Hi,
I wonder what is the difference between Transfer and Dial applications?
Could somebody give me an example of Transfer usage? (documentation and
voip-info looks poor a bit).
I'm using Asterisk 1.2.5 if it matters.
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Aurimas Skirgaila
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On Fri, May 8, 2009 at 3:45 PM, Jeff LaCoursiere j...@jeff.net wrote:
On Fri, 8 May 2009, Aurimas Skirgaila wrote:
Despite the VPN overhead, running VOIP through VPN is good idea because
VPN
reorders encapsulated UDP packets in correct order. Security matters as
well.
Reorders? How
:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Aurimas Skirgaila
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and name of clients: For your information there is no problem related to
the
outbound call
An help would be appreciated
Kind Regards
Salah.
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any
effect,
bchan=1-15,17-31
dchan=16
[zapata.conf]
group=1
pridialplan = unknown
switchtype=euroisdn
context = trunk-1
signalling = pri_net
channel = 1-15,17-31
Hardware - Dell PowerEdge R200. Now moved onto barebone test server, but
same errors persist.
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Mvh,
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.
Regards,
Am
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http://www.eSlackware.com http://www.eslackware.com/
IRC: Khratos @ #asterisk / -doc / -bugs
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Hardware - Dell PowerEdge R200. Now moved onto barebone test server, but
same errors persist.
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Mvh,
Aurimas Skirgaila
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Hi,
I wonder if I may freely use the default soundfiles that came with asterisk
(fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server?
Are there any official sources of royalty free music?
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Aurimas Skirgaila
, Aurimas Skirgaila wrote:
Hi,
I wonder if I may freely use the default soundfiles that came with
asterisk
(fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server?
Are there any official sources of royalty free music?
http://downloads.asterisk.org/pub/telephony/sounds
/listinfo/asterisk-users
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to build SIP header specifying the final destination - the
internal IP address?
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On May 16, 2011, at 8:56 AM, Aurimas Skirgaila a.skirga...@gmail.com
wrote:
Hi,
long time ago, I came up with an optimal configuration set for my
environment - good detection and little false positives. Unfortunately some
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hi,
We have a phone number from third party provider which is used for inbound
calls. How could I monitor if this *phone number* is reachable?
the initial idea doesn't sound elegant:
- on my SIP server I set couple seconds of ringing before Answer().
- the monitoring server calls to that phone
tickets on issues
consistently before they know they have an issue.
Bryant
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*Sent*: Thursday, February 09, 2012 9:34 AM
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