You can delete old files, it won't break anything. Also to prevent saving files
in multiple formats, edit voicemail.conf and change format parameter under
general.
--
Mehmet Avcioglu
meh...@activecom.net
On May 23, 2012, at 1:03 AM, Danny Dias wrote:
Thanks Jason,
But how to delete
Hello,
a client attempted to transfer a call today which failed and returned the
channel back to her. When this happened on the console we saw:
Got OK on REFER Notify message
the version that we are running is 1.8.9.2. Are you aware of any none issues
please with this version as I could
Hi
Can anyone help me with this error
Unable to execute 'dahdi_scan /etc/asterisk/dahdi_scan.conf'
i am using asterisk-gui 2.0 , asterisk 1.8, dahdi 2.6 and also my call
reached the destination but no voice is coming from destination my voice
reflects back
thanks
--
Hi, thanks for your answers...
Can i delete like this:
rm -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.*
Is that ok? will this break something?
A little doubt here, once the user hears the voicemail using the phone, the
message is automatically moved to Old folder, is
Please check out the scripts located in contrib/scripts
Regards
Hans
On 2012-05-23 11:42, Danny Dias wrote:
Hi, thanks for your answers...
Can i delete like this:
rm -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.*
Is that ok? will this break something?
A little
Hi Jared Kevin,
Thanks for taking the time to answer my questions. I wonder if I could just
be reading the tcpdump incorrectly? I'm still seeing rtp streams (and
Jared, I have modified the dial string to remove the L)..
Here's a screenshot of what I'm seeing in wireshark. I really appreciate
On Tue, May 22, 2012 at 11:32:19PM -0400, ft...@mindspring.com wrote:
The calls are routed just fine, but when a call is answered at one of
the extensions or externally (by a home telephone) the asterisk
extensions continue to ring one more time. Is there a way to have
Asterisk drop an incoming
Hi All;
I need to use Asterisk for 20 000 users, so which asterisk version to be used?
Is there asterisk version that supports 20,000 users on one hardware machine?
Can I use one strong hardware server i7 with 64 GB RAM and fast hard desk to
handle 20 000 users, and concurrent calls 2000? Or I
20.000 users is really a big number, as big as 2000 concurrent calls.
As previously stated on this list, it depends... it depends by the type of
calls for example. If all media is offloaded from the server letting the
phones to reinvite each other, than your server CAN support the call
volume. If
On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Hi,
I am unable to register vitelity SIP trunk, where its keep on sending
registration request, and I am using Asterisk 1.4.39.2, my registration
procedure as follows,
sip.conf
register =
Alejandro's setup looks correct; you can also get the correct config using
Vitelity's wizard tool for setting up the trunks.
The only thing I would add is that if your account is setup with a session
border controller you will need to use the SBC's IP address instead of the
IP the wizard gives
Word of warning - I have had a lot of issues with Vitelity's routing.
Lots of troubles to the Caribbean, lots of troubles with ordinary US 800
numbers (major corporations like Nicor, American Airlines).
Cheers,
Jeff LaCoursiere
SunFone
On Wed, 2012-05-23 at 09:22 -0500, Stephen J Alexander
On May 31, 2012 from approximately 9:00AM to 12:00PM (Central Daylight
Time, GMT-5), the servers that Digium uses to provide many services to
the Asterisk community will be relocated. This will mean that these
services will be unavailable during most, if not all, of this time
window. Once the
On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote:
Word of warning - I have had a lot of issues with Vitelity's routing.
Lots of troubles to the Caribbean, lots of troubles with ordinary US 800
numbers (major corporations like Nicor, American Airlines).
We had lot's
Roger Burton West wrote:
On Tue, May 22, 2012 at 11:32:19PM -0400, ft...@mindspring.com wrote:
The calls are routed just fine, but when a call is answered at one of
the extensions or externally (by a home telephone) the asterisk
extensions continue to ring one more time. Is there a way
Hello everyone,
Just thought to let you know of a weird issue in Asterisk 1.8.? + Dahdi
2.6.? (and 2.5.?).
