Displaying 20 results from an estimated 147 matches for "phonenumber".
2006 Feb 19
1
Cisco 7960 Register Problem
Hi all
I have a problem to register a cisco 7960 to an asterisk 1.2.2
I defined in sip.conf the next :
["phonenumber"]
type=friend
username="username"
secret="password"
host=dynamic
context=work
I am trying to catch the register requests with
sip debug
with no success (empty screen).
I can only catch the register messages with ngrep on host it's comming from.
#
U CISCO_IP:50339...
2006 Feb 13
0
Asterisk register ip phone
Hi all
I have a problem to register a cisco 7960 to an asterisk 1.2.2
I defined in sip.conf the next :
["phonenumber"]
type=friend
username="username"
secret="password"
host=dynamic
context=work
I am trying to catch the register requests with
sip debug
with no success (empty screen).
I can only catch the register messages with ngrep on host it's comming from.
#...
2006 Jan 17
2
change error messages for Validation helpers?
...ossible 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 than "Phonenumber
cannot be empty". Is that possible?
--Jeet
--
Posted via http://www.ruby-forum.com/.
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 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
2003 Oct 10
0
[Asterisk-User] Howto get the Caller Phonenumber ?
Hello,
Can anyone suggest us how to got the phonenumber of the caller.
In the environment variable, I just see the ip of the gateway.
Environment: 'agi_callerid' is 'XXX.xxx.XXX.xx'
Should I do some changes in the GW conf, or is it just not possible when
we got a call from PSTN ?!?
Thx in advance,
Ares
2006 May 09
1
Asterisk settings Net2Phone
...figure 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
host = sip.net2phone.com
username = PHONENUMBER
secret = PASSWORD
fromuser = PHONENUMBER
fromdomain = net2phone.com
context = incoming
insecure = very
canreinvite = no
*extensions.conf*
[outgoing]
exten => _9NXXNXXXXXX,1,Dial(SIP/net2phone/${E...
2006 Nov 13
1
Sending '#' with Dial
Hi!
I have a working asterisk-setup with four sip-clients. Everything works
great but when the users call someone the phonenumber shows up on the
receiving ends callerid-display.
To correct this my provider told me to send #31# before the phonenumber,
tried this with: Dial(SIP/#31#${EXTEN}@provider) but my asterisk tells me
that it isn't a valid extension.
The INVITE looks fine, '#31#<phonenumber>@provider&...
2009 Dec 23
1
AMI originate and PHP
Hi Guys,
I am trying to make a web form where a person is allowed to put in
$phoneNumber, $dialNumber, and $spoofNumber to make a call with spoof caller
ID. There are a few problems that I am facing with Asterisk AMI Originate
command. The reason why I want to use the darn AMI Originate is because I am
sending calls to mobile phones and I want to have some accountability and to
know if...
2004 Jun 01
2
BroadVoice usage?
...ry to call our assigned BroadVoice number, I immediately get
a BroadVoice message saying the number is busy.
I can provide the sip debug output but it basically shows that BroadVoice
appears to be not communicating with inbound or outbound requests.
Here are the entries:
sip.conf
register => phonenumber:password@sip.broadvoice.com:5060/phonenumber
[broadvoice]
type=friend
username=phonenumber
secret=password
host=proxy.broadvoice.com
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
extensions.conf
; calls via BroadVoice
exten => _6NXXXXXX,1,Dial,SIP/1925${EXTEN:1}@broadvoice
exten => _6...
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", "Zap/g1/[phonenumber]|60") in new
stack
-- Called g1/[phonenumber]
-- Zap/1-1 answered SIP/sipphone-9eb0
And then I get silence. The phone doesn't ring on the other end. I
have...
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 PhysicalAddress,
EmailAddress, or PhoneNumber.
I need to track the types of associations (i.e. home, work, cell, etc)
for each, so habtm definitely won''t cut it.
Do...
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
2005 May 06
1
CAPI on ptp with variable length digits in phonenumber: SOLUTION for EICON
...cht-----
Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von bladerunner
Gesendet: Freitag, 6. Mai 2005 14:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [Asterisk-Users] CAPI on ptp with variable length digits in phonenumber
hi again,
just ignore my mentioning of the sirrix-cards, just realised you have a PRI, i
overread it and thought you had a BRI. so i think your last hope is a
zaptel-card.
regards,
Am Freitag, 6. Mai 2005 13:04 schrieb Sebastian Buntin:
> Hi!
>
> we have a german PtP PRI connection h...
2005 Jul 26
2
Dial using URI(web) or using FORM(web)
Hello!
I have an Asterisk@home instalation with 7 users working OK, and I'ld like
to implement either a
-- Web dial feature, where the user would fill one form field with a phone
number and a connection would be created between his extention and the
entered number.
OR
-- Dial using an URI (callto:xxxxx link in a web page), having AstTapi
installed and configured in all workstations.
2005 Aug 27
1
SIP Registration failure
Hi list,
I'm in central-europe and signed yesterday a broadvoice account. My
Asterisk box is CVS 2005-08-25.
Problem I face is:
"Failed to authenticate on REGISTER to 'phonenumber@sip.broadvoice.com'
(Tries 2)" then
"Registration for 'phonenumber@sip.broadvoice.com' timed out" and finaly
"Giving up forever to register 'phonenumber@sip.broadvoice.com'"
If I do
keewi*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Por...
2009 Dec 18
2
To Asterisk AMI Gurus - Tacking issue with originate
Hello Everyone,
I am making a simple index.php file which will allow a web user to enter his
$phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged.
Following is the index.php and the contents of extensions_custom.conf. When
I submit the form nothing happens. I don't even see Manager Connected msg.
Your input will be much appreciated. I am thinking I have some syntax
2014 Aug 11
1
401 Unathorized
...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)
Using INVITE request as basis request - 41597440-0-320116780 at carrierIP
Found peer 'phonenumber' for 'phonenumber' from carrierIP:5060
<--- Reliably Transmitting (NAT) to carrierIP:5060 --->
SIP/2.0 401 Unauthorized
--------
Good call
--- (17 headers 14 lines) ---
Sending to carrierIP:5060 (no NAT)
Using INVITE request as basis request - 41604639-0-321360830 at carrierIP...
2008 Jan 18
1
Automatic call-out problem
...ith 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($outfile,"MaxRetries: 0\n");
fwrite($outfile,"RetryTime: 5\n");
fwrite($outfile,"WaitTime: 20\n");
fwrite($outfile,"Context: 0100q\n");
fwrite($outfile,"Callerid: $dbid\n");
fwrite($outfile,"Extension: $phonenumber\n");
fw...
2004 Jan 14
4
Multiple phonenumbers on one E1 PRI with Digium TE410P ?
Hi,
one short question: Is it possible for the zaptel driver to deal with
multiple phone numbers on one single E1 PRI line?
I could make my carrier route +49 xxx aaaaa-zzz and +49 xxx bbbbb-zzz
and others down one single PRI trunk to our asterisk box terminating in
a Digium TE410P.
Does the driver handle this and can I put calls coming in all on the
same physical interface put into