On Friday 24 October 2003 13:23, Adams, Gavin wrote:
So, I would like to be able to selectivity chop off any # characters
at the end of string, only if they exist. Basically as follows
(chopping off the leading '9' with ${EXTEN:1} syntax:
exten = _X.,1,Cut(newexten=EXTEN,#,1)
exten =
Rich Adamson wrote:
My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD server.
On a slower CPU linux system, Asterisk runs at 0.1% - both without any
active channels...
Any ideas, anyone recognizing the problem?
Is 'top' suggesting that * is actually consuming 98%?
Yes,
Did you have * working before the latest CVS updates of a few days ago?
Try:
disallow=all
allow=ulaw
allow=alaw
and see how that works for you. Put those lines into any SIP entries
in sip.conf to make doubly sure you've got all your permissions
straight. I have tested with my grandstream 102
I'm trying to set up * server behind NAT. The box
is set up as DMZ in my DSL router, i.e. all incoming connections without
explicit port mapping are forwarded to *. So far I'm unable to get this setup to
work for either IAX or SIP (tried IAXComm XLite softphones on public IP
address). Data
Peter Zeltins wrote:
I'm trying to set up * server behind NAT. The box is set up as DMZ in
my DSL router, i.e. all incoming connections without explicit port
mapping are forwarded to *. So far I'm unable to get this setup to
work for either IAX or SIP (tried IAXComm XLite softphones on public
Jan Janak wrote:
I experimented a little bit and Asterisk behind NAT with SIP works. I created
an account at iptel.org and use that account for outbound SIP traffic from
Asterisk.
I can confirm that Asterisk behind NAT can call out to IPtel.org
...and users connected to iptel.org can call me, if
Hi!
try to use in sip.conf :
register =x:[EMAIL PROTECTED]/xx
[iconnect]
type=friend
secret=
username=xxx
host=sipauth.deltathree.com
dtmfmode=inband
context=yourcontext
and in extensions.conf:
exten = _7X.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
This works for me
regards
Miklos
Hello..
Thanks for the reply.. I'll give this a check later today. Is the first
x in the register command your phone number at ICONNECTHERE? I am
using them with the demo account only as outbound so don't have a phone
number. Maybe this could be the problem.
Regards,
Robert
IAX should work well behind NAT without further configuration than just
a single port forward. RTP based protocols such as MGCP, SIP and H.323
require helper agents on the NAT box to work, although SIP can be forced
to work by slightly breaking it.
roy
On Mon, 2003-10-27 at 10:00, Peter Zeltins
Hi!
I don´t have an inbound number to, this registration is for an outbound
account
sorry if i don´t explain better in he first time
register=username:[EMAIL PROTECTED]/extension
hope this helps
miklos
- Original Message -
From: rnc Info Lists [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
hello!
How to turn off DTMF detection on modem (isdn) on active channel (when
the channel is open )?
Problem is that dtmf is detected when someone on (ISDN) telco side
speak then dtmf tones are send to from asterisk to internal line (x100p) .
tnx.
Tomaz
Just looking at the README.mysql file in the CVS..
It contains this line..
We will, where appropriate, make it available via a separate package
which will only be usable when Asterisk is used completely within GPL
(i.e. not in conjunction with G.729, OpenH.323, etc).
Does this mean that if
This also did not help.
I do not know what to do eals :(
I do not know what to try eals.
-- Bart
- Original Message -
From: Andrew Joakimsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, October 24, 2003 11:00 PM
Subject: RE: [Asterisk-Users] G729 stops asterisk in the
I guess the subject says it all.. :)
I am running the CVS from right now.. +- 12:25 GMT
MySQL CDR logging is installed and working..
Anyone got any ideas?
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[EMAIL PROTECTED]
hello,
can anybody help me with folloving problem
I have asterisk with the public IP and two UAs (snom100, x-lite) in the
same private network behind the same FW/NAT.
All is working good, but whan I tried to establish call between these
two UAs, first 10-15 second is nothing to hear and then is
Peter Hudec wrote:
hello,
can anybody help me with folloving problem
I have asterisk with the public IP and two UAs (snom100, x-lite) in the
same private network behind the same FW/NAT.
All is working good, but whan I tried to establish call between these
two UAs, first 10-15 second is nothing
My asterisk suddenly died in ast_verbose, called from app_dial...
leaving a core which told me the following:
(gdb) where
#0 0x4011a7c3 in chunk_free () from /lib/libc.so.6
#1 0x4011a548 in free () from /lib/libc.so.6
#2 0x080529fa in ast_verbose ()
#3 0x40493b76 in wait_for_answer
Hi!
