Displaying 20 results from an estimated 2000 matches similar to: "Inbound analog Telco line not answered"
2005 Jul 11
1
ASterisk@home + Broadvoice = Almost working installation...
Hello Guys,
I'm somewhat of a newbie and am desperately seeking for some help...
I've managed to get asterisk up and running on my server, and signed up with a
broadvoice account...
I'm having no problem dialing and communicating between extensions, but whenever
anyone tries to call my broadvoice account, they are greeted by no ring or
anything, but rather simply a direct to
2005 Jan 18
0
AMP and Asterisk PSTN extension config
Hi,
I have configured an Asterisk server with TDM01P (1FXO) for testing purpose. The interface I'm using is AMP. I want to configure my extension so that when I dial from my mobile phone to the asterisk line, I want it to transfer the call to any extension, say 3042 and after a particular number of rings, transfer the call to voice mail so that I can record my message.
My Zaptel.conf is as
2005 Aug 10
0
tdm400p / outbound zap prob
I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p
w/ 4 FXO. Incoming calls work fine, outbound I get this:
-- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack
-- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack
-- Executing GotoIf("SIP/231-af2b", "0?19") in new stack
2005 May 26
0
capi dial in/out configuration
Hi all,
I've recentrly starting to play around with *, when all I wanted is to
configure an fritz ISDN card with A@H.
Currently I'm stuck at the phase of what do I do with capi after
everything is installed.
I'm trying to understand how to setup incoming and outgoing calls at A@H
since I'm getting a bit lost with the default dial plan.
It seems that * answers but disconnect
2005 Jul 05
1
Newbie question reg. Asterisk and Channel Access Bank I and TE110p
Hi,
I have some problem to get this setup working. I have a CAC Channel
Banl I, with FXO and an Asterisk box ( I am using Asterisk@Home 1.2)
and I have a TE110p installed in this box.
What I want to do is, just to be able to dial one of those lines that
already are connected to the channel bank, and transfer that call
through TE110p and Asterisk to a user agent somewhere through
Internet.
2005 Aug 02
12
WHat does it take
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm
2005 Jan 28
4
FW: FAQ missing info? Asterisk@home V 0.4
Just installed V 0.4 of asterisk@home
Programmed up 3 sip budgetone extensions, they call call each other
fine.
Tried to dial '9' for an outside line through an X100P to a packet8 ATA
but got 'all circuits are busy now'.
Here is the console output.
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/30-8d25'
-- Executing
2005 Mar 06
2
Need help on * anf HFC.
Hi, I'm a newbie on * trying to setup an HFC card.
I'm locked for many days getting the all-circuits-busy. And no idea what
else to look for/how to diagnose.
I'm in Spain, I've tried changing many parameters on zapata/zaptelcong
with no luck, also NT & TE modes (honsetly, I've no idea what is).
Any clue will be very much appreciated!
I've installed *@home on my RH9,
2006 May 26
1
Not able to make any calls
Hi All,
I have registered "abhijit" for SIP in asterisk Server.
I am able to register my softphone (SJPhone) to the server using the
name "abhijit".
But whenever I try to make any calls I am gettinh the following error
message:-
*CLI>
-- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120
May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper:
2004 Dec 21
5
AMP - Fax Detections
Does anyone know of any obscur reference for detecting an incoming fax.
I currently have AMP running and everything else is working great.
Installed the spandsp patches and software... using the default AMP
extensions.conf, I start sending a fax, I hear it pick up and transfer
to voicemail after 20s.
Fax is set for system... Here is the detail from the extensions.conf
[global]
FAX_RX = system
2005 Mar 22
1
Call file misbehaviour
Greetings *`s,
I am manually creating call files and dropping them into
/var/spool/asterisk/outgoing to be picked up by *.
Presently, when I use local/internal parameters using SIP it works..ie I
make an internal call from device to device.
