Displaying 20 results from an estimated 4000 matches similar to: "T1 with Robbed Bit Signaling"
2010 Nov 07
7
Big practical systems
I don't want to start the "How many calls can Asterisk handle?" discussion
or "How many angels can stand on the point of a pin?" discussion either.
But can anyone contribute some practical knowledge of systems that take in
channel bank T1s or DS3s from "far away", and process the calls?
I am looking for real world, been there, done that, or "check the
2007 Apr 10
3
Learn some terminalogy before mounting this task.
All,
I have done research on VoIP for some time now. I'm a Cisco certified
Network Engineer however Telecom is not my strongest suit. I've been a
part of this mailing list for sometime but my delusions of grandeur in
migrating our 25 year old phone system to a new platform have been on
the back burner, until now. I have found my company is moving to a new
location(building) and this
2009 Oct 31
3
OT - Number Portability
Sorry for the off-topic, but perhaps this will be of interest to other
asterisk based ITSPs.
We are starting service in a rural area where the ILEC has the rural
"monopoly". From what we have read in the FCC docs this does NOT exempt
them from number portability, but what does it take for us to qualify to
receive their numbers? To date we simply have a few voice trunks to them,
2003 Jul 10
3
T1 config for robbed-bit E&M AMI
I have a couple of live T1s sitting around and they are not ISDN(like most
of the people that are using Asterisk seem to be using), they are regular
old 24 channel, robbed-bit, E&M wink start, D4AMI T1 circuits.
Can I get these T1s to work with a T100P Digium card and asterisk?
Searching through the lists and the documentation I haven't seen any
examples of how to configure this kind
2005 Mar 17
4
Caller ID on E&M Wink
I am an Asterisk newby, and I cannot seem to get Caller ID information
from our T1 line. When calls appear at the phones, they say the call
came from "asterisk" and unknown number.
I know how Caller ID information is passed on an analog phone line
(between the rings) but with a T1 line, I don't know technically how it
is done.
I don't see the caller's number in the
2005 Jun 13
2
T1 multiplexer (or ?) for failover in large installation
Hi,
Please forgive my terminology, still a bit new to T1s and such.
I'm looking for a way to have 5 T1s from a carrier terminate into some type
of box (multiplexer?), then be able to plug 7 asterisk servers into that box
(each with single port T1 card) and be able to have 2 * servers go down at
any given time and not actually have the carrier see that anything has happened.
Obviously if a *
2010 Nov 01
4
FW: Under heavy attack
Only 100? We had a single server over 300.
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Zeeshan Zakaria
Sent: Saturday, October 30, 2010 9:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Under heavy attack
My count has reached 100 for the day. The server serves doesn't serve
2010 Dec 22
2
Maximum E1 Ports on Asterisk ?
Hi All,
Just a little over thought. Sorry if someone already asked about this
before.
Is it possible to put all 16 Ports of E1 in One Asterisk Server ?
And if it's not possible is there any suggestion or alternative for me to
use more than 320 lines of outgoing calls on One Asterisk Server ?
Thanks
ZH
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2009 Mar 20
2
Looking for clues to this error message
[Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device
state of this queue member, SIP/3617001000, is still 'Not in Use' when it
probably should not be! Please check UPGRADE.txt for correct configuration
settings.
[Cary Fitch]
We are running 1.4.22 and this message popped up in console.
It could be causing our Queues announcement problem, because if all members
2009 Apr 02
4
FXO Ignore ring
Is there a way to program an FXO device to totally ignore incoming calls?
I want to put an FXO on a Fax line so that 911 calls can be sent via that
line, but all other activity on the line is between the Fax machine and the
phone company.
Perhaps munge the ring tone detect if nothing else?
Cary
2009 Feb 03
2
RBS T1 DID issue
Howdy,
New installation, trying to connect an RBS T1 with AMI/D4 framing and E&M
Wink. Using a Sangoma A102d and asterisk 1.4.22-2 on Centos5 (Trixbox
2.6.2.1).
Outbound calls work fine, but inbound calls fail to read the DID
information, and with debug set to 10 I get the following:
[Feb 2 19:40:23] DEBUG[25184] chan_zap.c: Monitor doohicky got event
Wink/Flash on channel 3
[Feb 2
2009 Nov 12
3
"POTS 4K linear codec"
I am not sure what the problems are and the reasons for the basic 64K modems
used in VOIP are. I understand the compressed codecs that get the bandwidth
down to 20-30 K. And perhaps the 64K units give much better potential audio
than you would get on a normal POTS line.
But, as phone circuits VOIP/SIP doesn't seem to perform as well as plane old
phones.
Multiple transcodings cause issues.
2009 Mar 20
3
Queues Announce help request.
I am trying to get a queue to do more than just play music and hold calls.
Specifically, making some "comforting" voice announcements would be nice.
Below is the queues.conf file relevant portions.
Member phone number is munged to protect the guilty.
We shouldn't need the announcement source info, but I have been trying
everything.
The problem is with the member busy, we get no
2004 Jul 23
6
Robbed Bit T1 Configuration
Is it possible to control the dialing format used when making a call from
asterisk separately from the inbound format? In my case I have a single
robbed-bit T1 configured for featd (to receive ANI and DNIS) on a TE410P.
This configuration works fine for inbound calls, however when an outbound
call is made on this same T1 asterisk dials outbound in the featd format. I
need asterisk to only dial
2010 May 26
3
"ring splash"
Something new to me. Recently installed a 1.4.30 box for a small office
with four POTS lines in a hunt (Digium TDM410P). Had the telco put a
"call forward" option on the main line of the hunt. They dial a feature
code from their desk phones (Polycom IP450) that results in forwarding the
main number to our VoIP service. This is all to let them "try out" our
dialtone
2009 May 07
4
Voicemail Alert
Can any one suggest a little code to either ring a cell phone when a new VM
message is recorded, or send a text message?
Basically outside sales people want to know they have a new message, but
don't want to be interrupted to take a forwarded call.
While a message by message notice would be nice, even just a single notice
on the first message would be an alert to call for messages.
2010 Jan 11
4
SIP over VPN -- no audio to other remote/VPN connected phones
Hello,
I am having a problem with my current SIP over VPN setup.
We have a server running asterisk at our office. All the phones in the
office are on the same network / local to this server. We also have two
employees with home offices using SIP phones over VPN to connect to the
asterisk server. These phones have no problem with calls to the phones
in the office, however there is no audio
2010 Nov 17
2
GSM and SS7 Questions
I have two questions for the group.
#1 - I'm looking to use some GSM SIM cards with my Asterisk PBX. Can
anyone recommend a gateway? I need about 10-15 SIM slots.
#2 - I'm also looking to connect Asterisk to an SS7 signaled DS1 (24
channels) for inbound and outbound voice calls. Can anyone offer any
suggestions for cards to use there?
2010 Jun 25
2
Big time system
We are an asterisk user... small time system 50-100 users or so.
But, we have an opportunity to get into a big time telecom activity.
It would have 2000 to 30,000 user lines per city, and we would like to have
those brought back to a central location for control and because transport
can be more economical than remote site rentals, maintenance and personnel.
We could take the local lines into
2009 Aug 31
4
How to stop IVR once system receives DTMF?
Hi,
We are trying to implement a complex business logic in Asterisk. Executing
"Wait_For_Digit" command after playing IVR. We want to stop the IVR once we
receive the digit. It is not recognizing the Digit until it completes the
IVR. How can we stop the IVR once we receive the digit?
Thanks
BB
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