Displaying 20 results from an estimated 5000 matches similar to: "Why use 'Answer'?"
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'
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 -
2004 Dec 23
1
Qestion about TDM over enthernet
who can tell me how to do TDM over enthernet ?
pc a connect pc b only use TDM card?
thank you
John.
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2004 Sep 02
5
Any way to _always_ execute certain commands in a dialplan context?
I've got a need to do something like the following:
[foo-context]
exten => _.,1,SetCIDNum(123)
exten => _.,2,SetCIDName(XYZ)
include => local
include => tollfree
But of course, this example won't work. The goal here is this: if a call
ends up being handled by the "local" or "tollfree" contexts, I want
those SetCID*** commands executed. Otherwise, I
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 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
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.
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 =
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 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"
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
2004 Sep 20
2
1 extension entry for multiple purposes?
Hey gang,
There must be any easy solution for this but my mind is frazzled on
compiling 2.4 with RTC as module. Bleh.
Currently extension 9000 is our VoicemailMain(@company) line. Some
employee's are complaining that the old system was better because you didn't
have to enter your mailbox number and that instead the old system took you
right to it.
I figured there was something similar
2005 Aug 30
2
Manipulate CALLERIDNUM
Can someone tell me how to do this...Given the following line:
exten => *97,3,VoicemailMain(${CALLERIDNUM}@default)
Is it possible to add some logic to manipulate the CALLERIDNUM to send
back 801 even if the extension is 601 and 901 even if the extension is
701? I have 2 branch offices where users have both Office and Home SIP
phones. I want them to share a VM box.
Branch1 = 8XX , Home =
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 Jun 29
5
Problems with OR Logic in the GotoIf Statement
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2004 Dec 18
4
Free World Dialup and Asterisk
Hi forum,
I have been fighting days and days configuring FWD and asterisk with NO success
I have the following scenario.
My sister in Spain with FWD dialup client
My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone.
Spain LAN
FWD
2004 Jul 27
2
Enum
You can play also with www.enum2go.com <http://www.enum2go.com/> or
wap.enum2go.com
Regards
Alex
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2004 Feb 03
1
Mediatrix sip fxo gateway workaround?
Possible Mediatrix 1204 fxo sip gateway workaround
Need some feedback from experienced * users relative to this workaround
please please please.
Problem: The mediatrix 4-port fxo gateway does not provide any mechanism
for * to select which "port" an outbound pstn call will use. (See lots
of previous posts over the past four days for more detail if needed.)
Our reseller has been
2005 May 28
1
cmd curl crashes asterisk:
I recently began using the curl cmd to do an external callerid
lookup on my own customer database. I've noticed certain lookups will cause
a crash and not show anything in the messages file or the console. The curl
command is connecting to an external webserver which has a oracle db
connection. The file its hitting is PHP and does a very simply lookup
showing the text like "C1234 Bobs
2005 Mar 20
2
Follow-Me Script
I am trying to implement a follow-me script
(http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a
brain fart as I haven't a clue where to get started with what to do with
this. From my main menu, I want the extension 300 to execute the script as
follows:
exten => 300,1,dial(sip/200,20)
exten => 300,2,playback(pls-wait-connect-call)
exten =>