However, when I try dial an outside number which I have set up in a
custom conf file, it bombs out with the following message :
2005 Sep 16
0
Unable to create ZAP channel - All circuits are busy
Hello,
I have *@Home 1.5 installed and all is working fine for incoming calls and
sometimes outgoing calls. Installed in the box is a digium TDM04B (4xFXO
Ports)
setup as ZAP1 to ZAP4. I have incoming calls coming in on lines 1-4 in that
order and outgoing calls prefering ZAP4 then ZAP3 then ZAP2.
When i try to dial out to the PSTN from a SIP phone it sometimes works
(normally after a reboot)
2005 Jan 09
2
ASTCC Trunk and Routes Configuration
Dear List members-
I am trying to configure ASTCC (Asterisk calling card application) but
having a hard time to configure it properly. My project deadline is
approaching and couldn't figure out how to make ASTCC functional. Here
are some details what I have done so far.
1) I have installed ASTCC successfully.
2) I can access astcc-admin.cgi script without any problem.
3) I have created
2005 Feb 28
2
Fax Failing
Hello All,
I am trying to set up faxing using Asterisk@home 0.6. I have followed
the instructions to the best of my knowledge. When a fax comes in, the
system seems to detect OK but does ot manage to make the fax to pdf to
email leap. Here is what I saw in the CLI when I tested. Any help
would be appreciated.
Thanks!
Wiley
-- Starting simple switch on 'Zap/2-1'
-- Executing
2006 Feb 13
0
problem with outgoing calls Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion)
hi
i've configured a TE205P on asterisk at home
this is my
zaptel.conf
span=1,0,0,ccs,hdb3,crc4,yellow
span=2,0,0,ccs,hdb3,crc4,yellow
bchan = 1-15, 17-31
dchan = 16
bchan = 32-46,48-62
dchan = 47
loadzone = it
defaultzone = it
and my zapata.conf
signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
2005 Sep 28
0
Trying to cut out the paper work...
Hello everyone,
Ok. I am at a bit of a loss.... and would like someone to point me in
the right direction...(btw www.google.co.za did not give me ANY solutions).
The issue at hand is simple, I get asterisk (1.0.9) to answer the
incoming call with no problems... it does the fax detection thing with
app "Answer" and well it goes to the perfectly right context and sets
the varibles
2006 Feb 13
1
problem with outgoing calls Unable to createchannel of type 'ZAP' (cause 34 - Circuit/channel congestion)
Nik,
Just curious - what is your telco setup? Do you have PRI with the
specified D channels? You need to make sure that your telco is set up
to have the D channels on 16 and 47. When you first start Asterisk, or
when you log on to the CLI, do you ever see messages stating the B
channels are successfully started?
Let us know.
-MC
-----Original Message-----
From:
2005 Aug 09
3
SIP-Trunk problem, Please help!!!
Hi,
We are using VOIP-SIP gateway to route outbound PSTN calls.
Recently, I am getting == No one is available to answer at this time
message, after making 5 SIP attempts (Retransmitting #5 (no NAT):),
and the calls are going out through alternate Zap-trunk.
I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls.
Strange thing is that this is happening randomly,
2006 Jan 09
0
Call Rules
Hi,
I apologise if this is not the correct place to post such a message. I use
Asterisk@Home package and all seems to be going well.
I have identified one problem and have not managed to find anyway to
fix(modify) it.
We have a menu option that diverts to a mobile. If the mobile is off the
network sends back a message to that effect. Now, this mobile does not have
voicemail and asterisk is
2005 Sep 19
0
problems with remote access to PSTN
Hi,
I configured a pair of asterisk box, one of them (the first) ISDN connected
to the PSTN.
I configured an "integrated" dial plan, and I SIP connected the two pbx.
No problems in dialing extensions defined in one of the two pbx from any
pbx.
My problem is this: How can I make the users logged in to the second pbx to
dial
an external number using the ISDN connection of the first ?
I