> Set your core debug level to greater than 2
SET DEBUG seems not to have any effect on my asterisk.
> Let us know what you find.
The effect was caused by an "misconfigured" phone: The phone did nod
signal "busy" but "ringing" due to an "call waiting indication".
Switching off call wating indic
on sip show channels I do get a lot of entrys like
192.168.1.47 11 07ba5a490b3 00102/0 unkn No
Init: INVITE
192.168.1.47 11 19090f115b8 00102/0 unkn No
Init: INVITE
192.168.1.47 11 7d8b8fde46f 00102/0 unkn No
Init: INVITE
How do they ap
Hi,
On asterisk 1.4.4 I have an strange effect on agents answering queue calls:
If an agent does set current call on hold the phone
immediately gets connected to the next incoming call.
What might cause this effect?
How can it be removed?
Elmar
___
Isaac Xiao schrieb:
Would any one advice how implement Diva Server BRI or PRI card to
support fax and data modem?
You can use any CAIP-fax software to send and recieve Faxes with an
Diva-server card.
The chan-capi driever has such functionality, using "capisuite" would be
another option.
Let's say an office has 20 people with 20 extensions and they want to
enter a code on their phone when they leave for lunch and a voice will
tel lthe caller like:
The person you are calling is out of the office and will return at 1 pm.
Is this something that is possible?
I'm tot shure if the
Please advice me how i can make it work?
It looks like your Phone is not compatible to G.723.1 or this codec is
disabled within sip.conf
Elmar
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or
On Upgrading do 1.2 I do have an PRoblem with my VOIP-Provider.
Making outbound calls does result in "Error 400" - exept if I do call my
own phonenumber.
I dind find the solution th this problem in current CVS source,
chan_sip.c has to be updated.
Elmar
_
Hi,
On Upgrading do 1.2 I do have an PRoblem with my VOIP-Provider.
Making outbound calls does result in "Error 400" - exept if I do call
my own phonenumber.
Usimg my SNOM190 directly or reverting to 1.0.9 does resolve the
problem immediately.
What has to be changed in SIP config to move f
Hi, somebody has implemented Asterisk in one organizacion with amount
of extenciones in the order of 20.000?
If you intend to implement an PBX with 20.000 phones attached you have
to estimate the number of simultaneous connections.
I would estimate this to be the bottleneck, not the existe
[EMAIL PROTECTED] schrieb:
Is there a way to detect (via batch) if asterisk is idle i.e. is there no
active channels ? (oh323 show channels via console)
use "asterisk -rx 'show channels'" to execute an console command from
batch and evaluate the output.
I need to reboot every day an aster
As you can see bellow, the node /dev/capi20 exits and permissions
seems to be good (read and write for user and group). Do you have
other ideas to help me to resolve this problem?
Is user "asterisk" member of group "dialout"?
Elmar
___
--Bandwidth a
I have an * box where calls come in and sit in a queue until an agent
is available. I noticed that at the end of the day, I end up with a
bunch of "zombie" mpg123 processes for calls that were once on hold and
this seems to be eating up memory.
There should not be several zombies remaining,
I would suggest chan-capi-cm for any configuration.
You know which quadbri cards it works with?
I'm using an Eicon-Diva-Server 4BRI.
Elmar
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@li
Can you tell me what are the main differences between chan_capi
(http://www.junghanns.net/en/chan_capi.html), and chan_capi-cm (
http://sourceforge.net/projects/chan-capi)
chan_capi-cm is directly derived from the last development source of
chan_capi.
It does contain lots of fixes and severa
I've a strange problem here. I can dial out via an AVM B1 card.
I have a sip client running. I can hear my conversational partner
but he can't here me. I'm using * 1.0.
For SIP and CAPI operation there is no soundcard required at the
asterisk server.
Perhaps your SIP client does require one
NORMALIZE="nice -n 20 /usr/bin/normalize --no-progress -a 1.0 --peak"
Which package comes this "normalize" from?
Elmar
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://list
Hi,
I do have two questions regarding call queues:
1) How can I reach that waiting calls are also removed on removing the
last agent listening to the queue. All I found is the switch to
prevent new calls enter the queue after the last agent left.
2) Currently my queue does ring the agent aft
Hi,
how can I setup an callqueue wich has beside the groups of phones
usually routing calls to an second group to which calls should be
routed which oterwise would stay too log in queue (e.g. after 3 Minutes).
Elmar
___
Asterisk-Users mailing list
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
General config might be ok - as long as national and international
prefix are "0" and "00"
[interfaces]
; msn=50
; incomingmsn=*
;controller=1
softdtmf=1
accountcode=
context=incomingtest
;echosquelch=1
;echocancel=y
Have you planned to integrate some echo cancel feature ?
Besides the Eicon-CAPI feature there is an "echosquelch" in the driver.
