Displaying 20 results from an estimated 3000 matches similar to: "dialling in from analog line -> only get 2 of 3 digits extensions"
2004 Apr 16
1
Matching variable-length extensions with chan_zap in overlap dialling
I've been having trouble matching variable extensions on a zap channel
(an E1 line). Doing it the extensions.conf way:
[pri1]
; Match 8078078- calls
include => m807nat
include => m807mob
include => m807oth
[m807nat]
exten => _80780782XXXXXXXXX,1,StripMSD(7)
exten => _2XXXXXXXXX,1,SetVar,clidest=${EXTEN}
exten => _2XXXXXXXXX,2,Goto(cli,s,1)
[m807mob]
exten =>
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
2003 Oct 07
1
Dialling problems
Hey all,
I'm having problems reliably dialling out my FXO card. About 30% of the time
I'll get a "your call cannot be completed as dialed". I'm thinking it might be
the dialling speed, but I can't find any configs that change that setting.
Any suggestions for troubleshooting?
Thanks,
Brad Waite
2010 Mar 29
3
Slightly more advanced dialling..
Hello,
I'm wondering if it is possible to ring X number of extensions
simultaneously, and each answered call can be handled with some code.
I can do a huntgroup-esque way of dialling, but I want all the dialled
numbers to be picked up.
I hope this makes sense.. If not please say..
Many thanks!
Andy
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2005 Jul 28
2
delay on pri dialling when asterisk is spliced between E1-Pri and legacy pbx
hello everybody,
one of our customers which wants a soft transfer between his old pbx to
asterisk and sip. the setup is as follows:
telco <---pri---> asterisk <---pri---> legacy pbx
everything is fine exept that when dialling from the legacy pbx it takes
about 3 seconds before the asterisk start to dial.
where does this delay come from?
has it to do with
2010 Oct 29
0
Asterisk 1.6 Overlap dialling timeout?
Hello,
I'm experimenting with Overlap Dialling in asterisk 1.6.
I've enabled this in sip.conf and on the SNOM 300 phone.
My problem is that asterisk dials out as soon as it matches an
extension without waiting to see if the user is going to type in more
digits.
Is there a way to set a timeout per channel or globally?
I'd like Asterisk to wait for a few seconds once its found a match
2004 Jul 24
2
yes shady dial running now but not dialling
hi there
was wondering if anybody knows this..
have successfully installed shady dial and the agent is now logging in successfully
i've enabled postgres debugging and i found out that no request had been made by the shady dialer to query the database for the numbers...
also i am able to login to the queue simply by entering the agent id...
it doesnt ask for the password...it simply plays the
2005 Oct 18
0
Slow dialling from PBX into * via E1
Hi :)
I have a little 'slow dialling' problem. When I dial, e.g.
200# for the Asterisk 'echo test' demo application from my PBX extension
1010, I see this in the console the instant I press the # key:
-- Starting simple switch on 'Zap/65-1'
-- Accepting overlap call from '1010' to '200' on channel 0/3, span 3
then exactly 3 seconds elapses, and
2007 Jul 27
0
Keep playing Background while dialling invalid dtmf extensions
hi asterisk users
How can i make asterisk "ignore" invalid extensions, and go on playing the
background soundfile?
Normally, asteriks will take the user to the invalid extension if the caller
presses anything other than 1 or 2 in the following context::
[example]
exten => s,1,Answer()
exten => s,2,Background(hello-world)
exten => s,n,Goto(s,2)
exten =>
2003 Dec 09
1
dialling peer problems
I'm trying to use Jeremy's suggestion of dialling using just the peer name
instead of user:pass@peer but I'm running into some really funky issues.
It does the same thing with both VoicePulse and another * server I have.
[voicepulse]
type=peer
host=gw5.voicepulse.com
trunk=yes
user=USERNAME
pass=PASSWORD
and in my dialplan:
Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r)
The log shows
2008 Feb 13
2
UK issue - Asterisk dialling 999... sort of
Hello
This is a fun one for the list...
Twice now, the Police have contacted us to say they have had a silent
call then hangup from our landline number to the 999 service. As a
matter of course, they follow up these calls in case someone is in
distress. Nobody here was in distress - well, no more than normal! The
Police aren't hugely happy when we tell them it must be a mistake.
Thing
2006 Feb 28
3
Capturing DIALSTATUS on a PARTICULAR channel if multiple-dialling?
Using 1.0.9:
If I have:
exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1)
How can I return the DIALSTATUS variable for the second SIP channel ONLY if
the second SIP channel is busy, regardless of the dialstatus of the first
SIP channel? What I want is, if the second SIP channel is busy go to n+1 or
n+101 regardless of the status of the first SIP channel.
tia
2006 Apr 30
0
Intermittent problem dialling out on a SIP channel
Hi,
Red Hat 9.0
Asterisk 1.2.7.1
I'm having a bit of an intermittent problem with my SIP account.
Often (but not always) when I start * or RELOAD my dial plan from the
CLI I get this message:
>Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822
add_realm_authentication: Format for >authentication entry is
user[:secret]@realm at line 31
>Apr 30 11:01:21 WARNING[12785]: acl.c:244
2003 Oct 14
5
dialling out
When trying to dial out
982420173 our main number
I get the engaged signal before I finish entering the phone number
Any ideas ????
Regards Mick
2004 Aug 08
2
pbx answers after answering from analog phone
I am setting up my * for at home office and still have analog phones
attached and answer from those analog phones and not necessarily through the
pbx. I found that with the X100P cards, they see the 2nd ring and will be
ready to answer the line. I used a Wait to pause and allow another 2 rings
before * answers. But found that if we answer the line after the 2nd ring
and before the 4th, * still
2005 Jan 28
3
Sipua SPA-2000 and liong delay after dialling number
When I use an analog phone connected to a Sipura SPA-2000 it takes about
3-4 seconds before the number is actually dialled.
Very annoying especially if you are connecting an intercom to it.
Can I change this behaviour and do I need to look at * config or the
config of the SPA-2000?
Thanks!
2003 Jul 08
1
chanh323 dialling
what is the format for an h323 entry in the dialplan?
can I use chan_h323 without compiling anything else
or should I compile oh323?
basically what's the best way :)
cheers
Dave
2004 May 01
1
Outbound Dialling on ISDN using CAPI - Individual Dial out Plans using msns
Hi All,
Could somebody please help me to understand the following: -
We have 8 msn's 383590, 383591 etc.
What I would like to do is for the person at extension 1 dial out on 383
590, the person at extension 2 dial out on 383 591 etc.
I have got myself so confused that I need major help!!!
If you could give me a simplistic example, including which files I put the
coding in (i.e.
2005 Feb 13
1
Broadvoice international dialling question
I'd be grateful if someone could point me in the right direction.
I have a Broadvoice trunk attached to Asterisk which I use for frequent
calls to the UK using the following in extensions.conf
exten => _0[1-68].,1,Ringing
exten => _0[1-68].,2,Dial(SIP/BV/01144${EXTEN:1})
exten => _0[1-68].,3,Hangup
The caller hears immediate ringing, though it seems that Broadvoice
2005 Jul 20
0
Sipura 3000 x special dialling pattern (pin code)
I need to place a call using a "pin code". To access an external line,
the host PBX (a Ericsson MD-110) will require that I dial
*72*pincode#phone_number to complete any (trunk) call.
When I send the number, my Sipura 3000 will reject the call with
"Forbidden - wrong password on authentication for INVITE" (see below).
All other calls sent to the Sipura box without the