Displaying 12 results from an estimated 12 matches for "weerasooriya".
2004 Oct 08
0
re:uniqueid - how unique it is (Sathya Weerasooriya)
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2004 Aug 08
3
iconnect inbound - so do we know how to fix it
Just wondering whether we have a resolution to iconnect incoming problem,
which started few days ago.
Cheers
SW
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2004 Aug 05
2
new bounty for modifying calling card application to mysql
Hi,
I've just initiated a new bounty for the above;
http://www.voip-info.org/wiki-Asterisk+bounty+callingcard+to+MySQL
Any takers or any contributors please respond to me privately. I do not know
exactly how the bounty process works, but I can coordinate on this ?
SW
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2004 Aug 10
0
iconnect inbound - FIXED (kinda)
...tbound context
> otherwise, chan_sip will authenticate against the outbound
> peer instead of the inbound peer.
> 5) NOTE that the syntax of the registration statement has
> changed slightly as well (see wiki) and may need to be modified.
>
> On Aug 9, 2004, at 8:57 PM, Sathya Weerasooriya wrote:
>
> > Raj, yes your post helped me.
> >
> > Just to complete the whole thing and clarify the problem that was
> >
> > posted by Greg Blakely;
> >
> >> First, if there is no outbound iconnect section in sip.conf, my
> >> incoming
>...
2003 Nov 02
2
Read error on sound device
Hello,
I am posting this after spending hours digging through the list archives.
Problem : When asteirsk plays a voice prompt, the voice clip is really
choppy.
I figure that this is something to with the sound card, the timing of
playback etc.
But cannot seems to find an answer.
Here is the Notice which appear when voice prompt is played.
NOTICE[1217602880]: File sched.c, Line 209
2003 Nov 19
2
g723 to g723 SIP call - warning message
Hi,
I am calling from a grandstream phone with g723 codec through * to iconnect.
Incoming context as well as outgoing context set to g723.1 codec in *.
Call get connected and I can talk. However I get the following warning,
which scrolls on my screen until I hang-up.
[root@asterisk sath]# cat g723.1
- Executing SetCallerID("SIP/-08122ae0", "1001") in new stack
--
2003 Nov 12
0
sending MWI to a none local client
Hi,
I am using * to function as the voice mail system for Vocal. Since I do not
have a context in sip.conf file for each vocal client, I can't set the
mailbox=xxxx in sip.conf. How do I get the MWI to a Vocal client ?
Cheers
Sathya
2003 Nov 19
0
Getting in to h323
Greetings,
I am progressing well with this great product, the *. SIP to SIP calling,
Vocal to *, Voicemail all in the past. Did Iconnect, FWD etc. Also,
purchased couple of FXO cards and did zaptel as well. It's time to get to
h323 now. Read the mailing list for H323 and OH323 etc. need some help to
where to start.
Requirement is very simple, SIP calls need to be routed to a third party
2003 Nov 20
0
codec pass-through feature
Hi Gurus,
I we seen references to 'codec pass through feature' in the mailing list.
SIP to SIP and SIP to chan_h323 as well. Could someone help me to understand
this feature, or point me to some examples etc.
Appreciate any pointers here.
Thanks a bunch
Sathya
2004 Aug 07
2
astcc help
Hi,
Really appreciate if someone who got astcc working lists the steps to make
it work. I've got it installed and using the gui could get the database
created. Would like to know how those two .conf files be populated and some
pointers to the important fields in the database.
Thanks
Sathya
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2004 Oct 07
0
uniqueid - how unique it is
Hello,
a.. uniqueid: Unique Channel Identifier (32 characters)
I would like to know how is the unequeId constructed. ?
I need a primary key in cdr database and would like to know whether I can
make uniqueid the primary key. If asterisk channel does not have any idea
of previousely crated ID's and if this id is randomly crated number then
there is a chance that unequeid get a duplicate
2005 Mar 10
0
SIP to H.323 no audio
Hi,
I am trying to make a call from SIP to H.323 using chan_h323. Asterisk
CVS-HEAD-03/10/05-10:08:22. As given in chan_h323 readme I compiled pwlib
and open h323 versions 1.8.1 and 1.15.1. Call seems to be get connected but
no audio path.
I see following;
-- AGI Script Executing Application: (DIAL) Options:
(H323/YYYY#XX112422428@XX.103.19.91/XX112422428|60|HS(63840))
-- Setting call