Try sip show channels from the CLI
- Original Message -
From: Scott Moseman [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, October 12, 2007 6:12 PM
Subject: [asterisk-users] Display channels and codecs
Is there an easy way to show all active channels AND the codecs
On Sat, Oct 13, 2007 at 05:45:26PM -0700, Dominic Son wrote:
Hi.
You mean to use the AGI funtion in the particular programming
language? yeah. i tried, same results.. : T
I guess that this is a permissions issue. Check what you get in the
standard error.
--
Tzafrir Cohen
I don't seem to be able to find the necessary hardware
specs for an Asterisk server. What I have in mind is a
dedicated server to serve 50 or so people. All users
will use SIP phones and there will be an ISDN gateway
for outgoing/incoming calls. Do you have any
suggestions about the server specs
Olivier wrote:
I was told yesterday (by Cantata guy) that T.38 demands a good level
of QoS.
That surprised me a lot as I thought the whole purpose of T.38 was to
avoid SIP and ToIP latency.
T.37 is the answer to reliability, but most people don't want to use it
for totally stupid reasons.
On Sun, 14 Oct 2007, YT Lim wrote:
I don't seem to be able to find the necessary hardware
specs for an Asterisk server.
Look more. There are 100's of pages on it. Start at
http://www.voip-info.org/wiki/
What I have in mind is a
dedicated server to serve 50 or so people. All users
will
Alan Lord wrote:
As I said, 1.4.12 builds fine. I'll do a bit more digging and if I find
a cause I'll report it upstream.
I started to investigate this problem, but it seems it was something to
with me rather than the build process...
Hi
Thanks for your effots,
Please
Hi Ray,
Am Dienstag, den 09.10.2007, 10:10 -0500 schrieb Ray Chen:
Hi Philipp,
Thank you for your response to my question. I am working on a
project which uses Asterisk as the voice engine. I need to get
the ingress and egress sip call id for a call
Ok, this is what worked:
EXEC System rm -rf /var/lib/asterisk/sounds/blah.gsm
the -rf eliminates the hassle.. a dream come true it worked !
On 10/13/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Oct 13, 2007 at 05:45:26PM -0700, Dominic Son wrote:
Hi.
You mean to use the AGI funtion
On Sun, Oct 14, 2007 at 05:43:27AM -0700, Dominic Son wrote:
Ok, this is what worked:
EXEC System rm -rf /var/lib/asterisk/sounds/blah.gsm
the -rf eliminates the hassle.. a dream come true it worked !
-r sure wasn't needed . -f then? But this is the default of rm. The
shell got in your way?
Hi All,
I've been trying to send a message to the list for the past 3 days, but
I neither get bounces nor the message appearing in the list, so someone
on IRC sugested I reply to an existing message.
My subject is related to this message, although slightly different.
Apologies if my actual
Actually, forget everything else.
Even when I simply pick up the handset and dial 6600, I get those errors
in console, so it's not related to paging or call files or anything
special, I guess..
Any ideas?
bu
[EMAIL PROTECTED] wrote:
Hi All,
I've been trying to send a message to the list
On Mo, 15 Okt 2007, [EMAIL PROTECTED] wrote:
I've been trying to send a message to the list for the past 3 days, but
I neither get bounces nor the message appearing in the list, so someone
on IRC sugested I reply to an existing message.
Same with me here!
--
Volker Sauer * Poststrasse
http://en.wikipedia.org/wiki/Network_File_System_(protocol)
On 10/12/07, Pepo [EMAIL PROTECTED] wrote:
Using two Asterisk connected between they, How do I can check the voicemail in
a remote system but working like *97?
I mean dont want ask the voicemail box, just the password and go to the
Tilghman Lesher wrote:
On Saturday 13 October 2007 18:08:49 Turbo Fredriksson wrote:
Quoting Philipp Kempgen [EMAIL PROTECTED]:
exten = s,1,Answer()
exten = s,n,Goto(s-${DIALSTATUS},1)
This still doesn't make sense because you did not Dial()
before jumping based on ${DIALSTATUS}.
Ok, make
The Management Interface has an Action: Status that sends back all active
channels. No codecs, but it's a start.
Girts
On 10/13/07, Dovid B [EMAIL PROTECTED] wrote:
Try sip show channels from the CLI
- Original Message -
From: Scott Moseman [EMAIL PROTECTED]
To:
Hello everybody,
Which one is a better choice
1. Gateway device with FXO - SIP ( example Addpac
http://www.addpac.com/addpac_eng2/addpac_product_view_detail.php?class_id=19item_id=59
)
2. Digium (Wildcard TDM400P)
3. Sangoma (A200 Analog FXO/FXS)
All i need is to put asterisk in
Volker Sauer wrote:
On Mo, 15 Okt 2007, [EMAIL PROTECTED] wrote:
I've been trying to send a message to the list for the past 3 days, but
I neither get bounces nor the message appearing in the list, so someone
on IRC sugested I reply to an existing message.
Same with me here!
Yep, me too.
Hi
I have a question if there was a major change in CDR?
Few days ago I have upgraded to 1.4.12.1 from 1.4.4 and something bizarre
happened. After the upgrade I have no call details in the cdr table when the
call did not go through because of for example: Unable to create the channel
of type Sip -
Hello all,
I'd like to thank everyone's input which I'll sumarize and comment on
bellow.
As in all complex solutions, there are no quick answers and no 100%
correct solutions. There are trade-offs to be made among
very different possiblities... Of course, the purpose of my original
Hello Andrew,
In order to help you, could you please provide your dialplan ?
BR
Mathieu
Andrew Nowrot a écrit :
Hi
I have a question if there was a major change in CDR?
