Displaying 20 results from an estimated 700 matches similar to: "Dial using URI(web) or using FORM(web)"
2006 Feb 17
5
A unique 'click to call' project - Could use some advice
Hello List,
I work for an IP communication provider in upstate NY as the engineer
assisting our technical support team.
We provide a number of different Telco systems to residential
subscribers; and in an effort to more effectively trouble shoot
termination problems I came up with the idea of creating a click to call
system that will allow our agents to effortlessly place test calls.
On a
2005 Sep 15
2
Caller ID for auto outgoing calls
Hi. I'm using /var/spool/asterisk/outgoing files to place automatic
calls, but I'm having trouble setting the Caller ID for the second half
of the call.
In other words, when we call the first number, we want the Caller ID
set to our number, but then when we connect them to the second number,
we want _their_ number to be the Caller ID.
I've tried the following (and various
2006 Jan 17
2
change error messages for Validation helpers?
Is it possible to change error messages for Validation helpers? I am
writing an app against a existing database (so no control over column
names), but when there is validation error (e.g. with
validate_presence_of) I would like to customize the field name. For
example for telephone whose field name is PhoneNumber I would like to
chnage it to "Telephone Number cannot be empty" rather
2005 Mar 05
7
BroadVoice configuration changes for Outbound
Today, We have added INVITE Authentication. This seems to bring a large
amount of problems to people in the way since they can't make outbound
calls. Here's what needs to be done. You need to add three variables to
your peers or friends, username, authuser, and secret.
username=<phonenumber>
authuser=<phonenumber>
secret=<registration password>
Dan
2005 Mar 25
5
Re-write callerid?
Is it possible to rewrite caller id's?
I would like to have sip phones appear by their local cid
(like Henk <208>) but when they call out using the PRI I would like their
full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid to the main phonenumber but surely there
must be a better solution?
Thanks!!
Remco
2004 Jun 01
2
BroadVoice usage?
Hi all,
I've been trying to use BroadVoice as a SIP service provider. They don't
officially
support * but are helpful when it comes to answering questions for setup
parameters. They claim they have no firewalls or access lists that need to be
set up so I can get access to their servers.
However, something's still not quite right when I use the parameters.
It looks like our Asterisk
2005 Feb 01
2
Outbound calling with TDM400P
I am trying to place an analog outbound call from a Sipura SPA-841
through a * server with a TDM400P and 4 FXO's. When I call in from an
analog line everything works fine, I can talk over the SIP phone. When
I call out, * says:
== Spawn extension (from-sip, [phonenumber], 1) exited non-zero on
'SIP/sipphone-d29d'
-- Executing Dial("SIP/sipphone-9eb0",
2006 May 09
1
Asterisk settings Net2Phone
Hi,
I?m looking for settings to configure net2phone carrier in my asterisk. I
found this configurations, but it?s not work. I don?t known if this
configuration is for voice line or voice access account.
Anybody can help me, with other configuration?
Thanks.
----
*sip.conf*
[general]
useragent = X-Lite release 1103m
register => PHONENUMBER:PASSWORD@sip.net2phone.com
[net2phone]
type = peer
2006 Nov 13
1
Sending '#' with Dial
Hi!
I have a working asterisk-setup with four sip-clients. Everything works
great but when the users call someone the phonenumber shows up on the
receiving ends callerid-display.
To correct this my provider told me to send #31# before the phonenumber,
tried this with: Dial(SIP/#31#${EXTEN}@provider) but my asterisk tells me
that it isn't a valid extension.
The INVITE looks fine,
2014 Aug 11
1
401 Unathorized
I have an asterisk 1.8.x box that intermittently returns a 401. Calls come
through the same peer all the time, from the same carrier. However
intermittently the asterisk box returns a 401.
Below is the output of a failed call (1st) and a successful call (2nd). I
can't see any difference until we get to these lines.
Bad call:
--- (17 headers 14 lines) ---
Sending to carrierIP:5060 (no NAT)
2008 Jan 18
1
Automatic call-out problem
Hello!
My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on
Fedora Core 4. I am making automatic call-out campaign with this setup
on 4 PRI. The scripts for this:
====================================================================
caller php script write this to outgoung folder:
fwrite($outfile,"Channel: Zap/g1/$phonenumber\n");
fwrite($outfile,"MaxRetries:
2003 Jun 11
3
Telephone Tree
Hi everyone,
I'd like to use Asterisk to build a phonetree (www.phonetree.com) type
of application, like this:
1. Read a text-based name/phonenumber file.
2. Call every number and play a recorded message.
3. If a beep is detected, replay the message from scratch (to leave
messages on an answering machine).
4. Write results to a log file.
Does anything like this exist already? Can this be
2009 Aug 31
2
Asterisk Regular expression to validate any phonenumber
Hi
I am using asterisk version 1.6.0.5
I have build up one utility that will fire Originate Action on Manager...
In which, i have define number to call eg. 919912312345 (MobileNumber)
How can i know that this number format is true for Indian Number...
In originate action, user can enter any international number.. How can I
came to know this number format is right for that country...??
IS there
2005 Jan 05
4
Broadvoice / * re-register issues
HELP!
Ok, so I have the following SIP.CONF:
[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw
register =>
##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234
[sip.broadvoice.com]
type=peer
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
2006 Jun 26
2
n-way has_mant :through
I''m trying to setup some mildly complex associations for a project we''re
working on and can''t seem to find much documentation on n-way has_many
:through associations.
I have the following models: Person, PhysicalAddress, EmailAddress,
PhoneNumber.
Each person can have multiple PhysicalAddresses, EmailAddresses, and
PhoneNumbers, and multiple people can share the same
2007 Feb 08
3
Automatic Dial, Play message
Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?
For example, you have a database of
FirstName LastName PhoneNumber
Jon
--
***
Forrest Beck
IAXTEL: 17002871718
jonforrest.beck@gmail.com
2006 May 03
1
How would you go about calling a list of numbers and 'speaking' a message?
Hello
I have been asked by a client to process a list of telephone numbers.
Asterisk should call each number in turn and if the recipient of the call
answers, play a message - eg from a wav.
How would I go about doing that?
Angus
2003 Nov 02
2
one way sound with x-lite (sip) -second attempt
Hi all,
Still having the one way sound problem.
Any suggestions how to hunt the problem down ?
Regards,
Thorsten
---------------------------------------------------------------
Hi all,
We have a very basic * installation for testing purposes.
The * is connected to PSTN with BRI and setup with X-Lite
over plain lan. (local IP's)
OS: Linux/Debian unstable.
Asterisk CVS-10/29/03-23:46:26
2008 Mar 17
6
Handling 3 different call ending causes
Hello List,
I'm using a dialstring like the one below. I want to have three different
things happening depending on exit cause.
Dial(SIP/${phonenumber},20,gL(20000[:5000][:5000]))
These 3 things could happen:
1, Caller hangs up
2, Callee hangs up
3, The 20 seconds is up and call is terminated from Asterisk.
Is there a way to separate these 3?
Thanks,
Best regards,
Tobias
--------------
2005 Jul 07
2
Extension Problems
Better read up on why a sip phone should register with asterisk. Do a 'sip show peers' and that will be the list of phones that can "receive" calls.
-------------------
I've double checked this. Everything is logging in fine, because I can
make calls, check my voicemail, everything except recieve calls on the
SIP devices.
David Phelan wrote: