Chad,
You are absolutely right on this one. I had setup the Queue time out for
agent set to 15 seconds and retry to 2 seconds. So, I think during those two
seconds Asterisk for some crazy reason hits another extension and then comes
back to the same extension to ring again. So, I have setup the ag
Hi,
Is there any way to see the status of PRI from the dialplan? I`d like to
know whether it was Up or not before I attempt to dial on it. Second best
would be to know if was down when I tried dialing (but I need to
differenciate between Down and Up-but-provider-was-congested).
Mike
I would second that.
If you don't set a dial string in your handset then it waits for N
seconds before submitting the call. Pressing # will force an immediate dial.
Mark
On 11/04/2010 07:19 PM, Cary Fitch wrote:
> Watch the console as you dial. Dial the number and “#”. The ring
> should be “i
What do you see in the asterisk console when this happens?
Thanks,
--Warren Selby, dCAP
On Nov 4, 2010, at 7:12 PM, Bruce B wrote:
> Hi Everyone,
>
> We have three different Queues set to "leastrecent" strategy and from time to
> time I hear someone complain that they receive short rings (par
Have you tried 'core show channels'?
Thanks,
--Warren Selby, dCAP
On Nov 4, 2010, at 7:30 PM, Michelle Dupuis wrote:
> Is the a CLI command that shows all channels in use at one time? (Whether
> IAX, SIP, SCCP, etc)?
>
> As well, when I "SIP SHOW CHANNELS" I see phones registering showing as
- Original Message -
> On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese
> wrote:
> > Has anyone seen a how-to on getting Asterisk to work with Google
> > Talk
> > and Google Voice?
> >
> I wrote one last week:
> http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/
>
> Also
You can use flash operator panel or you can right API to get active
call list.
--
Sent from my iPhone
On Nov 4, 2010, at 8:40 PM, Zeeshan Zakaria wrote:
How about 'show channels'.
As for filtering, you'll have to do it separately using a format like:
asterisk -rx 'show channels' | grep '
On Thu, 4 Nov 2010 20:12:54 -0400
Bruce B wrote:
> Hi Everyone,
>
> We have three different Queues set to "leastrecent" strategy and from
> time to time I hear someone complain that they receive short rings
> (partial ring cycle) and since it's not their turn even if they
> pickup the phone the
Hi,
Have you noticed a marked increase in CPU load when using MixMonitor?
I use PHPAgi and Asterisk 1.6.2.9-2.
Mickael.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a li
How about 'show channels'.
As for filtering, you'll have to do it separately using a format like:
asterisk -rx 'show channels' | grep ''
You can filter the output further using awk. But each filtering will take a
second or two based on what you are filtering.
Zeeshan A Zakaria
--
www.ilovetovo
Is the a CLI command that shows all channels in use at one time? (Whether IAX,
SIP, SCCP, etc)?
As well, when I "SIP SHOW CHANNELS" I see phones registering showing as
channels in use. Is there a way to filter this output?
Thanks!
MD
--
__
Hi Everyone,
We have three different Queues set to "leastrecent" strategy and from time
to time I hear someone complain that they receive short rings (partial ring
cycle) and since it's not their turn even if they pickup the phone the call
is not given to them since the Queue is actually hitting s
I have back ported the 'r' feature for app Dial from 1.8.0 to 1.6.2.X.
The link to the diff is below.
http://files.bluecrow.net/asterisk/backports/1.6.2/asterisk-1.6.2.4-app_
dial-play-indications.diff
I made the diff against 1.6.2.4 and later patched a 1.6.2.13 system.
All hunks passed.
Watch the console as you dial. Dial the number and "#". The ring should be
"instant". Or if not, look at the console and report what you see.
Cary Fitch
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jy
Sent: Thursda
I am new to asterisk and using it for a research project. Have set up an
server (version 1.6.2.6) and 2 SIP phones (Linksys spa901) which are
registering fine with the server. They are able to call one another,
however, the problem is it takes roughly 8-10 seconds for the called phone
to ring. I
On Thu, Nov 04, 2010 at 05:50:25PM -0400, Jeremy Kister wrote:
> On 11/4/2010 5:30 PM, Warren Selby wrote:
> > I've never really looked that closely at them, sorry. Are they causing some
> > kind of issue on your box, or are you just curious?
>
> just curious; i didnt think it was the expected be
On 11/4/2010 5:30 PM, Warren Selby wrote:
> I've never really looked that closely at them, sorry. Are they causing some
> kind of issue on your box, or are you just curious?
just curious; i didnt think it was the expected behavior and wanted to
fix it.
It actually appears that the child mpg123
Thanks Warren. That should do.
