Displaying 20 results from an estimated 1000 matches similar to: "Autofallthrough delays before hanging up calling channel?"
2004 Dec 24
3
Preventing Asterisk from sending 'h' across to SIP Provider
Hi,
I want to prevent Asterisk from sending the h extension across to the SIP
provider or to prevent it from hitting the script at all. The SIP Provider
does not know what to do with the h extensions once it receives it. My SIP
Provider takes all digits and forwards them off to a softswitch for
processing. Everytime a call hangs up, it complains about running AGI scripts
on hungup
2006 Apr 10
4
callerid name inboune from PRI
I switched PRI vendors recently, and one of my questions was "do you provide caller ID name in addition to number?"
AT&T Local did not, But XO communications said they did.
Before I call to complain, is there an setting to turn this on in asterisk?
I want to make sure that I have my side covered before I call XO.
My current zaptel.conf is:
context=from-pstn
switchtype=national
2005 Aug 17
1
Comfort Noise incomplete - No translator path exists for channel type MGCP (native 4) to 256
I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below
I go off hook
2006 Apr 04
2
Fax over 2 bridged TE110P channels
Hi,
I have an asterisk installation with 2 E1 cards
Software version is
Asterisk 1.2.6
Libpri 1.2.2
Zaptel 1.2.5
I'm having problem with fax transmission, let me explain better my
setup:
My fist TE110P E1 card is connected to the telco line
the second TE110P E1 one to an Nexspan PBX
so the server is basically sitting between the line, and the pbx.
every call coming from the line is
2005 Dec 05
3
PRI indications.
Hello,
i have succesfullu setup asterisk with Sangoma E1 card, evrything works well
but i don't know how to pass indications from telco switch to the user - when
users call bad number telco switch shuld talk "unallocated number" but its only
send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients?
My /etc/zaptel.conf:
span=1,0,0,CCS,HDB3,CRC4
dchan=16
2005 Mar 17
2
PRI Cause Code Help
Hello,
I just got off the phone with my PRI provider, and was told that I am
not sending an expected message when I reject a call with a Cause Code
for Unassigned(1) and Congestion (42). Busy works fine though...
Anyhow, they are seeing the RELEASE COMPLETE message with cause code 1,
however the tech told me they expect a PROGRESS indicator with a value
between 1 and 10.
Any ideas on how
2005 Jan 27
2
Busy - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
hi,
well, most of the things work right now due to the help of peter
svensson, but after heavy use of our ericsson BP250 today several
problems appeared.
i split into several mails as they are seperate problems.
* i can't signal Busy to the calling party.
asterisk receives busy from the ericsson PBX but does not forward
this to the external caller. i tried with exten =>
2005 Aug 25
1
PRI signaling experts please help
Hi all,
Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed to protect the
innocent):
-- Called 12345678@sip-outbound
-- Got SIP response 486 "Busy here" back from
2009 Dec 22
4
asterisk & x-lite
Hello All,
I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can call the other one but I can' t hear any voice. My
configuration files are below:
[root at localhost asterisk]# cat sip.conf
[general]
canreinvite=yes
[1001]
username=1001
password=1001
type=friend
context=phones
host=dynamic
[1002]
callerid=1002
username=1002
password=1002
type=friend
2007 Jun 27
4
Customized Ring Tone
Hello all,
I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the
Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my
home PBX in such a way that whenever someone calls on my trunkline (PSTN)
number, he/she will hear a customized ring tone, probably playing an MP3
file, instead of a boring standard ring tone while the extension number that
is
2014 Feb 04
1
How to Busy signals on DAHDI
Hello,
On a Asterisk 1.6.1 powered system, I've just discovered that using Busy()
application in dialplan was no enough to send a Busy signal on incoming
Dahdi channel.
On this specific install, adding an Answer()) and a Playtone() statement in
dialplan triggered sending of busy tone but I'm still surprised by my
findings.
Should I expect public switch to send a Busy tone to caller
2010 Sep 15
1
One way audio when overlapdial is set to yes
Hi Group,
I am currently facing a dead end and any help will be much appreciated.
I have an a104d installed in an asterisk box, two of which is configured on ISDN
pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am
getting one way audio when a local on the hipath tries to make a pstn call but
no issue on incoming calls from pstn going to the hipath locals.
local
2005 Aug 26
1
Dial command nor progressing on Zap channels
Hi all,
Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed to protect the
innocent):
-- Called 12345678@sip-outbound
-- Got SIP response 486 "Busy here" back from
2005 Jun 20
1
Looking for PRI Outbound Caller ID Configuration
I'm having trouble setting the outbound caller ID on calls I make from my
PRI trunk group. The PRIs are served out of a 5ESS. Telco has set the PRIs
up for user provided caller id information, so I believe I just don't have
it set up right in my dialplan or something. I can't seem to find an
example of setting the outbound caller ID specifically for a 5ESS. Does
anyone have an
2005 Sep 14
2
PRI to PRI passthrough with DID intact
I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink
T1 ---- Asterisk.
I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk
extensions over the T1.
I do not get DID nor CID on the Asterisk, so I want to use PRI between the
PBXs.
I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are
different cards)
I see this as my least
2005 Mar 15
1
Unknown signalling 896?
I've been beating my head a bit against the 1.0.6 Debian builds of
Asterisk, using an E100P (E1, single span) board.
In machines I've built in the past (back in 1.0.0 days), config I'm
using and that card and 1.0.0 driver combo worked fine.
ztcfg reports no problems:
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
31 channels configured.
And zttool sees the card, and
2004 Dec 03
2
DIALSTATUS weirdness (CHANUNAVAIL instead of BUSY, NOANSWER instead of CHANUNAVAIL)
Just throwing this out here, hopefully someone can tell me why.
*CLI> show version
Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running
Linux
Zap/g1 is pri_cpe to Bell Canada
5551234 is a normal POTS line I have busied out (handset offhook)
exten => 1234,1,Dial(Zap/g1/5551234,,g)
exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
2005 Jul 03
1
asterisk strips off trailing digit from incoming calls
so here it is, the problem that's been nagging me for the past 2 days:
connected a box to my telco's NTBA <-> zap/asterisk. which works:
box:/etc/asterisk# cat /proc/zaptel/1
Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7)" HDB3/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In
2015 Aug 05
2
Asterisk uses "Anonymous", but why?
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
<a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage" target="_blank"
2009 Jun 20
1
PRI cause codes
I am trying to retrieve the cause code of a outgoing call over a PRI
where the number called is out of service. When an out service number is
called I get a recording that the number dialed is not a working
number. I see cause code 1 in in the CLI as soon as the call is dialed
the Telco recording goes on for 30 sec. then hangs up. Any idea on how
retrieve info that the called number is is