Extracted from http://www.voip-info.org/wiki-Asterisk+cmd+Dial:
' When options /t/, /T", "h", "H", "w", "W" or "L" (with multiple
arguments) are applied, Asterisk will remain in the media path, even if
/canreinvite=yes'' (a SIP channel option) has been specified.'
Then how is it possible to limit a call without the L option ?
Benoît Mérouze wrote:
Hi,
I've got some problems with bridged calls, the quality is extremely
poor (more or less blanks or one way voice issues). But if I do a
normal call with the same provider, there is no problem.
Reinvite is enabled, but what are the parameters in the dial command
that force asterisk to stay in the loop ?
Are the H (to allow caller to hang up by dialing *) or L (to limit the
call) parameters ones of them ?
As an example, here is a Dial command I execute to bridge a call to a
new one :
SIP/kddi/0033172699611|30|HL(1620000:60000:30000)
Thanks,
Benoit
[EMAIL PROTECTED] wrote:
Hi,
Check if reinvites are enabled, and that you don’t use any parameter
in the dial command that forces asterisk to stay in the loop.
Ohad
------------------------------------------------------------------------
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Erick
Baum
*Sent:* Wednesday, June 14, 2006 5:00 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [Asterisk-Users] voip to voip bridge
Has anyone had any good experiences with a voip to voip bridge...
where you have an incoming call on a voip line which is redirected
out another voip line to a regular phone line? Whenever we do this,
the connected call is kinda lagged and the quality isn't always that
great. It seems to me this is just a problem with the inherent delay
in the voip connections. But I was wondering if there's any special
configurations that could make the situation better?
Erick
--
Benoît Mérouze
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