I do not have the possibility to check does TDM800 works with asterisk 1.6.2.
I checked i have this code in chan_dahdi file.
But when I try to call, I get only
chan_dahdi.c: Using channel 11
devicestate.c: device 'DAHDI/11-1' state '2'
rtp.c: Channel 'DAHDI/11-1' has no RTP, not doing anything
ch
https://issues.asterisk.org/view.php?id=6643
CP
On Thu, Apr 1, 2010 at 7:36 AM, Ondrej Valousek wrote:
> Hello,
>
> Did anyone manage to force asterisk to put Remote-party-ID attribute on
> the SIP outgoing call? I.e. When A calls B, I want that A gets a name of
> B displayed on his phone.
> Not
Does TDM800 with FXO ports work with 1.6.2?
You should have also got other 'polarity related messages' during the call
setup.
One in particluar which prints debug info when a DAHDI_EVENT_POLARITY get
fired.
Code below.
ast_debug(1, "Polarity Reversal event occured - DEBUG 2: channel %d, state
%
Hi,
I'm running Asterisk 1.4.26.3 and I've noticed an interesting problem
when trying to play a Busy tone over a IAX trunk from the PSTN.
It seems as though Busy(20) returns non-zero immediately (it does not
wait 20s), so the caller never hears the busy tone, but
the call just appears to hang up.
I
Hello gang,
Is there a piece of software out there that can validate a
dialplan before I run it though my asterisk (1.4 and 1.6)? Right now I'm
just doing live run-time debugging, but that's slow and not always accurate
and my dialplan now exceeds 2000 lines. Any ideas?
Than
> Why aren't you using check_expr in the utils directory?
Aren't they two different things? I thought check_expr looks at a whole
file for syntax errors while testexpr2 just parses one expression and
returns its value. But if testexpr2 doesn't exist anymore, shouldn't
the documentation be update
On Tuesday 06 April 2010 10:56:56 Richard Kenner wrote:
> I'm trying to build it and run into all sorts of problems. First,
> "make testexpr2" doesn't work at top level, nor in the "main"
> subdirectory. If I manually try the commands for it in main/Makefile,
> it doesn't have a "main" and calls
On Tue, 6 Apr 2010, bob gailer wrote:
>> Verify the username, password, context, and extension all exist.
>>
> I do not understand or see username.
> I do not see context.
> I do not see or have passwords or know how to specify them.
> Extension exists. I can call the other way with no problem.
S
call not succsessful.
I use nokia gsm gw witch have polarity reverse i try on my old asterisk 1.4.17
with digium tdm800 with fxo ports card polarity reverse works fine. But then i
connect to asterisk 1.6.2 with sangoma a400 with fxo ports card polarity don't
work.
polarity reverse is 600 mil
On 4/6/2010 10:31 AM, Steve Edwards wrote:
> On Tue, 6 Apr 2010, bob gailer wrote:
>
>
>> I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
>> the other fails:
>>
>> -- Executing [...@macro-dialout-trunk:19] Dial("SIP/526-09eec7c8",
>> "IAX2/InterOffice/210,300,tr") i
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Steve Edwards
> Sent: Tuesday, April 06, 2010 9:25 AM
>
> On Tue, 6 Apr 2010, Deric Page wrote:
>
> > Is there a way to limit the number of simultaneous ou
Is the call successfull?
The 'Ignore polarity reversal on line seizure' may just be a warning.
What equipment, which Telco is the FXO card connected to?
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Beha
Did you tried the good old ram disk?
Flavio E. Goncalves
www.asteriskguide.com
-Mensagem original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Em nome de David Backeberg
Enviada em: Tuesday, April 06, 2010 12:50 PM
Para: Asterisk Users Mail
I'm trying to build it and run into all sorts of problems. First,
"make testexpr2" doesn't work at top level, nor in the "main"
subdirectory. If I manually try the commands for it in main/Makefile,
it doesn't have a "main" and calls "ast_log". If use -DSTANDALONE2
instead, those go away, but t
On Tue, Apr 6, 2010 at 12:36 AM, huu giang wrote:
>
> Dear List,
>
> Are there any way of configuring of Asterisk so it'll cache sound files in
> memory, and when Asterisk receive a call, instead of loading sound files from
> the disk, it will load from the memory and so Asterisk can process muc
On Tuesday 06 April 2010 03:16:45 Olivier wrote:
> In asterisk-addons-1.6.1.2's changelog, you can see that 6 changes were
> committed between versions 1.6.1.1 and 1.6.1.2.
> But if I'm not mistaken, you cannot read anything there about Asterisk to
> Asterisk-addons compatibility.
>
> What is the r
On Tuesday 06 April 2010 07:46:05 Jay Vocaire wrote:
> I have been working on getting Asterisk and Exchange 2010 UM working
> together, and so far I am pretty happy. The one thing not working right
> now is MWI.
>
> I searched a bit and found this:
> https://issues.asterisk.org/view.php?id=13028
>
Jay Vocaire wrote:
> I have been working on getting Asterisk and Exchange 2010 UM working
> together, and so far I am pretty happy. The one thing not working right now
> is MWI.
>
> I searched a bit and found this: https://issues.asterisk.org/view.php?id=13028
>
> Now, please pardon me for bei
Thank you for your interest in my question and quick response. I am
relatively new to Asterisk, so I have a few specific questions regarding
your suggestions.
Then I will post to the list with a more meaningful subject and results.