MySQL CDR logging is installed and working..
Same question, same situation for me.
Can you connect via localhost socket for CDR? That didn't work for me
(on two machines), I need to use hostname port.
In voicemail.conf, however, there is no paramter to specify a port (or
socket), at
WipeOut wrote:
Peter Hudec wrote:
hello,
can anybody help me with folloving problem
I have asterisk with the public IP and two UAs (snom100, x-lite) in the
same private network behind the same FW/NAT.
All is working good, but whan I tried to establish call between these
two UAs, first 10-15
I experimented a little bit and Asterisk behind NAT with SIP works. I created
an account at iptel.org and use that account for outbound SIP traffic from
Asterisk.
I can confirm that Asterisk behind NAT can call out to IPtel.org
...and users connected to iptel.org can call me, if my server
Hi all (Michael),
how it is possible to get the ip address of the calling party ?
(i know by using h323... but there're a few unknown segfaults...) and so i
want to use oh323, but i have to get the ip from the caller to permit or
deny the call with AGI.
Is it possible at all ?
Thanks,
Thomas.
Philipp von Klitzing wrote:
Hi!
MySQL CDR logging is installed and working..
Same question, same situation for me.
Can you connect via localhost socket for CDR? That didn't work for me
(on two machines), I need to use hostname port.
My MySQL is on another server so I haven't tried
I am having the same problem. Here are my findings:
In asterisk/messages log file:
Oct 25 10:55:11 WARNING[19474]: File app_voicemail2.c, Line 2388
(vm_execmain): Couldn't read username
CLI debug output is as follows when accessing the VoiceMailMain2 from
extension 8500:
Executing
Hi!
You will probably have to use canreinvite=no in the UA definitions in
the SIP.conf for those two phones..
I have this so
In your case you want the opposite: canreinvite=yes
Philipp
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OK, I got IAX to work. Which SIP bone I should break to force it into
working? ;)
Peter
IAX should work well behind NAT without further configuration than just
a single port forward. RTP based protocols such as MGCP, SIP and H.323
require helper agents on the NAT box to work, although SIP can
On Mon, 2003-10-27 at 05:39, WipeOut wrote:
Just looking at the README.mysql file in the CVS..
It contains this line..
We will, where appropriate, make it available via a separate package
which will only be usable when Asterisk is used completely within GPL
(i.e. not in conjunction with
I've just set up a new asterisk server from CVS.
And without G729. And I still get the same problem.
So that mean that it is not G729. I think is something wrong with CVS.
Because it was working and right now wen i have newst CVS asterisk is
just rejecting incomming IAX calls.
What can be wrong?
From: Victor Medrano [EMAIL PROTECTED]
Download binary with java , works fine with 2000 + Xp
regards
Ok where can I find this binary with java?
Thank you!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven M.
Sokol
Sent: Friday, October 24, 2003
Hi list,
I have an asterisk box with 8 zap channels (E400P, only one span, EM
siginaling). And sometimes on the console, these messages apear about
some channels:
-- Starting simple switch on 'Zap/4-1'
-- Starting simple switch on 'Zap/5-1'
And then:
Philipp von Klitzing wrote:
You will probably have to use canreinvite=no in the UA definitions in
the SIP.conf for those two phones..
In your case you want the opposite: canreinvite=yes
A try to sort out these kind of opposite messages:
When asterisk connects two SIP phones, it tries to be in
Hi All,
I have a PBX on a PRI ISDN30 line on a particular project. I would like
to migrate to *.
Just as I can currently go between an analogue line and handset using
Line - X100 - * - TDM - handset so I can create a transparent
migration, I would like to do this with PRI.
Example: PRI - E100P
Please, help
I could not compile g723 format with pwlib-v1_4_11 and openh323-v1_11_7
I'am planning to use h323 channel driver, because of it that versions of
libraries have been cvs'ed from openh323.org
I am getting next compile errors:
gcc -pipe -Wall -Wstrict-prototypes
I am trying to achieve the same thing.
I have bothe asterisk and X-lite behind NAT.
sip uses port 5060
X-lite can be configured to use an rtp port, and you can specify your
external address...
y configured my nat to foward 5060 to my *
ports 8000 and 8001 too.
I also tried with the sip with no
You may have a file called dialplan.xml being TFTP'd to your phone. It has
a number of rules in it for helping the phone to determine when it has
complete number. It may need some tuning to bring it in line with what you
need.
I've found that the phone appears to treat the contents of the file
Can someone point me to the echo cancellation settings for a pure sip
setup?
