Does Asterisk 1.6 fully support RFC4235?
Or is it the same implementation as 1.4?
Thanks.
-- James
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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 ABE in
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
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 for a single
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.
hi,
the -g option is right.
make sure that the system allows core files (ulimit -a).
Regards
--
Marcus
De: Gustavo A Gonzalez ggonza...@despegar.com
Para: asterisk-users@lists.digium.com
Enviadas: Quinta-feira, 30 de Julho de 2009 11:17:50
Assunto: Re:
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 network
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 call
[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
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
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
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 pay
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 calls back
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
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
The first time is always free :)
On Thu, Jul 30, 2009 at 1:50 PM, John Todd jt...@digium.com 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
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.
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.
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
On Thu, Jul 30, 2009 at 1:19 AM, hadi motamedimotamed...@gmail.com 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
On 7/30/09, Steve Totaro stot...@asteriskhelpdesk.com wrote:
The first time is always free :)
On Thu, Jul 30, 2009 at 1:50 PM, John Todd jt...@digium.com 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
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