Steven Critchfield wrote:
On Tue, 2004-09-14 at 11:30, Maciek Kaminski wrote:
Apparently there are subtle diferences between meaning of MeetmeJoin
event depending on channel type.
Problem is: after originating a call from channel to MeetMe room i.e.:
[meetme]
exten = 1,1,Answer
exten = 1,2
Apparently there are subtle diferences between meaning of MeetmeJoin
event depending on channel type.
Problem is: after originating a call from channel to MeetMe room i.e.:
[meetme]
exten = 1,1,Answer
exten = 1,2,Meetme(kolejka|dqM)
than:
Context: meetme
Exten: 1
Priority: 1
ActionID:
I am sending this again since I haven't get it back for twelve hours:
When I originate call to IAXComm, more or less one of tree calls fails
for no aparent reason. Originating calls to SIP clients works as
expected. Anybody has similar problems? Is it asterisk or client problem?
Asterisk log:
Jul
it introduce to much latency? Or should I insist on buying ISDN
interface for asterisk box? What hardware would You recommend for this
setup?
Another question: anyone has successfully deployed call center solution
using soft phones? If yes, with which soft phones?
Maciek Kaminski
I've got following problem with manager api:
In my Asterisk installation when I connect two channels (IAX,SIP) I get
following sequence of events(these are events for *single* connection,
come one by one without any delay):
Event: Link
Channel1: [EMAIL PROTECTED]:5036]/3
Channel2: SIP/kamyk-9950
May problem is: I need to know when and between which channels
connection is setup and hungup. Is there a way to learn this from
manager interface? There are Link/Unlink events but then appear more
than once during single connection, ie while calling from IAX to SIP I get:
Event: Link
Channel1:
originate should be made
asynchronous and make identyfing channel name via events possible.
P.S.: There is a patch in mantis
(http://bugs.digium.com/bug_view_page.php?bug_id=772) that makes
originate asynchronous but it has not been approved yet.
Maciek Kaminski
On which channel types manager redirect work? I try it on IAX chanel and
it fails with message Redirect failed. Which channel syntax should by
used for redirect command? There seems to be e.g.:
[EMAIL PROTECTED]:5036]/27 for channel instance and e.g.:
IAX/[EMAIL PROTECTED] for channel type.
Wilson Pickett wrote:
Hello list,
I am drowning in Asterisk How To sites. Maybe someone here can help me.
I ordered a devel kit and while I'm waiting for it I installed Slackware 9.1 on a Pentium III 500mhz box with 192 Megs RAM.
I downloaded and installed Asterix as per the CVS instructions
:
-
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3;rport
CSeq: 1974 INVITE
To: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
From: Maciek Kaminski sip:[EMAIL PROTECTED];tag=B62B188
Call-ID: [EMAIL PROTECTED]
Subject: sip:[EMAIL PROTECTED]
Content
it? But that is the way that I have seen almost all UAs work. The
implication is that your offer must be a valid, not a conditional offer :
when you say you accept GSM on port 8000, you better have a listener on 800
ready to go.
Optimistic strategy...
Maciek Kaminski
Regovich, Timothy wrote:
Where is that quote from?
RFC - 3264 An Offer/Answer Model with the Session Description Protocol
(SDP) chapter 6.
Maciek Kaminski
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Steve wrote:
Hi,
If anyone else had a problem I got kphone to work with Asterisk.
I have problems with kphone + Asterisk. KPhone does not seem to ACK
invites, ie.
KPhone --- sends INVITE -- Asterisk
KPhone -- sends 101 Trying --- Asterisk
KPhone -- sends 202 OK --- Asterisk
Hi,
What linux SIP UAs do You successfully use with Asterisk?
Maciej Kaminski
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