Yes, please post your script. I have a pile of Polycom 501 phones to
configure.
Yhanks!!!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ted Serreyn
Sent: Friday, July 15, 2005 2:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Most likely you are using PHP4, upgrade to PHP5.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of J P Edmund
Sent: Friday, June 24, 2005 4:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] CallerID name
/php.old, symlinked the
/usr/local/bin/php to /usr/bin, and did the same for pear, and it
worked.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
[mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia
|Sent: Friday, May 20, 2005
, 10 ); }
-Original Message-
From: Oswaldo Arratia [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 23, 2005 10:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] CallerID name lookup AGI script
Hi List,
I've managed to install this great
)
{ $formatted = 1 . $number; }
else
{ $formatted = $number; }
To
elseif( strlen( $number ) == 10 )
{ $formatted = $number; }
else
{ $formatted = substr( $number, 1, 10 ); }
-Original Message-
From: Oswaldo Arratia [mailto:[EMAIL PROTECTED]
Sent: Thursday
:[EMAIL PROTECTED] On Behalf Of Oswaldo
|Arratia
|Sent: Friday, June 17, 2005 7:51 AM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] 2nd Dialtone after answer
|
|
|Hi
|I am trying to achive this for a specific need of a customer.
|
|He has a DID
I bought a Dell SC1425 and installed a T1/E1 card from Digium and I tried to
configure it using [EMAIL PROTECTED] scripts and did not work, so I went the
long way and
configure with zaptel's instructions and voila! It works like a charm.
Oswaldo
-Original Message-
From: [EMAIL
Hi
I am trying to achive this for a specific need of a customer.
He has a DID pointed to an Asterisk server, I need to provide him dialtone
when the calls hits the server. How can I achieve this?
Let's say something like this:
Exten = s,1,Answer
Exten = s,2, Provide Dial tone
Exten = s,3, Dial
Hi List, I posted this question before but nobody replied. Maybe this time
there is somebody that can help me out.
I have a server with Digium Card with FXO ports.
I would like to add a fixed 30 seconds delay before it shows the calls as
answered.
If I call out and somebody picks up the phone it
the
source code...
- Joshua Colp.
(file in #asterisk on Freenode)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Tuesday, June 07, 2005 7:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Answer
stuff
to compensate with adding 30 seconds to the answered time. Instantly
changing CDR records isn't exactly what Asterisk was made to do easily.
- Joshua Colp.
(file in #asterisk on Freenode)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Hi
Has anyone made a Cisco 7960 work with the 7914 expansion module? I'm just
trying to use the expansion module as additional Speed Dial buttons, I do
not care if the LED show of the line if busy or not, I only want to appear
the name and when you press the button it calls the configured
, June 06, 2005 3:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco 7960 + 7914 using SIP
Oswaldo Arratia wrote:
Hi
Has anyone made a Cisco 7960 work with the 7914 expansion module? I'm
just trying to use the expansion module as additional
Hi List
I bought 1 Dell SC1425 server and 1 Digium TE110P T1/E1 card.
I installed Asterisk from aah 1.0
In the CLI I type 'genzaptelconf -svd' as I have done with other servers and
FXO cards to detect and configure the cards; this time it is not recognizing
the T1 card.
Any ideas why this might
admin/password is the login info
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler
Sent: Wednesday, June 01, 2005 6:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] 1.1b1 has been
Hi there,
I am trying to the the cid_rewrite.php script but if I run it from the
directly I get this error:
./cid_rewrite.php
br /
bParse error/b: parse error, expecting `T_OLD_FUNCTION' or `T_FUNCTION'
or `T_VAR' or `'}'' in b/var/lib/asterisk
/agi-bin/astlib_jm.php/b on line b73/bbr /
br /
: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Oswaldo Arratia
|Sent: Friday, May 20, 2005 11:34 AM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] CallerID name lookup AGI script
|
|Hi there,
|
|I am trying
Hi,
I've been trying to find out how to add a delay before the Zap channel
reports a call as answered.
Here is my problem,
I send a call to an Asterisk server that has a FXO card, * receives the
call, modifies the number to dial out, opens the Zap line to dial and
dials.. It works. The problem
I apologize for this off-topic e-mail but I don't seem to find an answer
anywhere else and I know there are experts here in this field.
I need to provide POE to 14 Cisco IP Phone 7960 and 2 Polycom Soundstation
IP 4000.
Which POE hub/switch can I get so it's compatible with cisco and polycom and
,
but they are not necessary for 0-133 ft distances. The only thing I see is
that you may have to change some of the pin outs for the RJ45 ends. Is this
a single T1?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, May 13, 2005 7:37 PM
To: 'Asterisk
Hi,
A customer has a Avaya PBX and is looking to migrate to Asterisk, they have
a T1 from the telco going into a CSU and then from the CSU to the Avaya PBX.
They will buy a Digium T1 card for the Asterisk, can this T1 coming from the
Telco plug directly into the Digium card or do they still need
Thank you very much for your replies!!!
