Hi,
I input 0203# after mailbox? voice prompt from Voicemail cmd
on extensions.conf such as
exten = 0021,1,Ringing
exten = 0021,2,Wait(1)
exten = 0021,3,Voicemail
exten = 0021,4,Hangup
*CLI -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/0103-09a308b0, ) in
new stack
-- Executing [EMAIL
Ravi,
there is a easy way of creating all those commands in linux.
just run the following in a shell:
for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
permit udp host 192.168.5.0 eq $x any conduit permit udp host;done
This will create all your PIX rules at ones.
I think you
Hi,
My problem is in subject. As I read in documentations and
voip-info.org I can't user ChanIsAvalil because it not detects BUSY
information on SIP channel. I've tried to use SIPPEER function, but it
gives OK (9 ms) back on BUSY SIP channel. I use Asterisk 1.2.15, SIP
extensions are Linksys
- Original Message
From: Olivier [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, 9 February, 2008 6:55:15 AM
Subject: Re: [asterisk-users] External MWI question for Asterisk
Do you mean your
Hi,
I have installed asterisk real time and sip buddies information is being
stored in DB. Now I have a question,
Asterisk Realtime Server -A
Third party SIP server-B
Question: Is there any configuration in * RT that it can register with
defined sip user on Server-B
I was only able to find sip
Thank you for replying. The probleam is how do I use the
Asterisk_manager-API and implement them in my C code. Like how do I call a
API in my C program. It will be of great help if I can have an example.
By traffic I mean how much bandwidth or data transferring is taking place in
a call that is
Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy
hints to phones?
I'm not reporting this a s a bug because (although I have it working
with Asterisk 1.4.17, the hardware involved is different.
Thanks.
___
-- Bandwidth and
Hi, may be this question is a bit silly, but I couldn¹t find any document or
post or anything that say that if this is possible or not.
I want to show information on my phones cisco 7960/40 when a call arrive.
May be a bit more than a caller ID, show more detail level, if is that
possible.
I
Javier Temponi wrote:
Hi, may be this question is a bit silly, but I couldn’t find any
document or post or anything that say that if this is possible or not.
I want to show information on my phones cisco 7960/40 when a call
arrive. May be a bit more than a caller ID, show more detail level,
Soumya Kat wrote:
What I would like to know is how to get information such as SIP users,
number of SIP connections and traffic associated with those from asterisk
using a C Code.
Russell Bryant
There is actually no good way to do this inside of Asterisk right now.
It's
certainly all
I tried the following ACL command
access-list asterisk permit udp 0.0.0.0 192.168.5.0 range 1 2
and I got the following response back
[no] access-list id [line line-num] deny|permit icmp
sip smask | interface if_name | object-group
network_obj_grp_id
dip dmask |
try:
access-list asterisk permit udp any host x.x.x.x eq 1
- Ravichandran Rajagopal [EMAIL PROTECTED] wrote:
I tried the following ACL command
access-list asterisk permit udp 0.0.0.0 192.168.5.0 range 1
2
and I got the following response back
[no] access-list id [line
Adrian Marsh wrote:
In the Make menuselect, I noticed theres no .SLN voicefile selection for
the basic audiofiles - has SLN been depreciated?
No, the sln format is still supported. We have just never distributed any
files
in that raw format. Previously, we only had gsm recordings. For
Thomas Kenyon wrote:
Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy
hints to phones?
I'm not reporting this a s a bug because (although I have it working
with Asterisk 1.4.17, the hardware involved is different.
What type of device are you subscribing to, is it
Is anyone successfully running asterisk on an HP proliant while using
their management software, hpasm?
I have two DL360's and two TE220B's. The cards have their own IRQ's.
No matter what combination of settings I use, the cards fail the
patlooptest if hpasm (ver 7.9.1) is running. If I
Russell Bryant wrote:
Thomas Kenyon wrote:
Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy
hints to phones?
I'm not reporting this a s a bug because (although I have it working
with Asterisk 1.4.17, the hardware involved is different.
What type of device are you
I made the following changes and I am still facing one way audio with my call
flow.
-Original Message-
From: Wendell Hamilton [mailto:[EMAIL PROTECTED]
Sent: Saturday, February 09, 2008 1:58 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Joris
Dear List:
Can anyone refer me to a resource to better understand how the SIP protocol
is used by carriers to provide voice circuits between * and the PSTN?
Thanks a bunch!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Did you only open up the one port (1)? You need to open up a range, if
you're doing it this way, like 1-10020 and then set your rtp ports in
asterisk to the same range.
- Ravichandran Rajagopal [EMAIL PROTECTED] wrote:
I made the following changes and I am still facing one way
According to voip-info, the syntax for the VoiceMail command is as
follows...
VoiceMail([/flags/]/[EMAIL PROTECTED][EMAIL PROTECTED]boxnumber3]/)
If you check the syntax for the VoiceMail command, it indicates that the
mailbox parameter is /not/ optional, so I'm surprised this works at
all.
Why are you specifying the password and server IP in the dial string
when it's included in sip.conf? It's unnecessary.
I believe that Dial(SIP/gs102/1234) will achieve what you want.
ast guy wrote:
Hi,
I'm trying to call a SIP server while providing the SIP server
username/password in dial
In my efforts to solve a mystery of asterisk slowly loosing it's ability to
take incoming and outgoing calls I set asterisk to restart b-channels every
60 seconds hoping I would find something odd after some time.
So now I am looking at the CLI a few hours later and look what happens when
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