I suspected as much :)
Well, it IS a calling card; people call an access number, dial an
international number.
Assuming typical ALOC to be 8 mins which is seen quite often in
International calls esp. in ethnic communities, and since the service
hasn't launched yet, it's hard to tell what the
I believe one of the patches involved in fixing for The Great Voicemail
Problem* about a year ago was to make voicemail automatically renumber the
mailbox files if it saw a gap.
* from memory: The Great Voicemail Problem is a bug where if you received a new
voicemail while listening to a
Hi Guys,
Seeing an issue with 1.6.2.17.2 and also 1.6.2.14
When we do call forwarding if the call coming in to be forwarded
asterisk sends the invite out to our ITSP as
username@anonymous.invalid instead of username@domain.
When call comes in with CLI and is forwarded it sends it as
I use find on a cron schedule to remove old recordings everyday. Im sure
you can do the same
find -H /var/log/asterisk/asterisk_rec/* -mtime +90 -type f -exec rm -v {}
\;
anything older than 90 days
On 27 May 2012 09:20, Eric Wieling ewiel...@nyigc.com wrote:
I believe one of the patches
I can't receive an incoming call from a DID provider to my NATted Asterisk box.
I'm testing this by dialling my DID with Skype, since I can't dial it from my
mobile phone (as it's an iNum). I specified the public IP to Asterisk using
externhost but also tried externip, and it didn't help. I
The users list probably isn't the best place for this discussion. Send me a
note directly if you like.
--Don
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail]
Sent: Sunday, May 27, 2012 1:28 AM
To: Asterisk Users Mailing List
On Sat, May 26, 2012 at 9:52 AM, Moises Silva moises.si...@gmail.comwrote:
There is nothing hybrid like that (GSM + Analog) in the NorthAmerica or
Europe to my knowledge. We at Sangoma (from Canada) have a 4-port GSM card
though which uses chan_dahdi (patching needed at the moment).
Hi list,
we are upgrading our Asterisk production server from 1.6.24 to 1.8.12
version and face the following problem: one of our peer
(voicetrading.com) doesn't accept our calls anymore, we receive a
timeout error Packet timed out after 32000ms with no response.
Switching back to 1.6 make
Hi;
In Voicemail.conf
If I am using
format=h263|gsm ,and i want to store only audio , then it is not storing.In log
it shows that video is deposite less then 5 second. If i want to store video
and audio both then it will store properly.
If am using
format=gsm|h263 ,then my