Thanks,
Stephen
Subject: Re: [Asterisk-Users] ReplayTV connecting through Asterisk box
Has anyone had any luck getting a ReplayTV DVR box to connect
through an Asterisk box? Mine seems to dial just fine, but can't
A very strange problem..
* dies with seg fault when calling from SIP to
h323
WARNING[589848]: File chan_oh323.c, Line 2429
(alerted_h323_connection): Call with reference 183 in unexpected state
(4). -- Called 113506 -- H323:183
answered SIP/alex-1e48Segmentation fault
I'm using
Hi!
I am trying to achieve the same thing.
I have bothe asterisk and X-lite behind NAT.
Here (see below) comes some collected wisdom I took from reading this
list and searching its archive during the past 2 weeks or so.
sip uses port 5060
X-lite can be configured to use an rtp port, and
Hi,
Is there anyway to join a channel without bringing the other two SIP users into the conference ?? If so plz provide some sample.
TIA
Azher
Do you Yahoo!?
Exclusive Video Premiere - Britney Spears
--- Olle E. Johansson [EMAIL PROTECTED] wrote:
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk on FreeBSD
Date: Mon, 27 Oct 2003 08:24:22 +0100
Rich Adamson wrote:
My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD
Hi there,
up till now I had this two-box setup in mind:
* no.1: public IP
* no.2: private IP, registers with no.1, serves a small office with
clients behind NAT
See we'd get something like this:
SIP client (GSM) -- *1 -- IAX2 (iLBC) -- *2 -- G.711 -- MGCP UA
The codec of the SIP client (on
On Monday 27 October 2003 07:11, Philipp von Klitzing wrote:
MySQL CDR logging is installed and working..
Can you connect via localhost socket for CDR? That didn't work
for me (on two machines), I need to use hostname port.
In all likelihood, you have an authentication problem. Note that
Tilghman Lesher wrote:
On Monday 27 October 2003 07:11, Philipp von Klitzing wrote:
MySQL CDR logging is installed and working..
Can you connect via localhost socket for CDR? That didn't work
for me (on two machines), I need to use hostname port.
In all likelihood, you have an
Hi!
Can you connect via localhost socket for CDR? That didn't work
for me (on two machines), I need to use hostname port.
In all likelihood, you have an authentication problem. Note that
specifying '%' (or a name) for the hostname portion in your GRANT
does NOT match 'localhost'.
Olle E. Johansson [EMAIL PROTECTED] writes:
My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD
server. On a slower CPU linux system, Asterisk runs at 0.1% - both
without any active channels...
Any ideas, anyone recognizing the problem?
On the BSDs, your friend is ktrace (or
Searching the archives there has been some discussion
about the need for QOS routing on a mixed voice data
broadband like ADSL.
Has anyone run * on a production system with voice and
data.
Can anyone share what has to be done to secure the
voice and throttle back the data?
If a linux router is
On Monday 27 October 2003 12:21, Philipp von Klitzing wrote:
In voicemail.conf, however, there is no paramter to specify a
port (or socket), at least not from what I read here on the
list.
That is correct; it was never added. Given the licensing problem
with MySQL, it is not likely
On Mon, 2003-10-27 at 09:07, Stuart Mackintosh wrote:
Hi All,
I have a PBX on a PRI ISDN30 line on a particular project. I would like
to migrate to *.
Just as I can currently go between an analogue line and handset using
Line - X100 - * - TDM - handset so I can create a transparent
fred alexander [EMAIL PROTECTED] writes:
Can anyone share what has to be done to secure the
voice and throttle back the data?
Many routers allow you to prioritize certain types of traffic --
effectively letting the packets jump the queue. If you strictly
prioritize the voice packets over data
Ok, I got Gastman win32 running (no crashes so far). I entered some SIP
phone extensions in the GUI to represent my cisco 7960 phones using
extension SIP/311, but when I use the phone to make a call, a new icon
called SIP/311-fgeh, appears for the length of the call. Since the system
seems to
fred alexander wrote:
Searching the archives there has been some discussion
about the need for QOS routing on a mixed voice data
broadband like ADSL.
Has anyone run * on a production system with voice and
data.
Can anyone share what has to be done to secure the
voice and throttle back the data?
i am new to asterisk, and looking to develop an application using a
dialogic card. as far as i can tell, drivers for these cards are
available, but are not free. is that still true? if so, whom does one
contact about licensing?
thanks,
tad
___
You are right. I do not have mpg123 installed. Is it not included in
Asterisk build? I would appreciate it if you could give some instructions
on how to install this process.
