Hi Guys,
I'm having sound problems when diverting a call using [EMAIL PROTECTED] 1.5. I
am using the following configuration in extensions_custom.conf,
extensions_additional.conf and extensions.conf
[custom-Sales]
exten = s,1,SetVar(DivertNumber=02)
exten = s,2,Dial(SIP/116, 15)
I have a problem receving calls via the ISDN line, using the followin
components
Asterisk 1.0.9 with [EMAIL PROTECTED]
chan_capi-cm-0.6
AVM Fritz card
datalink protocol = point to multimode
I can make calls out with no problems so the issue is only incoming calls.
When I make the call from an
Thanks Armin
The call is rejected by Asterisk, so it looks like your dialplan
has no rule for accepting calls to '99546476'.
Armin
My dial plan as shown below is,
[capi-in]
exten = s,1,Dial(Sip/123,20)
exten = s,2,Voicemail(123)
exten = s,3,Hangup
I believe I should be able to receive
James and Armin,
Turn on asterisk debugging too. Capi seems to be working okay, maybe
asterisk isn't picking up the call for some reason. Maybe:
asterisk -r
set verbose 9
set debug 9
capi debug
then make an incoming call and copy the output into an email and send it
to the list (unless it
Armin,
2. Incoming DTMF tones are not detected via this ISDN line and AVM Fritz
card
Did you set for softdtmf/relaxdtmf?
Armin
I have tried with the following in capi.conf:
softdtmf=on ;enable/disable software dtmf detection, recommended for
AVM cards
relaxdtmf=on ;in
Hi List,
I'm having a problem with detecting incoming dtmf tones with chan_capi,
using an AVM Fritz card. I have set up softdtmf and relaxdtmf, both to 0,
expecting that the capi module will detect the tones, but it did not. I also
set both to 1, expecting that the asterisk dsp functions will
Can anyone please provide some help. I have installed an AVM fritz card on
an asterisk box ([EMAIL PROTECTED] version 1.5). I have installed the card
driver and chan_capi-cm-0.6. According to the installations guide I can now
see that the CAPI channel in asterisk is up,
*CLI capi info
Contr1:
anything except a tone dropping the call.
Armin, I will appreciate if you can put me in the right direction?
Cheers
PolAus
From: Armin Schindler [EMAIL PROTECTED]
To: Esteban Guana-Jarrin [EMAIL PROTECTED]
CC: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Incoming calls via CAPI
Remco,
Did you figure out what was happening, did you get it working?
Regards,
PolAus
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Hi Yair,
Please let me if you managed to fix the DTMF tone issue, which you were
experiencing couple of months ago. If not can you share any advancement.
I'm currently experiencing the same issue, I can make outbound calls but
DTMF will not work when dialing IVRs. My configuration is [EMAIL
Hi
Can someone please help with the following,
We are using [EMAIL PROTECTED] 1.5 and SIP trunks to communicate to the PSTN
network. We are having some problems with the call quality.
Although we can hear the other person's voice quite clear when making or
receiving a call, we get complaints
Hi Stewart,
Are you sure that this is an Asterisk problem? Configure an IP phone,
ATA, or softphone to connect directly with the provider, and check the
quality. If it's bad, use tools such as http://www.testyourvoip.com/ and
http://www.pingplotter.com/ to troubleshoot.
If standalone
You could be saturating your upload traffic? What is the upload speed of
you connection?
hads
Hads,
Our link here is a symmetric DSL (2M/2M) the upload link isn't saturated, it
doesn't go above 300 kbps in 5 minutes average intervals.
I have configured an IP phone and connected
Hi List,
I'm planning to setup and put in production a server with an [EMAIL PROTECTED]
2.8 edition and I would appreciate if someone that has done it
shares/provides some information in regards to the following questions I
have,
1. Since one of the most attractive features of the @home
Helo List,
I'm having an issue using the AND () operator in the code of my dialplan.
The dial plan is coded to detect inbound DTMF digits from callers. key 1
is equivalent to yes and key 2 is equivalent to no in my dial plan.
When a caller presses 1, yes is passed as a varialble and same when
Helo List,
Sorry I missed the rest of my email in my previous post. Please see below.
I'm having an issue using the AND () operator evaluation in the code of my
dialplan. The dial plan is coded to detect inbound DTMF digits from callers.
key 1 is equivalent to yes and key 2 is equivalent to
MM,
The $[] made it work. Thanks a lot for your assistance.
obligado :)
See the debug output below.
-- Executing NoOp(SIP/123-d14f, no) in new stack
-- Executing NoOp(SIP/123-d14f, yes) in new stack
-- Executing GotoIf(SIP/123-d14f, 0?7:4) in new stack
-- Goto (test-check,s,4)
--
Hi List,
Can any one please let me know how to pass arguments to the agi script from
the dialplan?
I read that it is possible to pass arguments to an AGI script here,
http://home.cogeco.ca/~camstuff/agi.html, by entering the variable followed
by a vertical bar but it doesn't seem to work
Hi List,
I am experiencing an issue in a server that I have installed asterisk;
configured an AVM FRITZ card to work with the capi module.
Once istalled the card works perfect; however every time I reboot the
machine I found that I have to re install the capi4k-utils before I can load
Hi List,
I am experiencing an issue with a server running asterisk; I installed an
AVM FRITZ card and configured it to work with the capi module.
Once everything is installed the card works perfect; the issue is that every
time I reboot the machine I have to re install the capi4k-utils
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