Displaying 20 results from an estimated 1000 matches similar to: "Inbound PRI calls drop after 5 seconds using Sangoma A101"
2006 Feb 28
1
FW: Re: Delay on Phone ringing
Skipped content of type multipart/alternative-------------- next part --------------
asterisk1*CLI> soft hangup Zap/1-1
Requested Hangup on channel 'Zap/1-1'
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm'
== Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
--
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 04
1
no ring to callers?
OK, i've got asterisk @ home 1.3 up and running with Broadvoice.
BUT I have one nagging problem to sort out. When you call my BV # the
calling party gets no ring indication, just silence until either I
answer the phone, or the call bounces over to voicemail. below is the
console output when a call is recieved. what am i missing here?
thanks
Bernie
-- Executing
2005 Jan 26
1
Inbound analog Telco line not answered
I have an X100P clone hocked up to an analog line of my PRI. I can use it
to dial out.
but when I call the extension it answers and says "GOODBY"
I have a Livevoip DID which successfuly rings to ext 202
I am using asterisk@home and through the AMP inface the line should ring to
ext 202
Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf
Extensions.conf
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
2006 Feb 21
1
Outbound Routing does not use Multiple Trunks
Hello,
I have a TDM400 and currently have 2 of the ZAP Trunks configured
on it. Zap/1-1 and Zap/2-1. I am Running Asterisk@home Version 2.4
with AMP version 1.10.010
In my Outbound Routing I have the Trunk Sequence set up so that 0 is
Zap/1-1 and 1 is ZAP/2-1 What I see is that when Trunk Sequence 0 is
full, it does not open Trunk Sequence 1. I have found that this is true
even if I
2005 Aug 21
0
Problem with auto-attendant config, I think..
I never heard anything on the AMP list, so figured maybe someone here might
be able to help me sort this one out..
I was making some updates to my attendant config, which is really very
basic, and now incoming call processing stopped. Not sure exactly what the
heck happened, but figured maybe someone could help me with a clue as to what
broke. Now incoming calls are not being answered at
2006 Mar 26
1
AAH: DNID not set if caller suppresses CID?
Hi,
using asterisk@home, with quadBri from junghanns.net I am facing a
strange problem:
I have set incoming routes for some extension / DID:
[ext-did]
include => ext-did-custom
exten => 23,1,SetVar(FROM_DID=23)
exten => 23,2,Goto(ext-local,23,1)
exten => 57,1,SetVar(FROM_DID=57)
exten => 57,2,Goto(ext-local,57,1)
exten => 66,1,SetVar(FROM_DID=66)
exten =>
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 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
2007 Jan 25
1
IAX softphone fails through PRI trunks with Hangup
I've a call center using IAX softphones provided by a third party.
We've observed problems where the IAX phones seem unable to use our PRI
trunks. A sample anonymized call is provided below with the PRI debug
calls embedded. Any thoughts,
comments or suggestions would be welcome. In anonymizing it, I preseved
the format
and number of digits sent.
-- Accepting AUTHENTICATED
2006 Feb 22
0
Outbound problem sip chanel
I setup my aah box with a sip trunk at irisxa.iristel.net
Incaming it is ok but when I try to dial 8 and the nr where I want to call I
get all line is busy.
In my log I have these:
Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command'
Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command'
Feb 22 14:33:19 VERBOSE[2721] logger.c: --
2006 Jan 27
0
Digium Wildcard TDM400P call pickup timing
I have an analogue trunk to an AT&T Definity.
It has a DISA context defined.
From a Definity handset call the analogue port extension 1008 and wait
for dial tone from asterisk. It takes between 3&4 rings.
Likewise from Asterisk SIP handset <PBX Access No><PBX Extn> takes
nearly 10 secs to ring.
Is this configurable?
Ian Cowley
-----Original Message-----
From:
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 27
1
Extensions go straight to voicemail
Hello,
I have setup a test server with asterisk/AMP and have several 7960's
connected to it. The asterisk server has a public ip and all the
7960's are behind nat'd routers. When I try to call from extension
to extension I get directed straight to voicemail. I do not have any
cards installed and instead direct everything to an Ondo server. I
have been told it's not an AMP
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 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 Jan 27
5
External IAX2 phone defined as internal behaving as from PSTN
Have asterisk@home 1.2.1
The server is on an internal network eg 10.10.10.10
It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
50.50.50.50
The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
extension 1055.
Outbound calls to 1055 work perfectly.
Inbound calls from 1055 get picked up as if it were an external call
(see below) and goes straight to the ring
2006 Mar 31
4
cannot set outgoing cid
Hi,
sorry for the long debug output below. I configured Asterisk with AMP to send
the whole number including the extensions of the callers to the called party.
Whatever I configure in AMP it looks like it is used, In my eyes it is ok, but
doesn't seem to work.
033811234451 is the call id i configured, and it seems to use them, but the
caller will only see a 0338189040 instead of my
2006 Mar 22
3
PRI DMS100 -> Nortel Meridian Option 81
Hello all,
I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option
81C system. The PRI line is currently setup as DMS100. Here are the
relevant lines from zaptel.conf and zapata.conf:
zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone = us
defaultzone = us
zapata.conf:
[channels]
language=en
context=from-internal
musiconhold=default
switchtype=dms100