When you specify any cadence in an app (Dial, Queue) then caller id does
not work.
For instance with the default cadences (everything commented out in
chan_dahdi.conf) :
Dial(DAHDI/54)
Hi Guys,
is there any way to disable all Asterisk Features? We are having false dtmf
detections and randon calls being put on-hold and suspect that dtmf
features is the cause.
Changing features.conf aparently keeps the default options. Since we dont
use it, is there any way to disable it?
On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote:
On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote:
Word of warning - I have had a lot of issues with Vitelity's routing.
Lots of troubles to the Caribbean, lots of troubles with ordinary US 800
numbers
On Wed, May 23, 2012 at 07:13:01PM +0300, Roeften wrote:
Hello everyone,
Just thought to let you know of a weird issue in Asterisk 1.8.? + Dahdi
2.6.? (and 2.5.?).
When you specify any cadence in an app (Dial, Queue) then caller id does
not work.
For instance with the default cadences
On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote:
On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com wrote:
[...]
Just wanted to point out that after experiences with dozens of
termination
On Wed, 2012-05-23 at 13:02 -0400, Alejandro Imass wrote:
On Wed, May 23, 2012 at 12:32 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Wed, 2012-05-23 at 11:00 -0400, Alejandro Imass wrote:
On Wed, May 23, 2012 at 10:40 AM, Jeff LaCoursiere j...@sunfone.com
wrote:
[...]
Just
Dear;
So it is a hardware issue and not software?
I am afraid that asterisk software it self is not able to support 20 000 users
and 2000 concurrent calls.
About the high availability: is there a method that if the first asterisk
server down, then the call will stay connected and failover to
you can find more details @AsteriskSCF project.
On Wed, May 23, 2012 at 11:16 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Dear;
So it is a hardware issue and not software?
I am afraid that asterisk software it self is not able to support 20 000
users and 2000 concurrent calls.
About the
We have an Asterisk server which connects to another Asterisk server
acting as a PSTN gateway. This gateway machine has Digium TE210P card
connected to a pair of PRIs.
For the most part, all is working well, however there are some specific
telephone numbers that my users have attempted to
We have an Asterisk server which connects to another Asterisk server
acting as a PSTN gateway. This gateway machine has Digium TE210P card
connected to a pair of PRIs.
For the most part, all is working well, however there are some
specific
telephone numbers that my users have attempted to
the solution lies in kamailio/opensips's despatcher module.
Sent from my iPhone
On 23 maj 2012, at 20:46, bilal ghayyad bilmar...@yahoo.com wrote:
Dear;
So it is a hardware issue and not software?
I am afraid that asterisk software it self is not able to support 20 000
users and 2000
On 05/23/2012 07:16 AM, bilal ghayyad wrote:
Hi All;
I need to use Asterisk for 20 000 users, so which asterisk version to be used?
Is there asterisk version that supports 20,000 users on one hardware machine?
Can I use one strong hardware server i7 with 64 GB RAM and fast hard desk to
On Wed, May 23, 2012 at 02:02:51PM +0500, p070075 Muhammad Atif Ramzan wrote:
Hi
Can anyone help me with this error
Unable to execute 'dahdi_scan /etc/asterisk/dahdi_scan.conf'
i am using asterisk-gui 2.0 , asterisk 1.8, dahdi 2.6 and also my call
reached the destination but no voice is
On 05/23/2012 02:59 PM, Richard Mudgett wrote:
We have an Asterisk server which connects to another Asterisk server
acting as a PSTN gateway. This gateway machine has Digium TE210P card
connected to a pair of PRIs.
For the most part, all is working well, however there are some
specific
Hello All,
I use IAX2 as the incoming connection from my DID provider. For whatever
reason, this works best for me, SIP connections lag very frequently and
only have about a 50% success rate for incoming calls (they get dropped
mysteriously).
I'm trying to implement a fax/voice switch. I have
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