Hey,
we're experiencing a weird problem with Asterisk 1.8.13.1
(1:1.8.13.1~dfsg1-3+deb7). Calls that leave and enter Asterisk via
a PBX (sipgate.de) work perfectly fine, almost 100% of the time.
However, calls that are routed to sipgate.de, which then routes the
call back to our Asterisk
Hi Ishfaq,
Am 25.07.2014 um 14:34 schrieb Ishfaq Malik i...@pack-net.co.uk:
Not that I know of but since you are using a database you can update multiple
rows at once.
Yep, or write a PHP script that fills in the required info automatically. I was
just wondering if there was any native
Thanks Patrick,
Assuming security+evaluation refers to Common Criteria,
Common Criteria is one, but not necessarily the only type of security
evaluation. Often times organizations with resources will perform an
evaluation against its own standards before adopting or accepting a
system. I was
Are there any quality Outlook integrations for asterisk out there? The
closest I'm finding is at http://camrivox.com and they don't support
Outlook 2013.
dw
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By chance, I managed to fig into this a bit and found the exact
moment when audio stops. It is exactly the moment when the
counterparty picks up and RTP debug output says:
Got RTP packet from46.244.255.146:8058 (type 00, seq 000680, ts
340914880, len 000160)
Sent RTP packet to
On 28-07-14 12:28, Jeffrey Walton wrote:
[snip]
Is there anything that includes the development process? I'm
interested in the secure development items and testing.
Info about the development of Asterisk can be found here:
http://asterisk.org/community/developers