Few days ago I have upgraded to 1.4.12.1 http://1.4.12.1 from 1.4.4
and something bizarre happened. After the upgrade
On 10/14/07, Andrew Nowrot [EMAIL PROTECTED] wrote:
Hi
I have a question if there was a major change in CDR?
Few days ago I have upgraded to 1.4.12.1 from 1.4.4 and something bizarre
happened. After the upgrade I have no call details in the cdr table when the
call did not go through because
Hi List;
Can someone advise me why in the below context, it
does not run the Background step? Once I dial 1000,
then it hangup and give congestion signal? If I
comment the ResponseTimeOut, then it run the
Background but it does not wait till caller enter the
digits, once the sound file finish,
bilal ghayyad wrote:
Hi List;
Can someone advise me why in the below context, it
You never told us what version you are running.
If it's version 1.2, make sure you have set priorityjumping=no in your
extensions.conf or use the waitexten application.
Doug
--
Ben Franklin quote:
Those
Does 'sip show peers' actually show the phone as registered?
PaulH
On Mon, 2007-10-15 at 02:05 +1000, [EMAIL PROTECTED] wrote:
Actually, forget everything else.
Even when I simply pick up the handset and dial 6600, I get those errors
in console, so it's not related to paging or call
We use dell 860 rackmount server - not too expensive, readily available
and can handle well over 50 phones.
PaulH
On Sun, 2007-10-14 at 16:58 +1000, YT Lim wrote:
I don't seem to be able to find the necessary hardware
specs for an Asterisk server. What I have in mind is a
dedicated server
On Sunday 14 October 2007 17:35:04 bilal ghayyad wrote:
Can someone advise me why in the below context, it
does not run the Background step? Once I dial 1000,
then it hangup and give congestion signal? If I
comment the ResponseTimeOut, then it run the
Background but it does not wait till
bilal ghayyad wrote:
Can someone advise me why in the below context, it
does not run the Background step?
[Test_Bilal]
include = KuwaitInternal
include = EgyptInternal
exten = 1000,1,Goto(s,1)
exten = s,1,Answer()
exten = s,2,ResponseTimeout(5)
exten =
Morning All,
Has anyone here successfully implemented skills based routing within queues?
The concept behind skills based routing is fairly straight forward, and I
know I could do it with multiple queues, agent penalties and a bit of AGI to
put the call into the right queue.
However doing this
nick,
I am actually playing with skills based routing right now...
how would you propose to send multiple calls requiring different skills
into a single queue and have agents w/o that particular skill in the
same queue?
daveC
Nick Brown wrote:
Morning All,
Has anyone here successfully
Hi
I just got an AA50 from Digium and the paging command reboots
asterisk when you use it. Digium says it is a requested feature and is
of low priority. Is there any other way to page 10 Grandstream gxp2000
phones with meetme or some other command than the page command.
Thanks in advance.
Kelly opal wrote:
I just got an AA50 from Digium and the paging command reboots
asterisk when you use it.
Can't help you with this, but do you mean it reboots/crashes
the machine? Or does it restart asterisk?
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied -
I’m not sure if this will work on the Grandstream phones but I use this for the
Linksys phones.
exten = ,1,SIPAddHeader(Call-Info:\;answer-after=0)
exten = ,n,Dial(SIP/201)
exten = ,n,HangUp
I would guess it would work with multiple phones, i.e., exten =
Hi,
I am curious. What version of asterisk is running on that AA50?
Regards,
Joseph
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kelly opal
Sent: Sunday, October 14, 2007 5:46 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AA50 Paging
Hi All,
We successfully installed MFC/R2, chan_unicall.so with asterisk ver 1.2.6.
asterisk is loading properly and we can see US show channels working fine.
We are using digium Te120P card.
Now we are trying to setup E1 link with Nortel DMS 100, which is resides at
one of telco provider in Sri
Hi
It restarts asterisk. The unit does not reboot.
Kelly
On Mon, 2007-10-15 at 04:22 +0200, Philipp Kempgen wrote:
Kelly opal wrote:
I just got an AA50 from Digium and the paging command reboots
asterisk when you use it.
Can't help you with this, but do you mean it
Hi
I tried that. Unfortunately it is the Dial command. The first phone
to answer wins and the rest are dropped from the channel.
Thanks
Kelly
On Mon, 2007-10-15 at 12:25 +1000, Klaverstyn, David C wrote:
I’m not sure if this will work on the Grandstream phones but I use
this for the
Hi
Digium support says it is built on the 1.4 platform.
Kelly
On Sun, 2007-10-14 at 22:28 -0400, Joseph Begumisa wrote:
Hi,
I am curious. What version of asterisk is running on that AA50?
Regards,
Joseph
From: [EMAIL PROTECTED]
[mailto:[EMAIL
About memory, I think 512MB will be more than enougth. And hard drive
requirements depends on the configuration of your voice boxes, but any
modern server will be OK, I don't think that you need more than
20GB...
On 10/14/07, Paul Hales [EMAIL PROTECTED] wrote:
We use dell 860 rackmount server
The model AP200 that you are giving as example is 2 port only... and
i'm not sure about the price...
I know that codec conversion is one of the most cpu-intensive task
that asterisk has to do, so, you can chose a Digium/Sangoma card with
a powerful server doing the work or you can also use a VoIP
20GB should be fine - unless you want to do a lot of recording.
PaulH
On Sun, 2007-10-14 at 21:07 -0600, Edgar Guadamuz wrote:
About memory, I think 512MB will be more than enougth. And hard drive
requirements depends on the configuration of your voice boxes, but any
modern server will be
Hi Guys,
I have noticed a weird behavior in 1.4.12. When using Authenticate or
DISA in the dial plan the channel immediately switches to gsm format (if
you request a password) or slim (if you run DISA without password). The
debug log says...
===
[Oct 14 21:23:00]
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