Regards,
Bruce
On Thu, Nov 4, 2010 at 2:54 PM, Warren Selby wrote:
> On Thu, Nov 4, 2010 at 12:56 PM, Bruce B wrote:
>
>> Hi Everyone,
>>
>> I am doing a queue show and I can't see any column that shows a queue
>> member priority. Is there any other command that
Hi
I been testing my clients installation for any problems with the farsouth
gateway and what I noticed is that the calls are being dropped at transfer.
After testing
I found that when the reception hits the transfer button on her sip phone, the
caller gets a dialtone and the reception goes o
On Thu, Nov 4, 2010 at 4:22 PM, Jeremy Kister
wrote:
> On 11/4/2010 5:07 PM, Warren Selby wrote:
> > It is because you're using quietmp3 as your mode.
>
> Can you explain what the processes are doing?
>
> killing them doesn't affect music on hold or any other mp3 playback.
>
> strace shows that th
On 11/4/2010 5:07 PM, Warren Selby wrote:
> It is because you're using quietmp3 as your mode.
Can you explain what the processes are doing?
killing them doesn't affect music on hold or any other mp3 playback.
strace shows that their behavior doesnt change during a call.
--
Jeremy Kister
htt
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Thursday, November 04, 2010 4:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] useless mpg123 processes hanging ar
On Thu, Nov 4, 2010 at 3:51 PM, Jeremy Kister
wrote:
> Running Asterisk 1.6.2.11 on debian 5.0.6 with mpg123 1.4.3
>
> when i start asterisk, i immediately see two mpg123 processes spawned
> which sit there forever. I can't imagine it's normal behavior, but if
> it is, please explain :)
>
> #
On Thu, Nov 04, 2010 at 04:51:46PM -0400, Jeremy Kister wrote:
> Running Asterisk 1.6.2.11 on debian 5.0.6 with mpg123 1.4.3
>
> when i start asterisk, i immediately see two mpg123 processes spawned
> which sit there forever. I can't imagine it's normal behavior, but if
> it is, please explain
Running Asterisk 1.6.2.11 on debian 5.0.6 with mpg123 1.4.3
when i start asterisk, i immediately see two mpg123 processes spawned
which sit there forever. I can't imagine it's normal behavior, but if
it is, please explain :)
# /etc/init.d/asterisk stop
stopping asterisk.
#[...]
# /etc/init.d/
On Thu, Nov 4, 2010 at 3:24 PM, pepesz wrote:
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
> nonce="5fcd5fa1 "
>
I'm surprised to see the extra whitespaces in the nonce value.
> What can be the problem?
>
If your working configuration worked with 1.6.2 but not 1.8, please
c
Am 04.11.2010 18:16, schrieb Glenn O Larsen:
>
> Hi Stefan,
>
> Yes, the 1.4-svn works a lot better... Do you have the bug # ? I tried
> to find it, but I couldn't locate it.
>
> I'm still able to make the Asterisk not respond (timeout for phones
> trying to call) when all clients are subscribin
Dear All,
Today I upgraded asterisk 1.6 to 1.8.
As the result of this when peers trying to register to asterisk the system
shows:
NOTICE[24698]: chan_sip.c:23417 handle_request_register: Registration from
'"50" ' failed for '
192.168.1.80:5062' - Wrong password
even though on 1.6 everything was O
On Thu, Nov 4, 2010 at 12:56 PM, Bruce B wrote:
> Hi Everyone,
>
> I am doing a queue show and I can't see any column that shows a queue
> member priority. Is there any other command that can show the member
> priority on the Asterisk 1.4x CLI?
>
> We are using this format of dialplan to login ag
Hi Everyone,
I am doing a queue show and I can't see any column that shows a queue member
priority. Is there any other command that can show the member priority on
the Asterisk 1.4x CLI?
We are using this format of dialplan to login agents:
exten => 123,Answer()
exten => 123,n,AddQueueMember(500
On Thu, Nov 4, 2010 at 2:50 PM, Stefan Schmidt wrote:
> Am 04.11.10 13:14, schrieb Glenn O Larsen:
>> What often happens, is that most of the peers is getting UNREACHABLE
>> or Lagged When I try to call during this time, I get a timeout...
>>
>> Any ideas on where to start debugging?
>>
>> I'm
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Beers
Sent: Wednesday, November 03, 2010 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Gotoif changed in 1.8
On Thu, 2010-11-04 at 12:07 -0400, Silver Thorne wrote:
> Hey Everyone;
>
> I inherited an Asterisk box where the dialplan is a real mess. ( I would
> actually be embarrassed to post some of the stuff!)
>
> So, here is what I need to do - and again, I am looking for fishing nets
> and places to
Le 26/10/2010 14:49, Shaun Ruffell a écrit :
> [...]
