Hi All:
I have a x100p card and asterisk installed to my computer, can I use this
card and a softphone to call other phone?
Thanks
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Refer to: http://www.microsuncn.com
Best Regards
Alan Zheng
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Hi
My asterisk output is:
chan_sip.so = (Session Initiation Protocol (SIP))
Asterisk Ready.
-- Registered SIP '201' at 192.168.0.55 port 33906
-- Saved useragent X-Lite release 1011s stamp 41150 for peer 201
-- Executing [907768385...@default:1] Dial(SIP/201-083e75c0,
2009/10/4 Alan Zheng machinecat1...@gmail.com
Hi All:
I have a x100p card and asterisk installed to my computer, can I use this
card and a softphone to call other phone?
yes
Thanks
--
Refer to: http://www.microsuncn.com
Best Regards
Alan Zheng
Wanted to update everyone, that IP 64.34.173.199 belong to a company
Voxalot, they have hacked our system twice and they don't even care to reply
to any emails sent to them, and they don't even respond over the phone,
beware of them.
Regards
Vijay Gandhi
GIPL(An ISO 9001:2000 Company)
Hello, you can change the O.S for Debian etch 4.0? Is more better and
more easy to install the library and asterisk dependences.
Please let me know
Regards
Josue
2009/10/4 Angus Asterisk aster...@iteloffice.com:
Hi
My asterisk output is:
chan_sip.so = (Session Initiation Protocol (SIP))
That's interesting. Do you have experience working with both SUSE and
Debian? Why is Debian easier?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Josué Conti
Sent: 04 October 2009 15:58
To: Asterisk Users
Looking at the function below in chan_alsa.c shouldnt't the variable
hookstate be set to 1 for auto answer??? I think this is affecting the
ringing I hear after the call.
Jerry
static int alsa_call(struct ast_channel *c, char *dest, int timeout)
{
int res = 3;
struct ast_frame f
Hello, I had problems with SUSE to Debian that they do not appear, and
for the compilation of packages with Debian has always been all that
easy, try Debian Etch 4.0, you'll like it.
Best Regards
Josue
2009/10/4 Angus Asterisk aster...@iteloffice.com:
That's interesting. Do you have
Josué Conti schrieb:
I had problems with SUSE to Debian that they do not appear, and
for the compilation of packages with Debian has always been all that
easy, try Debian Etch 4.0, you'll like it.
Do you recommend Debian 4 (Etch, oldstable) for a particular reason?
Should you have any problems
On Sun, Oct 04, 2009 at 11:28:23AM +0100, Angus Asterisk wrote:
Hi
My asterisk output is:
chan_sip.so = (Session Initiation Protocol (SIP))
Asterisk Ready.
-- Registered SIP '201' at 192.168.0.55 port 33906
-- Saved useragent X-Lite release 1011s stamp 41150 for peer 201
Just detecting this tread...
Moving to Debian is quite a big step.
How about updating to openSUSE_11.1 and use the prebuild asterisk
packages (either zaptel or dahdi) .
On the OBS they are available for 1.4.x, 1.6.0, 1.6.1
hw
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Hi
I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls
to that trunk, I am getting all circuits are busy now, do I have to do
something specific?? I am using elastix.
Thanks.
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Is inbound working?
Can you see action on the CLI when you send a call to the lines attached
to the card?
PaulH
B.Masoud @ SH wrote:
Hi
I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls
to that trunk, I am getting all circuits are busy now, do I have to do
I'm using QSIG between Asterisk and an NEC SV8300. Whenever I make a call
from the SV8300, I see:
[Oct 4 21:02:49] ERROR[5729]: chan_dahdi.c:12226 dahdi_pri_error: !! Unknown
IE 50 (cs5, len = 3)
I see an IE 50 in the Q.932 specification, so I don't understand why
this error is occuring.
I found the problem.
The function send_sound() in chan_alsa.c does this:
if (FD_ISSET(sndcmd[0], rfds)) {
if (read(sndcmd[0], cursound, sizeof(cursound))
0) {
ast_log(LOG_WARNING, read() failed:
%s\n, strerror(errno));
On 3/10/09 3:55 AM, das sandesh wrote:
I am using the command:
./sipp -sn uac -d 200 -s repective context pattern IP Address -l 200
Its 10 calls per second and 200 concurrent calls, similarly I used 2 ssh
sessions each sending 100 concurrent calls. But this was limiting to
only 150
On Sunday 04 October 2009 20:05:09 Richard Kenner wrote:
I'm using QSIG between Asterisk and an NEC SV8300. Whenever I make a call
from the SV8300, I see:
[Oct 4 21:02:49] ERROR[5729]: chan_dahdi.c:12226 dahdi_pri_error: !!
Unknown IE 50 (cs5, len = 3)
I see an IE 50 in the Q.932
Whenever I try to drag calls to the Parking Lot or On Hold, FOP would
drop my calls. I have searched online and have found
similar problem, such as the link below. I have tried their solution
but still the FOP is not working correctly. I even installed the
HUDLite server and is getting the same
The two patches attached on the issue apply against libpri branch
1.4 and asterisk trunk, respectively. Both are required. Given
that it's been 5 months since I first created the patches, I have
redone them tonight, in order to facilitate testing.
Thanks!
I checked out that branch with
On 5/10/09 3:06 PM, Jerry Geis wrote:
I found the problem.
The function send_sound() in chan_alsa.c does this:
Best bet is to open an issue on the bugtracker and post your patch
--
Cheers,
Matt Riddell
Director
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At 04:32 PM 10/4/2009, you wrote:
Hi
I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls
to that trunk, I am getting all circuits are busy now, do I have to do
something specific?? I am using elastix.
Sometimes you can't make a call on DAHDI until a call has been
On Mon, Oct 05, 2009 at 10:37:52AM +1100, Paul Hales wrote:
Is inbound working?
Considering he has no channel of type 'ZAP' and he uses Zaptel: no.
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