Displaying 20 results from an estimated 6000 matches similar to: "PRI signaling experts please help"
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 Aug 24
1
Busy number signalling
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 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
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 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 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 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
2009 Aug 13
1
Autofallthrough delays before hanging up calling channel?
I am seeing some curious behaviour with a 1.2.32 system, which I do not
understand and so can't work out how to fix it.
I have a PRI routed to context default. Here is the complete default
context:
[default]
exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1})
exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN})
exten => _X.,1,Dial(IAX2/m1peer/${EXTEN})
exten =>
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
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
2005 Jun 30
1
No BUSY on PRI
I'm using a TE405P and stable version of Zaptel. When I call a BUSY number on my E1 PRI, I don't get a busy status. The caller hears a busy tone, but
the CDR logs a NO ANSWER when the caller hangs up.
Is this normal for this version of Zaptel?
2005 Aug 19
4
any ISDN/PRI signaling experts out there?
I have officially engaged in a pissing contest with the local Telco over
PRI calling name delivery.
The telco publishes their calling name delivery over PRI feature as
being bellcore gr-1367-core compliant.
The gr-1367-core spec states that the calling name is to be included as
a facility IE in the setup message, or sent in a subsequent facility IE
message with an indicator in the setup message
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
2007 Jul 17
5
Asterisk PRI Busy Problem
Hi,
I've an PRI coming to my asterisk ,calls are coming fine and my agents are
able to answer no prob. but I've an agreement with my telco with some
incoming no if the no of calls on these no are more then 3 then send to
another no. they use busy signal to divert call on another number so I'm
sending the call to Congestion() if no of calls in this group are more then
3. But my
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
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
2008 Nov 15
1
PBX -> PRI -> * -> Telco not working
Hello all.
I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box.
NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco
Incomming calls from the telco to the asterisk box to the NEC work fine with
indials and everything. Works sweet.
Outbound from the NEC to the Asterisk box fail. Giving an long dial tone
that then times out.
Ie, pick up NEC handset, dial
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 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