As always, this list have been very helpful!
O.A.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Loftis
Sent: Friday, May 13, 2005 8:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
No, this is a T1 from the bell company
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Walker
Sent: Friday, May 13, 2005 9:52 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] PBX replacement
Are you
replacement
Is the T1 coming into the Avaya dialer for outbound or inbound calls?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, May 13, 2005 6:57 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE
- Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT
Try setting externip=(asterisk public ip address)
Hth
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 12:56 AM
To: 'Asterisk
-Users] Asterisk behind NAT
Can you show your outbound peer configuration? If you are registering,
please include that as well.
Thanks
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 9:44 AM
To: 'Asterisk
] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 9:44 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT
I have... Externip=x.x.x.xand nothing... Does not seem to help in
anything. Still my provider sees the private IP
Sent: Friday, April 15, 2005 2:01 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk behind NAT
On 13:45, Fri 15 Apr 05, Oswaldo Arratia wrote:
[gw2]
type=peer
port=5060
host=2.4.6.8
disallow=all
defaultip=2.4.6.8
allow=g729
Hi,
Put this line
Hi List,
I've spent hours researching on this topic, found tons of info, so far it
doesn't work yet.
Here's the scenario
Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to
send calls to an outside provider.
My SIP phones (outside * NAT) are able to register with no
Hi List,
I installed [EMAIL PROTECTED] 0.8 and no problem. Now and I am trying to install
oh323 and I am running in a problem making asterisk-oh323-0.7.1 (I also
tried 0.6.5 with the same result)
Here the list of the files I downloaded and the way I installed them
Untar the files
#tar zxvf
Here is what should work for you.
In your Cisco
dial-peer voice x voip
huntstop
destination-pattern x - Extension number you want to dial
progress_ind setup enable 3
session protocol sipv2
session target ipv4:y.y.y.y - Your * IP
session transport udp
dtmf-relay rtp-nte
codec
Did you solve your problem? I have the same setup and it works for me.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Rothschild
Sent: Saturday, March 19, 2005 5:49 PM
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Subject:
Hi List
I've been using Asterisk for quite some time with no major problems, but
I've been facing this bug from the beginning and now I want to see if that
is fixable.
We have a provider who terminates our USA LD traffic and the problem comes
when relaying the caller ID I send them from my
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Petrisko
Sent: Thursday, March 17, 2005 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Caller ID problem
On Thu, Mar 17, 2005 at 11:59:06AM -0500, Oswaldo Arratia wrote:
I send a call with valid
-Commercial Discussion
Subject: Re: [Asterisk-Users] Caller ID problem
On Thu, Mar 17, 2005 at 11:59:06AM -0500, Oswaldo Arratia wrote:
I send a call with valid caller ID info (areacode+number); my provider
gets the call and routes it properly, the end receiver gets the call
and does not see
I don't know if it is possible doing it directly on voicemail.conf... The
way I'd do it is creating a distribution list since I manage my own e-mail
server and putting the distribution list's e-mail address in voicemail.conf.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi,
Is there anybody out there that can e-mail me the following
files?
Get oh323 fromhttp://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gzGet
pwlib
Hi,
Thanks to Andrew Kochetkoff for sending Asterisk-oh323 files while
inaccessnetworks web page was down.
Now, I have a problem when compiling Asterisk-oh323 versions 0.7.0 or 0.7.1.
I get the following error:
/usr/include/_G_config.h:52: confused by earlier errors, bailing out
make[1]: ***
Has anyone figured out how to make a Sipura to dial an extension
automatically as soon as you pick the the handset?
I need to make all my users go thorugh a menu to place a call. Users should
not be able to dial directly, only through the menu.
Any ideas?
O.A.
Hi,
I had the same problem and I fixed it by modifying the SigTimer. I made it
SigTimer:0x03C00064 in the phone's configuration file.
What happens is that ForwardToVMDelay value has no effect if VoiceMailNumber
is not provisioned OR the value is 0 or greater than the ring timeout value
(see
ForwardToVMDelay=20s and since you raised the No Answer Timeout to 60s
this does not happen. Right?
BR,
Alen
Oswaldo Arratia wrote:
Hi,
I had the same problem and I fixed it by modifying the SigTimer. I
made it
SigTimer:0x03C00064 in the phone's configuration file.
What happens
Cisco uses firmware for IP phones.
And the phone models that can do SIP are 7905, 7912, 7940 and 7960.
7902 can only use SCCP, but you can use SCCP with * with basic
functionality.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher L.
Wade
Hi there,
I've been running * for some time now and thanks God no problem so far,
everything is configured using text files, and I'd like to move everything
to realtime database configuration to ease management using a GUI
application.
I've read about Realtime function of * and I see something
What type of equipment does your DID provider have?
I had the same problem with Cisco and solved it by adding progress_ind
setup enable 3 on the voip peer.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson
Sent: Tuesday, January 04, 2005
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