Elmar
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-
Does It sound like correct?
The AGI should set an Variable to indicate if to block or not.
> What do you think about execute a script for
every call, query the database each time..? Do you think It'll
overcharge the system?
Usually that should not be a problem since it is run at call
initiation
For example, if you use an Point-to-Multipoint ISDN connection (not
'Anlagenanschluss'), then you won't get an immediate 'BUSY' on SIP
Busy/Congestion.
It's not possible to signal the caller 'Busy' or 'Reject', because there is
a timeout on the ISDN-Bus for ANY OTHER device which may answer the
Then I hope to receive some reports on what is buggy/not working, wishlist
and hopefully also some reports on what works well.
There are at least two anoying bugs:
1. The Busy-Applicatzion does not work, there seems to be no was to
singnal Busy to the caller is no SIP-Phone is ready to answer the
Hi,
how can I determine the status (busy, offline, ringing, duration of
current call) of an SIP phone?
Elmar
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
I tried the config from Damian and works for me.
I do not intend to use an Voicemail here.
The Initial "Answer" results in the caller to be charged for the call
even if no connection is made. That's not really MY problem, but I
would prefer to have an correct solution.
Elmar
What do i have to confiure so that a call comming in the * server through
chan_capi recognizes a normal busy line beep if the SIP phone is busy?
Presumably you have to fix the code and recompile chan_capi.
I did try the same without any success, I'm shure that it's an
chan_capi bug.
The only meth
Is it possible to have on the same machine an ISDN Fritz Card and a
TDM400 with two FXO ports? If so, is there any place I can find
instructions to configure it?
There should be nothiong special in using two cards. Just insert both
cards into different slots an configure each card according to t
Lastly, my capi.conf (as below) only defines one controller as this is
what we are testing with. My understanding is that the interface block
(starting with 'msn=470' and ending with 'devices=2') needs to be
repeated for each of the four BRI adapters, but with the correct MSN for
each.
If you
exten => s,1,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]&Local/[EMAIL
PROTECTED])
[context]
exten => 1,1,Dial(SIP/1)
exten => 2,1,Wait(5)
exten => 2,2,Dial(SIP/2)
exten => 3,1,Wait(10)
exten => 3,2,Dial(SIP/3)
Basically, use the 'local' channel for your dial, then you can wait a
bit befor
Hi,
I intend to let several SIP-phones on my asterisk ring cascaded on
incoming calls.
First only phone 1 should ring, after 5 seconds phone 2 should ring in
addition and after additional 5 Seconds phone 3 should also ring.
How can I realize that correctly?
Currently I do use
Dial(SIP/1
Hi,
I did recognice an rather strage behaviour on "Music on Hold":
Situation
Caller C does call Person A
Person A puts C on hold to ask B
MOH is (correctly) activated for C
After talking to B A does hangup to transfer C to B
In this moment MOH is activated fo
Hi,
I'm using Asterisk 1.0.2 with several lokal IP-Phones and An
Eicon-Diva-Server card for external connections.
If one of the Phones is called from external while it is switched off,
or busy the external caller still gets an ringing indication after the
dialplans "Busy" command is executed.
Hi,
I do have an Problem in my configuration:
At least on outgoing calls the start of conversation (where the other
party typically tells the name) ist lost.
I'm working with CHAN_CAPI and an DIVA-Server Card on Linux 2.6.9.
What can be done to fix that problem.
Elmar
> Hi,
> Have you made sure to stop and restart Asterisk after making each change?
> Mike
Yes I did an "/etc/init.d/asterisk restart" after changing the values.
Elmar
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listi
Hi,
I do have a problem on my asterisk-configuration:
Linux 2.6.9
Eicon Diva-Server ISDN-Card
Chan-CAPI
Grandstream BudgeTone 100
The problem is that when calling from the budgetone I can hear
the opposite party at suficient volume but the opposite party
can hear
Hi,
how to configure * to send an SMS to an mobile phone (Germany, D2).
in extensions.conf I did insert:
[smsdial]
exten => _X.,1,SMS(default,,${EXTEN},${CALLERIDNAME})
exten => _X.,2,SMS(default)
exten => _X.,3,Hangup
In the outgoing directory I do playe an call-fil
Has anyone heard of success using the EICON DIVA Server 4 BRI with
chan_capi as a PSTN interface with ISDN/BRI and * ?
I'm currently installing such a setup. Up till now I did not have any
trouble - but I'm still learning to manage *.
At first you should make shure that the CAPI is working correc
caller id on/off, ...
^
Should I interpret it that simple ISDN cards supported by I4L doesn't
support CLI/CLIP/CLIR?
No, it yust says that you cannot select by software if to transmit
caller id. If the line is configured to generayyl transmit ID it
should be ok for you.
Elmar
__
38 matches
Mail list logo
