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 question,
>
>You did run ldconfig, right?
Yeppers. Right after edit
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'.
>>
>> ldd /usr/sbin/asterisk
>> linux-gate.so.1 => (0xe000)
>When we call in on the analog line, I can see the call begin in the cli, and
>after 15
>seconds I see the call switch over to my sip provider, and after about 30
>seconds I get
>the 3 raising tone signals and the call is hungup.
Sorry guys, been a long day staring at the tube:) Answer() followe
I have an issue with Dahdi trunk and Asterisk 1.6.0.1 where my analog line is
call
forwarded on no answer or busy to my sip provider.
When we call in on the analog line, I can see the call begin in the cli, and
after 15
seconds I see the call switch over to my sip provider, and after about 30
s
---cut-
-= Info about function 'PP_EACH_USER' =-
[Syntax]
PP_EACH_USER(|)
[Synopsis]
Generate a string for each phoneprov user
[Description]
Pass in a string, with phoneprov variables you want substituted in the format of
{VARNAME}, and you w
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'.
>
> ldd /usr/sbin/asterisk
> linux-gate.so.1 => (0xe000)
> libdl.so.2 => /lib/tls/i686/cmov/libdl.so.2 (0
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 question,
You did run ldconfig, right?
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purch
Thanks for your reply, Alex.
At 00:14 11/29/2008, Alex Balashov wrote:
>Paste 'ldd /usr/sbin/asterisk'.
ldd /usr/sbin/asterisk
linux-gate.so.1 => (0xe000)
libdl.so.2 => /lib/tls/i686/cmov/libdl.so.2 (0xb7f98000)
libpthread.so.0 => /lib/tls/i686/cmov/libpthread.s
[EMAIL PROTECTED] schrieb:
> I'm facing a problem that's occurring more often..
>
> my setup is as such :
>
> PSTN- Sipura 3102- Asterisk- siphones.
>
>
> whenever a call takes place both outbound as well as inbound, if there were
> a bit of silence, the channel gets closed.
> if there were a
Hi All,
I'm facing a problem that's occurring more often..
my setup is as such :
PSTN- Sipura 3102- Asterisk- siphones.
whenever a call takes place both outbound as well as inbound, if there were
a bit of silence, the channel gets closed.
if there were a bit of latency, the system detects it
Max Alex schrieb:
> Actully we are getting the anonymous callerid from the originated phone
> (blocked from phone) so we need to override the callerid and then pass to
> network.
> we need to send out caller id. That is why we tried to override it.
>
> But we are not able to override it.
I don't
phone can ring, i also can pick up it(I seems originate did not create a new
Zap channel,just used an exsiting channel?).
But the second problem produced, i received the Dialing, UP, Newexten events
before my phone ringing. It is supposed that i send an originate c
There are probably other OpenR2 users that can help you in asterisk-r2
mailing list (http://lists.digium.com/pipermail/asterisk-r2/)
I have not tried Trixbox, but the first release of OpenR2 will be next
week, probably that will help to have Trixbox a proper support.
Moy
On Sat, Nov 29, 2008 at
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
> > but the callerid is not overriding while i am sending call to th
Portech makes larger rack mounted modular multi-channel gateways as
well. Not sure about the ISDN interface, but certainly with T-1/E-1
PRI.
Michael
On Sat, 29 Nov 2008 14:57:02 +, Julian Lyndon-Smith wrote:
>Thanks Gordon,
>
>I have been playing with the Portech, but was wanting a "larger"
Thanks Gordon,
I have been playing with the Portech, but was wanting a "larger"
solution (20+ channels)
Julian.
Gordon Henderson wrote:
> On Sat, 29 Nov 2008, Julian Lyndon-Smith wrote:
>
>> I've been asked to purchase a gsm gateway for use with our asterisk
>> server (for our use, not resellin
I've been asked to purchase a gsm gateway for use with our asterisk
server (for our use, not reselling)
I have a spare ISDN port on the server, so I have use either a PRI or
VOIP gsm gateway.
What would people recommend ? Has anyone used the QuesCom 400 ?
I would also love to know a rough idea
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