> First, Digium technical support would be more than happy I'm sure to
> help you trouble shoot this. That being said...
>
> First thing I would do is update to the current trunk of dahdi-linux.
> Revision 9397 [1]
> http://svn.asterisk.org/view/
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Silver Thorne
Sent: Thursday, November 04, 2010 11:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Multiple extensions -
Hey Everyone;
I inherited an Asterisk box where the dialplan is a real mess. ( I would
actually be embarrassed to post some of the stuff!)
So, here is what I need to do - and again, I am looking for fishing nets
and places to cast them - if I don't figure it out, I will never
freakin' learn!
On 11/04/2010 10:16 AM, Giampaolo TUCCI wrote:
> I'm sorry but doesn't work !
> I have used:
> -> options wctdm opermode=TBR21 fxshonormode=1 -> nothing
Did you try?
"modprobe wctdm fastringer=1 fxshonormode=0 opermode=TBR21"
--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis D
On 11/01/2010 10:58 AM, Gareth Blades wrote:
Those SRV records are wrong. You have to specify both servers with
different priorities against the same hostname. You have sip and sip2
defines with different SRV records so whichever one you configure on the
phone thats the only record it will seen.
I'm sorry but doesn't work !
I have used:
-> options wctdm opermode=TBR21 fxshonormode=1 -> nothing
-> I have modified in usr/src/zaptel/ in all the files the next
occurrence
{20,"RING_OSC",0x7EF0} -> {20,"RING_OSC",0x7E6C}
{21,"RING_X",0x0160} -> {21,"RING_X",0x023A}
I have recompiled, reinstalle
You can do policy routing on Linux systems too, using iptables' mark
functionality, combined with iproute2. Mark packets patching whatever
parameters you wish (i.e., in the mangle table in the prerouting chain, match
packets on UDP port 5060), and then use ip rule/ip route to route as you
des
Am 04.11.10 13:14, schrieb Glenn O Larsen:
> What often happens, is that most of the peers is getting UNREACHABLE
> or Lagged When I try to call during this time, I get a timeout...
>
> Any ideas on where to start debugging?
>
> I'm running on Asterisk 1.4, with realtime users, with cache and
On Nov 4, 2010, at 9:41 AM, C F wrote:
> You see the problem is that asterisk will send as many packets as its
> admin does on the list. There is no way to change that. I suggest you
> first change the amount of packets per second you send.
>
> On Thu, Nov 4, 2010 at 5:38 AM, ali anjum wrote:
>>
You see the problem is that asterisk will send as many packets as its
admin does on the list. There is no way to change that. I suggest you
first change the amount of packets per second you send.
On Thu, Nov 4, 2010 at 5:38 AM, ali anjum wrote:
> Hi,
>
> (I have install trixbox2.8 with asterisk 1
Hi all,
I have configured a Mediatrix 8 FXS with Asterisk . The extensions on
Mediatrix are able to do external calls and receive calls from softphone quite
normal . However, when it originate internal calls, the call hung up as soon
as we pick up the phone, don´t matter if other end is
On 3.11.2010 ?. 02:29 ?., Dan Journo wrote:
Hi,
I've got a client with two ADSL connections for redundancy.
Is it possible to set up asterisk to connect to one SIP provider using
both adsl connections and load balance between the two connections?
Or to use one connection as the main one, an
Dears Friends,
I currently have 16 Cisco SPA525g phones with a SPA500s (Attendant
Console) connected to each phone.
All of the 16 phones, have their Attendant Console configured the same
way, where they are subscribing to each of the 16 phones.
When I power on the switch, where all the phones are
Hi,
(I have install trixbox2.8 with asterisk 1.6)
I am using speex codec for my Inter asterisk communication
Question1: I want to configure speex on 2.15kbs and packetization of 60ms
seconds for that is have configured "codecs.conf" for desired result and also
placed a line in general section
Hi,
I want to know that I have created a IAX2 trunk between two trunk I am
observing a packet rate of 50packet/sec mean packetization time=20ms but I want
to know that how to change the packetization time I have placed "trunk freq=50"
in general section of IAX but can not see any differen
Hi,
I want to know that I have created a IAX2 trunk between two trunk I am
observing a packet rate of 50packet/sec mean packetization time=20ms but I want
to know that how to change the packetization time I have placed "trunk freq=50"
in general section of IAX but can not see any difference a
Respected Sir,
I want your help regarding an issue on asterisk. I hope my mail will not
disturb your daily routine. My issue is I am connecting two asterisk over
IIAX2/SIP trunk. I have successfully connected multiple server and every client
from one server to call any other server's clie
The adsl lines are with separate providers, so that won't work.
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