Hi,
When typing "man dahdi_genconf", you can read :
This script generate configuration files for Dahdi hardware. Currently it
can generate three files: dahdi, chan_dahdi, users and chan_dahdi_full
where it say :
This script generate configuration files for Dahdi hardware. Currently it
can gene
Hi,
I'm runnung * on centos4 smp.
When system is busy, asterisk uses 99.9% cpu.
I want asterisk to use more 100% cpu to process more calls.
Is this possible?
Thanks.
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Hi,
Thanks for your reply.
Actually we are setting up the callerid in case of emergency calls when we
got the anonymous callerid from the caller.
But the calls are going with callerid anonymous and not set the callerid, i
want to know how can we sent some meaning ful information to the emergency
se
At 22:29 11/30/2008, Tzafrir Cohen wrote:
>On Sun, Nov 30, 2008 at 08:21:05PM -0600, Doug wrote:
>
>> I think that it might be important to add that these
>> .so files were compiled sometime back. They work
>> on our production system. I was not able to compile
>> the more recent versions o
On Sun, Nov 30, 2008 at 08:21:05PM -0600, Doug wrote:
> I think that it might be important to add that these
> .so files were compiled sometime back. They work
> on our production system. I was not able to compile
> the more recent versions of rxfax and txfax:
>
> /usr/lib/asterisk/modules# ls
Sorry this took so long to reply to
I'm using dialogic hardware. It's a single bri server card.
I sent the diag output to dialogic and they said that both calls were the same.
Yet, one always fails and the other always completes. Go figure.
Jay
-Original Message-
From: [EMAIL PR
At 14:30 11/30/2008, Tzafrir Cohen wrote:
>On Sun, Nov 30, 2008 at 12:02:44AM -0600, Doug wrote:
>> At 15:32 11/29/2008, Tzafrir Cohen wrote:
>> >On Sat, Nov 29, 2008 at 02:59:18PM -0600, Doug wrote:
>> >> Thanks for your reply, Alex.
>> >>
>> >> At 00:14 11/29/2008, Alex Balashov wrote:
2008/11/27 Olivier <[EMAIL PROTECTED]>
>
>
> 2008/11/27 Tzafrir Cohen <[EMAIL PROTECTED]>
>
>> On Thu, Nov 27, 2008 at 12:58:53PM +0100, Olivier wrote:
>> > output is:
>> >
>> > # strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI
>> Telephony'
>> > DAHDI Telephony
>> > DAHDI Telephony
I would also double-check the Sipura settings. On the Voice / PSTN-Line tab,
you have a section labeled "PSTN Disconnect Detection" (almost at the
bottom). Make sure that - at least for debugging this situation - the
"Detect PSTN Long Silence" and "Detect VoIP Long Silence" are set to "no".
Otherwi
Yesterday I pulled in the latest svn of Dahdi and added the files
from a recent kernel in the drivers/staging/echo structure and modified
the Kbuild file so it would compile without error. I insmod'ed the module
in, and modified my system.conf has echocanceller=oslec.
cat /proc/dahdi/1 shows:
Span
On Sun, Nov 30, 2008 at 12:02:44AM -0600, Doug wrote:
> At 15:32 11/29/2008, Tzafrir Cohen wrote:
> >On Sat, Nov 29, 2008 at 02:59:18PM -0600, Doug wrote:
> >> Thanks for your reply, Alex.
> >>
> >> At 00:14 11/29/2008, Alex Balashov wrote:
> >> >Paste 'ldd /usr/sbin/asterisk'.
> >>
> >>
[EMAIL PROTECTED] schrieb:
> wht I mean with "a bit" 1 minute. almost always the same.. it varies
> between 1 minute and 1 minute and 6 seconds.