On 4/6/2010 10:31 AM, Steve Edwards wrote:
> On Tue, 6 Apr 2010
I have a special requirement that insist an Asterisk server, 1.6.1.x,
is used.? I will have 2 SIP trunks coming into the server and I will
have to send calls to these SIP trunks with a round robin distribution
pattern.? I was thinking of using a group count function, if call
>>>
On Tue, 06 Apr 2010 20:49:50 +1200, Alec Davis wrote:
>
>Seconded - I've clients with the Plantronics CS60 and CS70's with the HL10's
>on a combination of Snom and Grandstream GXP2000 desk phones. A bit extra
>desk spaghetti, but they think it's worth it...
>
>
>Seems like it's either 2 or 3 devic
On Tue, 6 Apr 2010, bob gailer wrote:
> I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
> the other fails:
>
> -- Executing [...@macro-dialout-trunk:19] Dial("SIP/526-09eec7c8",
> "IAX2/InterOffice/210,300,tr") in new stack
> -- Called InterOffice/210
> -- Hungu
On Tue, 6 Apr 2010, Deric Page wrote:
> Is there a way to limit the number of simultaneous outbound SIP calls
> made by Asterisk? We've tried using the 'Asterisk sip call-limit'
> parameter but that doesn't seem to be working and one of our engineers
> says that parameter has been depreciated.
>> Are there any way of configuring of Asterisk so it'll cache sound files
>> in memory, and when Asterisk receive a call, instead of loading sound
>> files from the disk
On Mon, 5 Apr 2010, Luki wrote:
> Not directly, but it's not really needed. A long as the machine has
> enough RAM, the fil
I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
the other fails:
-- Executing [...@macro-dialout-trunk:19] Dial("SIP/526-09eec7c8",
"IAX2/InterOffice/210,300,tr") in new stack
-- Called InterOffice/210
-- Hungup 'IAX2/InterOffice-7578'
== Everyone is bus
Hi,
I do use the first solution based on DIALSTATUS variable. (
http://www.voip-info.org/wiki/view/Superdial+macro)
since it's included to a separated context named [superdial-macro], I don't
have to repeat it over and over, so the fact that it's not a oneliner
doesn't bother me at all :)
On Tue
Here's one way - put your dial command into a macro that polls via a "core
show channels" and only dials when the count is below X. Even using a
"slow language" like PHP or PERL, an AGI call/return would not add as much
time to the dial process as PSTN delay does.
Example:
- exten =
Is there a way to limit the number of simultaneous outbound SIP calls
made by Asterisk? We've tried using the 'Asterisk sip call-limit'
parameter but that doesn't seem to be working and one of our engineers
says that parameter has been depreciated.
Thanks,
deric.p...@nisc.coop
--
___
I have been working on getting Asterisk and Exchange 2010 UM working together,
and so far I am pretty happy. The one thing not working right now is MWI.
I searched a bit and found this: https://issues.asterisk.org/view.php?id=13028
Now, please pardon me for being ignorant of all of this, but I
Hello list,
I need a hand to find the best dialplan failover solution when
using two SIP Trunks.
My reasons to do failover are:
a) one of the two providers could be unreachable
b) both providers may be UP but one of them could return a SIP error
messa
James Lamanna wrote:
> I'm seeing a lot of "Exceptionally long voice queue length" errors in
> my logs, and then I seem to have a problem
> where Asterisk will drop the registration for a significant number of
> phones (they go UNREACHABLE), but then they
> come back approximately a minute later.
>
Hi,
I have a problem with polarity reverse
this my dahdi config
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echoca
> I'll be there but I don't know exactely when 'cause I'll at Paris this
> week for my Microsoft course
If you're on Twitter, follow @voipusers if you want to keep in touch
or email me if you prefer.
/r
--
_
-- Bandwidth and Co
Randy R a écrit :
> Several regulars from the VUC will be there, some of us are arriving
> Tuesday night. Anyone else considering the trip? Post here or contact
> me off list so we can meet.
>
> /r
>
>
I'll be there but I don't know exactely when 'cause I'll at Paris this
week for my Microsoft
Seconded - I've clients with the Plantronics CS60 and CS70's with the HL10's
on a combination of Snom and Grandstream GXP2000 desk phones. A bit extra
desk spaghetti, but they think it's worth it...
Seems like it's either 2 or 3 devices to make this work.
The lifter is not required, as mostly op
Several regulars from the VUC will be there, some of us are arriving
Tuesday night. Anyone else considering the trip? Post here or contact
me off list so we can meet.
/r
--
_
-- Bandwidth and Colocation Provided by http://www.ap
On Mon, 5 Apr 2010, Warren Selby wrote:
> On Mon, Apr 5, 2010 at 9:37 PM, Alec Davis wrote:
>
>> I've been asked for recommendations for a small call centre, an ethernet
>> SIP deskphone with a wireless headset.
>>
>> Similar approach would be a mobile phone with bluetooth head set.
>>
>> Eithe
Hello,
In asterisk-addons-1.6.1.2's changelog, you can see that 6 changes were
committed between versions 1.6.1.1 and 1.6.1.2.
But if I'm not mistaken, you cannot read anything there about Asterisk to
Asterisk-addons compatibility.
What is the rule for Asterisk to Asterisk-addons compatibility ?
Wonderfull ;)
On Mon, Apr 5, 2010 at 7:58 PM, Jason Parker wrote:
> bruce bruce wrote:
> > Thanks for the update Jason,
> >
> > How do the upgrades work if v1.6.0 is already install and one wants to
> > upgrade to 1.6.2 (once it's available)?
> >
> > yum upgrade asterisk*
> >
> > ???
> >
> > Tha
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