information:
>
> - First, Can ABE be installed in a Debian or Ubuntu OS 32 and 64 bit.
>
> - Second, Can ABE be installed in a newer version of Fedora like Fedora 10
> or 11.
>
> - Third, opcom, a reseller of
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