The duplicated messages was actually an accident, I did not believe the
first had arrived due to a 1 hour delay.
I have asked Digium the question, but had no reply on it. I do however
know there are members in this comunity that are working on mtp-2 that
might share some info, but I have a
On Tue, 15 Nov 2005, [EMAIL PROTECTED] wrote:
I would be interested to know if anyone have done any mtp-2 work on the
Digium boards.
chan_ss7 is reading mtp2 data from timeslot on a Digium board. At
this time, it has been tested on TE110P and TE410P.
See
Ahhhr,
Just realized that there even is a separate ss7 listing thanks.
Jan
Jacob Tinning wrote:
On Tue, 15 Nov 2005, [EMAIL PROTECTED] wrote:
I would be interested to know if anyone have done any mtp-2 work on the
Digium boards.
chan_ss7 is reading mtp2 data from timeslot on
Hi Jan,
I believe that MTP2 (Q.703) is availabe for Zaptel devices in source/library
form in the newly released GPL SS7 drivers by Sifira A/S http://www.sifira.dk/.
It is described as being mostly complete last time I checked.
I don’t know if its in a from that is usable for your purposes.
You
OS:Red Hat Linux 8.0 3.2-7
gcc version 3.2 20020903
asterisk-1.2.0 rc2
asterisk-ooh323c
when I reload in CLI, asterisk crash.
(gdb) bt
#0 0x4018ef89 in free () from /lib/libc.so.6
#1 0x406e84d2 in delete_peers () from /usr/lib/asterisk/modules/chan_ooh323.so
#2 0x406e5d64 in ooh323_do_reload
On 11/16/05, Abdul Lateef Khan [EMAIL PROTECTED] wrote:
Hi friends,
I want to change the standard 5060 sip port to our any defined port. i made
some change in sip.conf but it is not working, I have 2 softphone which are
able to register with 81 port but the any kind of hardphone is not able
In ast_channel (channel.h) structure there is ast_callerid variable,
but when I use that variable in code it gives error:
structure has no member named `cid'
How do I get caller id string ?
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On Wed, 16 Nov 2005, ast guy wrote:
In ast_channel (channel.h) structure there is ast_callerid variable,
but when I use that variable in code it gives error:
structure has no member named `cid'
How do I get caller id string ?
I think you can read the caller in chan-cid.cid_num and
On 11/16/05, Jacob Tinning [EMAIL PROTECTED] wrote:
On Wed, 16 Nov 2005, ast guy wrote:
In ast_channel (channel.h) structure there is ast_callerid variable,
but when I use that variable in code it gives error:
structure has no member named `cid'
How do I get caller id string ?
I
Dear All,
Sorry I am newbie in this forum but I wanted to check out if someday
we can see Asterisk perform CAC on contexts and limit the number of concurrent
calls a context can make to another context, I can see it performed on
iax and zap trunks but with my working scenario I really need to
Boot linux and use vmware when windows is needed. I know several laptop
owners that do this.
I do this on my desktop. It is the only way I found to run windows
without ever rebooting my machine :)
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Hi everybody.
I need develop a IAX softphone with Delphi, but i didnt find a OCX component.
Anyone know how can I
find this component ?
Tomas
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On 11/16/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi everybody.
I need develop a IAX softphone with Delphi, but i didnt find a OCX component.
Anyone know how can I
find this component ?
Tomas
I don't believe one exists as part of the standard distribution.
You're welcome to roll
Time Bandit wrote:
Boot linux and use vmware when windows is needed. I know several laptop
owners that do this.
I do this on my desktop. It is the only way I found to run windows
without ever rebooting my machine :)
Same here and I also find that vmware suspend of the windows virtual
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