On 2/11/2013 11:13 PM, Jeremy Kister wrote:
> [Feb 11 16:28:45] ERROR[28501][C-0012]: utils.c:1187
> ast_carefulwrite: write() returned error: Connection refused
[...]
can someone replicate this behavior ? Or is this just my config ?
opening issue in jira; this is a bug.
https://issues.as
I recently upgraded to asterisk 11 from 1.8.
I had VXML working via AGI in 1.8 - from extensions.conf:
[VXML]
exten => s,1,Answer
exten => s,n,Set(ENCODED=${URIENCODE(${ARG1})})
exten => s,n,AGI(agi://localhost/url=${ENCODED})
exten => s,n,Hangup
Using asterisk 11 on the same host with the same
I'd like to have an extension "join" a call. That is, C can join A and
B, just as if it were an analog extension phone.
ChanSpy works, sort of. The problem is that once A or B hangs up, the
channel is gone. With an analog extension, C would remain connected with
B if A hung up.
Can I throw A
On 11/02/2013 17.01, Jean-Denis Girard wrote:
So the solution would be that the network does not send the progress
indicator in the Disconnect message, or find a configuration parameter
on the gateway so that it ignores the progress indicator, right?
I believe the first one will be not a viable
Press "#" after entering the number to see if it's an extension pattern
matching delay
--Don
-Original Message-
On Behalf Of Steve Edwards
Sent: Monday, February 11, 2013 4:16 PM
Extension pattern matching (waiting for the digit timeout) can also induce
perceived dialing delays.
-
On Mon, 11 Feb 2013, Kevin Wright wrote:
I *am* now stuck with a long pause before I hear the outgoing ringing
though. Not sure if there's anything I can do to tackle that one.
Extension pattern matching (waiting for the digit timeout) can also induce
perceived dialing delays.
If you crank
I think the default DTMF tone duration is 100ms, if you are dialing 10 digits,
that ends up being 1 second delay just to dial the DTMF, not including
inter-digit delays. Try setting toneduration=50 in chan_dahdi.conf and see
what happens. If you make it too low your telco will miss some digi
I finally got an answer on IRC, turns out the problem was callprogress=yes
(a.k.a. screwupmycalls=yes). Changing it to no and dropping the r seemed
to work. I'd like the feature to work properly, but it's more important
that I'm able to actually make calls :)
I *am* now stuck with a long pause b
Hey,
Just quickly glanced over your data... one problem you have is that
you're passing the 'r' flag in your Dial() statement in extensions.conf.
That would definitely cause you to have never ending ringback from the
analog line (since answer supervision is often not present). You might
try
I forgot to add, cat /proc/dahdi/* yields:
Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER)
1 WCTDM/4/0 FXSKS (In use) (EC: MG2 - INACTIVE)
2 WCTDM/4/1 FXOKS (EC: MG2 - INACTIVE)
3 WCTDM/4/2 Reserved
4 WCTDM/4/3 Reserved
I'm not sure if that (in use) is correct when I'm n
On Feb 11, 2013, at 4:31 PM, A J Stiles wrote:
> On Monday 11 February 2013, Shitian Long wrote:
>> Hello,
>>
>> I am trying to connect two asterisks with PRI connection. One asterisk has
>> TE405P Quad PRI ports card, anther asterisk has TE110P 1 PRI port card.
>>
>> I am wondering if there w
Thanks for your message.
On Feb 11, 2013, at 2:34 PM, "Yves A." wrote:
> Hi,
>
> I think, you mean connecting the two boxes directly with a cable... not via
> PSTN, right?
>
> 1.) You need a special cross-over cable to connect one Port directly to
> another Port...
> (if you want to crimp i
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Le 11/02/2013 03:44, giovanni.v a ￩crit :
> I think the gateway is working in compliance to ETS 300 102-1 (5.3.4.1
> Clearing when tones/announcements provided). 30 Sec. is the time
> assigned to T306 on the network side.
Ok, thanks for your analysis.
Hi,
I used Asterisk 1.8 and I have a gsm modem with 8 port.
When I called target number, gsm modem and asterisk show me one of these
ports active. after hangup, the actived port is going to idl status and
ready to use. but after some call from extension, when I want to call
another number, asteris
On Monday 11 February 2013, Shitian Long wrote:
> Hello,
>
> I am trying to connect two asterisks with PRI connection. One asterisk has
> TE405P Quad PRI ports card, anther asterisk has TE110P 1 PRI port card.
>
> I am wondering if there would be some step by step guide that I could
> follow to t
I am really a beginner of PRI ISDN board, I am wondering if there is a "quick
start" chan_dahdi.conf configuration I could use.
I tried to install two "FreePBX" boxes follow the instructions from
"http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html"; connected them
between PRIs, It wo
On 02/11/2013 01:20 AM, Dmitry Melekhov wrote:
> Hello!
>
> We use meetme, but, as I understand it will be soon removed from
> asterisk (already marked as deprecated), so I'm thinking about
> confbridge migration.
> Really, we use self-developed (really my ;-) ) web interface to control
> meetme.
On 11/02/2013 5.18, Jean-Denis Girard wrote:
The
Mediatrix sends incoming calls from the PSTN to an Asterisk server via
SIP: this works fine. But when the caller hangs up, the Mediatrix
doesn't send "Bye" to Asterisk, so the call is not finished immediately
from the Asterisk point of view: the de
Hi,
I think, you mean connecting the two boxes directly with a cable... not
via PSTN, right?
1.) You need a special cross-over cable to connect one Port directly to
another Port...
(if you want to crimp it yourself, you can find the associated Pins via
Google... ethernet crossover
cables do
Hi
I have an installation based on Daniel-Constantin Mierla's excellent
Kamailio 3.3 / Asterisk 10 Realtime document (
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb)
but have come across an issue which is a potential problem.
In this installation all SIP clients regi
Hello,
I am trying to connect two asterisks with PRI connection. One asterisk has
TE405P Quad PRI ports card, anther asterisk has TE110P 1 PRI port card.
I am wondering if there would be some step by step guide that I could
follow to to this kind of connection?
Thanks
--
from longst
--
_
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