Displaying 20 results from an estimated 600 matches similar to: "Newbie extensions.conf I need to include [SMS] context."
2004 May 25
0
Question IAX and SIP bound to different IP's on the same * box
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2004 Aug 29
1
Empty Queues
Hi,
Is there a way to detect if the caller will be entering an agentless
queue? I'd like to be able to redirect any caller who tried to join a
queue with no logged in agents, to be redirected to the groups
voicemail. Is this possible? I know I could create a menu and an
announcement for voicemail (should the user wish to drop from the
queue); but they wouldn't know no one was taking
2004 Jun 08
3
SMS in the UK
2004 Dec 19
4
SMS - how to send one
I've read quite a bit in the older mailing list posts and the wiki but
I'm missing some simple point.
1) What is required to send an SMS to a mobile outside the office given:
Channel: ZAP/1
send it to $SMS_RECIPIENT (which includes the final "extra" digit)
via
$SMS_CENTER=the national message center server for sending messages
$MESSAGE= the message text
How is the .call file
2007 Jun 04
2
FX Dialing Odd
Here's a possible bug, or more likely, I'm just missing something.
We have a pots card in one of our asterisk boxes. Its a simple asterisk
setup with one FXO/FXS card and basic static extensions file, etc. When
we dial out over the pots line, 4 out of 5 times, it will work. However,
every 4 or 5 times, we get an error back from the provider that says
"The number you have dialed.....
2004 Apr 27
1
Queue() with H option
Has anyone used the H option for Queue() with Callback queues? I want
customers in my queues to be able to jump out to voicemail when they get
tired of waiting, but in my setup when I pretend to be a customer and
press '*' [when I am waiting in the queue] I see the message 'User hit *
to disconnect call.' but then just jump out to the outer loop where
queued callers wait to
2005 Jul 01
1
asterisk newbie and phones which don't want tocomunicate
hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)???
are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering...
________________________________
From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa
Sent: Fri 7/1/2005 6:43 PM
2006 Mar 24
5
GSM/DECT handsets (was gsm picocells)
Now that I actually try and google for it, I can't find any dual mode
GSM/DECT handsets, only pages telling me that they exist without any
actual information!!!
Does anyone know of any such handsets? (and even better, ones that are
available in Australia) I've searched a few of the major gsm
manufacturers (nokia, Panasonic, sonyericsson) but their web sites are
absolutely pathetic to the
2004 Dec 02
2
Asterisk with SMS
Hi all,
I am trying to setup the SMS of Asterisk. I have a Siemens SMS enable
fixed phone which connects to my Asterisk through PSTN. Currently, I
can use my fixed phone to edit and send messages to the Asterisk.
However, I cannot make my Asterisk to send messages to the fixed phone.
The SMS command displays TX and RX records, hang for a while and then
stops with non-zero exits.
I read
2005 Mar 15
2
Asterisk Queue strange behaviour
Hi.
I have a problem which I assume would be easy to fix, but I can't find
anything about it...
I wish to have people dialing my phone, and if it is busy, they are put
into a queue. And then I am dialed back when the previous call is
finished, and connected to the waiting caller.
Easy enough?
----------exten
exten => 6,1,Background(salesq-intro);
exten => 6,2,Queue(salesq|tT|||300);
2004 Jul 27
5
sip over h323
Hi List,
we are using openh323 gatekeeper for voip telefony. We also have a voip
over ss7 TELES Switch for voip into POSTN Network. Know we want to use
Asterisk for converting SIP to h323.
Now my question. Is Asterisk an full h323 gatekeeper like openh323? Do
we need openh323 GK for astrerisk, too?. And how can i tell asterisk
to sent all none SIP-ip calls to the gatekeeper over h323?
thx
2009 Feb 26
2
Problems with Outbound Calls
Hi everyone!
I'm quite a newbie at this Asterisk stuff so please bear with me.
We've recently decided to start training in Asterisk via AsteriskNow!
Asterisk version is 1.4.18.1 through AsteriskNow! 1.02
The box we have is paired with a Digium TE110P and we've managed to get
it to the point where incoming calls via a single DID (from NTT Japan)
can be received and answered
2005 Jan 27
1
Stumped by BroadVoice SIP
Hello guys.
I am a fairly new user to Asterisk, and I'm just having a tough time.
My goal is to set up a VOIP PBX. I have signed up with a BroadVoice
number, and I have three systems with SIP phones.
The PBX and the SIP phones are all behind a Cisco PIX running NAT.
I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with
little luck.
The SIP phones are two X-Lites on
2003 Jul 16
8
Call Pickup
Hi,
I have been trying to workout how to use the call pickup.
So Far, I have the following in zapata.conf
[channels]
signalling => fxo_ks
context => local
pickupgroup=1
callgroup=1
channel => 1-3
When I dial *8# all I hear is busy tone.
What have I missed?
thanks
Jay.
2005 Jun 07
4
Queue Log
Hello everyone,
This is is my first email to this group.
I'm am writing a small php program to pull some info out of our
Asterisk's queue_log. I'm having trouble figuring out what some of the
parameters mean.
Here's an example:
1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||"Ray Balbin 25"
<(716)250-3405>
I found a doc that tells me about everything from
2003 May 23
12
Unable to create channel of type 'Zap'
I've just installed an X100P, built the kernel module, and tried to use it
to make an outgoing call (via a phone connected to an ATA-186). However, I
just get a reorder tone and see this on the console:
-- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack
NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to
create channel of type
2005 Jul 15
2
[Aserisk-Users]no audio inside the net
Hi list, i've problems with my * server and the 4 phones which are
linked to it. i've 2 grandstream bt100 with the firmware upgraded to
101, a wi-fi phone (i don't know its brand) and another ip phone i
don't know its brand. with this sip.conf :
[general]
port = 5060
bindaddr = 192.168.100.229
context = default ;x changed from default to sip
localnet = 192.168.100.0/24
2004 Jul 18
18
Polycom IP 500 Voicemail
Hello All,
I have some Polycom IP 500 phones that I would like to have configured
for direct dialing to our voice mail system. So far I have been unable
to get the hard button labeled Voice Mail to connect to Asterisk without
first passing through the message center prompts. I have followed all
the Admin Guide instructions regarding the phones .cfg files and using
2006 Mar 29
3
SMS in Spain (it seems Protocol 2)
Hello,
(I have asked it some time ago in Asterisk-es mailing list, so excuse me if
anybody receive it twice.)
I am trying to send SMS in Spain using landlines. It seems that
app_sms.c only handles Protocol 1, but Spain and Italy are using
Protocol 2.
I have been searching in Internet without any results... anybody is
sending SMS from Asterisk (or any method) using Protocol 2? (so, it
seems,
2009 Aug 07
2
realtime config and extensions.conf
Howdy,
My first forray into using res_mysql.conf for realtime access of sip users
and extensions.
I have the following relevant section of extensions.conf:
---
[trunklocal]
exten => _NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[local]
include => trunklocal
include => trunktollfree
[longdistance]
include => local
include => trunkld
[international]
include