--- Vahan Yerkanian [EMAIL PROTECTED] wrote:
Andrew Kohlsmith wrote:
On Thursday 04 May 2006 11:31, Louis-David
Mitterrand wrote:
I've got this low-ping 100%-up dsl connection
between two asterisk
1.2.7.1 servers. And oftentimes one of them would
declare its opposite
UNREACHABLE.
--- Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
No, you are supposed to realize that a) this
software cost you nothing. Not
one penny. b) this software is user-supported.
This means that in order to
make it better you need to help. and c) we don't
owe you anything. Not a
thing.
I
--- Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
Are you specifying the remote Asterisk box by IP or
by hostname. If by
hostname, then specify it by IP. Asterisk's DNS
lookup support has issues.
In the trunk peer details in AMP I'd set host= to a
hostname. I've switched it to IP
--- Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
If Asterisk has a DNS lookup failure it will never
retry that lookup.
Never meaning until the next reload command is
issued, or until the next restart command is issued,
or until the next time the OS reboots, or until the
next time asterisk
--- Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
If Asterisk has a DNS lookup failure it will
never
retry that lookup.
Never meaning until the next reload command is
issued, or until the next restart command is
issued,
or until the next time the OS reboots, or until
the
next
Please please, if anybody has experience using
/var/spool/asterisk/outgoing/ with SIP and IAX2
trunks, please explain what's going wrong here.
If I make the file 2.call containing:
Channel: SIP/sipphone
MaxRetries: 1
RetryTime: 5
WaitTime: 10
Context: outgoingtest
Extension: 1747555
Priority:
Tim Panton [EMAIL PROTECTED] wrote:
I think you are misunderstanding the way call files
work.
They connect _2_ ends,
here's what the wiki says:
[snip]
So the _channel_ has to be the whole thing -
including the number
'far' you want to dial
the _extension_ and _context_ are the 'near'
Tim Panton [EMAIL PROTECTED] wrote:
I think you are misunderstanding the way call files
work.
They connect _2_ ends,
here's what the wiki says:
[snip]
So the _channel_ has to be the whole thing -
including the number
'far' you want to dial
the _extension_ and _context_ are the 'near'
--- Tim Panton [EMAIL PROTECTED] wrote:
If you are using IAX2, you don't need to port
forward the ports.
Just have PBX2 register _often_ and that will keep a
mapping in your
router.
Where is this set? Is this the minregexpire and
maxregexpire settings in iax.conf, which default to 60
--- Angus Comber [EMAIL PROTECTED] wrote:
I have been asked by a client to process a list of
telephone numbers.
Asterisk should call each number in turn and if the
recipient of the call
answers, play a message - eg from a wav.
How would I go about doing that?
Make your message as
--- Damon Estep [EMAIL PROTECTED] wrote:
Qualify=yes will send a SIP OPTIONS periodically and
keep the NAT open,
if you use 1 to 1 NAT (versus PAT where it is many
to one NAT) it will
work because port 5060 on the private address will
still be port 5060 on
the public address.
Tried that,
--- Tim Panton [EMAIL PROTECTED] wrote:
Yes. That is the way that IAX2 likes to work.
Ok.
However, not all providers will allow it, some
require a fixed IPaddress
and port for them to send calls to.
Is this the reason for the recommendation I've seen in
various forums to have port 4569
--- Josh McAllister [EMAIL PROTECTED] wrote:
Just a shot in the dark... but have you tried
Answer() before
Playback()?
http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+answer-before-playback
says New versions of Asterisk have added Answer
capabilities to several functions like
I have a PSTN termination provider foo which will
accept standard U.S. calls in the form 110 digit
ph#.
I have an outbound route named foo, with dial
pattern 5|., with the only entry in trunk sequence
being IAX2/foo.
I have an X-lite local extension, on which I can
dial
5110 digit ph#,
Bruce Reeves [EMAIL PROTECTED] wrote:
I use teliax.com and exgn.net to do my initial test
of toll free calls into
my system.
How's your experience been with their audio quality,
and with their inbound call completion reliability?
__
Do You
I have an asterisk system behind NAT, and need to
connect to public PSTN originators via SIP or IAX2,
but don't have the option of forwarding any ports
(4569, 5060, etc) to the asterisk system. However, the
NAT system does properly establish transient UDP
forwarding on the basis of outgoing
I have a PSTN termination provider foo which will
accept standard U.S. calls in the form 110 digit
ph#.
I have an outbound route named foo, with dial
pattern 5|., with the only entry in trunk sequence
being IAX2/foo.
I have an X-lite local extension, on which I can dial
5110 digit ph#, and
Calling from a local extension on my local
network, I get good voice quality from asterisk, and
asterisk reliably recognizes my dtmf input.
I set up a sipphone trunk (free) and called in to it
via a separate sipphone account on another computer,
and got slightly lower, but still good, audio
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