Displaying 20 results from an estimated 1000 matches similar to: "Any way to _always_ execute certain commands in a dialplan context?"
2003 Nov 18
1
Asterisk with External Voicemail
If anyone could help me with this, I'd appreciate it!
I've got an Asterisk deployment where I'd like to use an existing external
Octel voicemail system. I've been trying to define an extension that if
the call isn't answered in a few rings, to dial our external voicemail
number. That voicemail system works by seeing the CALLED number and
routing the call to the
2005 Sep 21
7
add 0 (zero) to incoming callerID - how?
I have an asterisk box and SIP / IAX2 phones.
To call out, users have to add 0 (zero) before a real telephone number.
That means, that if they want to call someone that has a number 123456,
they have to call 0-123456.
Simple, right?
This has a serious drawback though - when someone calls us from the
number 123456, we see the callerID 123456, and we're unable to use the
callback/redial
2005 Jun 21
2
Polycom and CallerID
I'm having a problem with the callerID that the polycom IP600 phones are
displaying. I would like to modify the CIDName and leave CIDNumber as
exactly what the phone call came in as(provided they aren't hiding
callerID). Most of the calls will be going to the queue, but a few will
go directly to the SIP phones.
I've done a various combinations of using SetCallerID(),
2005 May 18
2
Call forwarding...
Sorry for posting this again, but it seems to have become attached to
another thread. Guess I replied to another message instead of starting a
new one...
Hi,
I'm trying to setup a call forwarding rule so that when an extention
doesn't answer the call is forwarded to my mobile.
I'm using voiptalk.org for incoming and outgoing calls and SIP phones
for extentions (so all IP based -
2005 Jun 29
5
Problems with OR Logic in the GotoIf Statement
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2004 Dec 22
2
Why use 'Answer'?
Why is it that newcomers always feel like inserting 'Answer' is a
necessary step in their extension.conf entries?
>[voiptalk.org]
>;forwards any calls starting with an "8" thru voiptalk.org
>exten => _8.,1,Answer
>exten => _8.,3,SetCIDNum(55555555)
>exten => _8.,4,SetCIDName(My Name And Surname)
>exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk.org,,g)
2003 May 05
6
IAXTEL toll-free gateway
I have been playing around with asterisk for a week or so now and
haven't had too much trouble getting things to work but one thing seems
to puzzle me. I have been patient hoping that there was a configuration
error on the server or that the toll-free gateway was down but nothing
has changed. I have the following configuration context for IAXTEL:
[iaxtel]
exten =>
2004 Nov 24
3
Haven't got a clue ...
On how to even search for this "feature" as I have no idea on what it can
be.
I've got a meridian linked to * (by EuroISDN) which is linked to a ISDN30.
I can make calls from the meridian, and receive calls into the meridian.
Great stuff.
However, if someone dials an invalid number, then instead of hearing a
"three tone", the line just drops and goes dead. The console
2005 May 12
1
ast_yyerror - 'space' in Caller-ID - string comparison
I've some code to manipulate incoming Caller-ID - so its suitable for
replying to...
[sipdef]
exten => s,1,NoOp(FWD SIP: "${CALLERIDNAME}" <${CALLERIDNUM}>)
; Alter incoming calles from pulver - add a '87'
exten => s,2,Gotoif($[${CALLERIDNAME} = ${CALLERIDNUM}]?3:4)
exten => s,3,SetCIDName(87${CALLERIDNUM})
exten => s,4,SetCIDNum(87${CALLERIDNUM})
exten
2005 Sep 14
4
Echo on SPA-3000 FXO
I've had an spa3k in service here at the house for a while now. After
some initial wrangling, it's been working okay. I've had to reboot it a
couple times and have noticed something rather annoying though.
My setup is pretty simple and, dare I say, common. I have the SPA-3000
"inline" between my incoming POTS line and the internal house phone. It's
setup to deliver
2004 Dec 20
7
'I'nvalid extension handling problems, even with workaround
Hello folks,
I'm having trouble configuring Asterisk to play an "invalid extension" message to
anyone dialing an undefined extension.
First I tried using the 'i' pseudo-extension, but it didn't work at all;
searching the wiki I found that page:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension
where it basically says that the 'i'
2004 Dec 02
1
firefly and caller id
Is there a bug in Firefly (3rdparty) wherein it does not show caller ID?
I am using SetCIDNum(12345) before I dial my firefly (IAX2) phone... no caller
ID. CallerID is passed properly to other clients.
-A.
2005 Jan 28
3
FWD and IAX2
Hi,
I had a FWD account set up with asterisk (using SIP) and it was working
fine both ways. I switched to IAX2 and now I can't get incoming calls
from FWD. People who call my FWD number get a "480 - user is not online"
message without any traffic reaching my box. I can call FWD numbers fine
over IAX2.
It seems fwd isn't trying to place the call over IAX2 because it thinks
2004 Oct 07
5
Display called Number or context on X-lite/X-Pro
Hi to all,
to manage properly a call center for multiple companies is possible to let the
X-lite/X-Pro softphone to display the number or context called from PSTN to
let operator answer with the correct name of the company??
I explain better. If a call come from PSTN to Number A for company A i want
the operator recognize it and answer "Good Morning, I'm Operator of company
A"
2004 Aug 05
1
iConnectHere and CallerId
Is it possible to send the CallerId to IconnectHere with Asterisk
when making outbound calls?
I read somewhere that it doesn't work.
I have set up everything to send the correct CallerId info to IconnectHere
but I get a "442-887-926267" caller id.
In [globals]
ICONNECT1=1713...(my number)
MYNAME=My Name
I set up the Caller Id in the dialing macro:
[macro-iconnecthere]
exten =>
2005 Oct 13
1
SetCallerID Problem
My number is not submitted.
I updated my asterisk but this error still occurs coz of the "" in the
SetCallerID tag thats why it will be a empty SetCallerID is submitted.
Is there a fix to correct this error?
-- Executing SetCIDNum("SIP/31-752a", "4989427xxxx") in new stack
-- Executing SetCIDName("SIP/31-752a", "4989427xxxx") in new stack
2009 Aug 26
2
application missed in asterisk 1.6.1 - SetCallerID()
Hi
A few day ago, I notice that some applications missed in asterisk 1.6.1
release even if *.so file which normally create them were compiled during
Asterisk install.
SetCallerID(), SetCIDNum(), SetCIDName(), SetLanguage() ... and maybe so
more.
anyone already notice that to ?
If it's not normal, anyone have an solution to it ?
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2004 Aug 20
1
Incoming MSN via ZapHFC -> to SIP
Hi there,
I've got a small problem with the zaphfc channel. No MSN of an any
incoming call which comes trough the ISDN card (Acer ISDN, with HFC
chipset and zaphfc driver) which will be forwarded to the SIP-Phone will
be displayed. Always it will be shown "asterisk" an the Display.
--- snip (zapata.conf) ---
[channels]
language=de
switchtype = euroisdn
signalling =
2004 May 25
1
using asterisk with iptel addreses
was wondering if anyone could give us a run through an explanation of the
wiki and other examples of connecting to iptel's sip express router using
asterisk pbx so i can understand better the call processing ..
given the example i work from on john todd's www.loligo.com site ;
exten => _3.,1,SetCallerID(${IPTELUSERID})
exten => _3.,2,SetCIDname(${IPTELUSERNAME})
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