> Nov 30 21:57:10] WARNING[23213]: chan_sip.c:1946 retrans_pkt: Maximum
> retries exceeded on transmission
> NzQxZGExNjZlOWQyYzhhOTdmZWY4ZmI1M2U1OTd
Hi Philipp,
wht I mean with "a bit" 1 minute. almost always the same.. it varies
between 1 minute and 1 minute and 6 seconds.
this is the output on CLI:
Nov 30 21:57:10] WARNING[23213]: chan_sip.c:1946 retrans_pkt: Maximum
retries exceeded on transmission
NzQxZGExNjZlOWQyYzhhOTdmZWY4ZmI1M2U
Grey Man schrieb:
> On Wed, Nov 26, 2008 at 11:47 AM, Richard Brady <[EMAIL PROTECTED]> wrote:
>> Hi folks
>>
>> I'm not sure what I am missing but I cannot find a predefined channel
>> variable to identify the SIP peer/user which has initiated a call and
>> established the channel.
>>
>> The one o
At 02:10 11/30/2008, Tzafrir Cohen wrote:
>On Sun, Nov 30, 2008 at 12:08:19AM -0600, Doug wrote:
>> At 15:13 11/29/2008, Doug Lytle wrote:
>> >Doug wrote:
>> >> Thanks for your reply, Alex.
>> >>
>> >> Do I need a symlink in "/usr/sbin/asterisk" to point
>> >> to "/usr/local/lib/libspan
At 06:44 11/30/2008, Doug Lytle wrote:
>Tzafrir Cohen wrote:
>>> >I'm going to ask a stupid question,
>>> >
>>> >You did run ldconfig, right?
>>>
>>> Yeppers. Right after editing /etc/ld.so.conf
>>>
>>>
>
>
>You never mentioned your distro.
Sorry about that!
# cat /etc/issue; unam
thanks
Found that but sometimes I need to detect dtmf ie when playing back a
recording
Robb
Philipp Kempgen wrote:
> Robert Boardman schrieb:
>
>
>> I cannot seem to find a way to stop atserisk inercepting DTMF tones and
>> regenerating them even on a zap to zap bridged call
>>
>> is this p
Robert Boardman schrieb:
> I cannot seem to find a way to stop atserisk inercepting DTMF tones and
> regenerating them even on a zap to zap bridged call
>
> is this possible?
One (ugly!) solution is to change the DTMF tone frequencies in
Asterisk so it doesn't recognize them any more:
http://a
zap channel on one card to zap channel on another
Robb
Alex Balashov wrote:
> You mean a zap-to-zap call hairpinned into the same adaptor, or another one?
>
> Robert Boardman wrote:
>
>
>> Hi All
>>
>> I cannot seem to find a way to stop atserisk inercepting DTMF tones and
>> regenerating
You mean a zap-to-zap call hairpinned into the same adaptor, or another one?
Robert Boardman wrote:
> Hi All
>
> I cannot seem to find a way to stop atserisk inercepting DTMF tones and
> regenerating them even on a zap to zap bridged call
>
> is this possible?
>
> Thanks
>
> Robb
>
> __
Hi All
I cannot seem to find a way to stop atserisk inercepting DTMF tones and
regenerating them even on a zap to zap bridged call
is this possible?
Thanks
Robb
___
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asterisk-
On Sat, 2008-11-29 at 11:26 -0600, Tilghman Lesher wrote:
> On Friday 28 November 2008 08:17:24 Philipp Kempgen wrote:
> > Max Alex schrieb:
> > > I have one issue regarding override callerid when i have anonymous call.
> > > I have added PAI in sip header and also set sendrpid = yes in sip.conf
>
Tzafrir Cohen wrote:
>> >I'm going to ask a stupid question,
>> >
>> >You did run ldconfig, right?
>>
>> Yeppers. Right after editing /etc/ld.so.conf
>>
>>
You never mentioned your distro.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little T
On Sun, Nov 30, 2008 at 12:08:19AM -0600, Doug wrote:
> At 15:13 11/29/2008, Doug Lytle wrote:
> >Doug wrote:
> >> Thanks for your reply, Alex.
> >>
> >> Do I need a symlink in "/usr/sbin/asterisk" to point
> >> to "/usr/local/lib/libspandsp.so.1.0.0" ?
> >>
> >
> >I'm going to ask a stupid
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