Thank you in advance,
Kang
I am seeing strange behavior that I don't understand. Voicemail2 and
voicemailmain2 work fine if I call from a sip phone directly connected
to *, but if I call either of them from an analog line on the other side
of a sip gateway, voicemail seems to ignore digits. If I am recording a
message and
Hi,
I am new in VOIP area, so any help is really appreciated. I setup asterisk
at home and I am trying softphone.
I download SJphone from SJlabs and I can place calls. Question is, how can
I make a call to that softphone
What would be config in asterisk and in softphone. I am trying to use SIP.
Hi list!
I have a little question about groups and Asterisk ... is there anyone out there that
can say if Asterisk can do any of this;
We have a customer that want call handling we cant give him with a traditional PBX,
and I'm running Asterisk @home so I thought I could give it a try ...
The
WipeOut [EMAIL PROTECTED] writes:
You can't use QOS on the internet.. Its just not supported..
*IF* your ADSL router supports QOS it will only be effective on
outbaound traffic.. Inbound would still come in as it always has..
If your DSL link is the bottleneck, rather than earlier hops back
2) Transcoding: To be avoided at all times
Transcoding is the conversion of a voice stream with one codec to a voice
stream with another codec (e.g. G.729 to G.7.23). Transcoding
dramatically degrades the voice quality. It has to be avoided at all
times.
I really dont know what they have
Sounds to me like you are using inband DTMF, which won't work unless the
codec is ulaw or alaw. Use out of band DTMF aka rfc2833 or info.
On Mon, 2003-10-27 at 13:50, Steve Dolloff wrote:
I am seeing strange behavior that I don't understand. Voicemail2 and
voicemailmain2 work fine if I call
If a linux router is need can that run on the * box to
reduce cost?
All help is gratefully received, so I can plan a
multi-office rollout.
Fred
You can't use QOS on the internet.. Its just not supported..
*IF* your ADSL router supports QOS it will only be effective on
outbaound
Hey all..I'm looking to start with a single FXS card but with 3
extensions for VM purposes only. I'd like to know if there is a way that
you can have different stuttering dialtones depending on which extension
has a VM. For example If x103 and x104 have VM can there be a distinctly
different
-Original Message-
From: Perry E. Metzger [mailto:[EMAIL PROTECTED]
fred alexander [EMAIL PROTECTED] writes:
Can anyone share what has to be done to secure the
voice and throttle back the data?
Many routers allow you to prioritize certain types of traffic --
effectively
Yes, its true. Contact to [EMAIL PROTECTED]
- Original Message -
From: tad [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 4:41 PM
Subject: [Asterisk-Users] dialogic support
i am new to asterisk, and looking to develop an application using a
dialogic card. as
What kind of gateway are you using? Did you set dtmf-relay in that gateway?
Regards,
Gus
- Original Message -
From: Steve Dolloff [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 4:50 PM
Subject: [Asterisk-Users] passing digits for voicemail from sip gateway
I
CW_ASN - Gus wrote:
Anyway, in certanly implemetations you don't need CCS7 to connect to CO. You
always can connect with PRI... same speed and same functionalities to user
side. In fact, CCS7 is the support for ISDN-PRI avanced features. If you
could connect with Lucent 5ESS you can have a PRI
I would appreciate it if anyone can give me some instructions on how to
install mpg123.
Thanks in advance,
Kang
Phillip
Has anyone successfully used a Luxon VoIP gateway with *?
Thanks,
--Ernest
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Lars:
Anything you want is possible to do with Asterisk... the matter is how much
time you want to spend to build that applications... I think that is posible
to do that with AGI scripts...
Regards,
Gus
- Original Message -
From: Lars Fredriksson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Perry E. Metzger wrote:
Olle E. Johansson [EMAIL PROTECTED] writes:
My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD
server. On a slower CPU linux system, Asterisk runs at 0.1% - both
without any active channels...
Any ideas, anyone recognizing the problem?
On the BSDs, your
MPG123 is not included in Asterisk...
Download the package:
http://www.mpg123.de/cgi-bin/sitexplorer.cgi?/mpg123/precompiled/
Install using:
rpm -ivh mpg123-0.59q-1.i386.rpm
Copy the file mpg123 from /usr/local/bin to /usr/bin
That's all...
Please read the posts, this issue was treated
Has anybody tried running Asterisk on a SPARC based system? I'd imagine
drivers would be the major issue. Any info is appreciated.
Jerimiah
Tularosa Communications
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Has anyone made * to work with iconnnecthere's demo account?
Todd Wallace
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Hi!
where i can find info about using gastman and astman?
Thanks!
Miklos
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Adams, Gavin [EMAIL PROTECTED] writes:
For Cisco routers, look at the fair-queuing modes (but stay away from
weighted fair queuing as that can have a deleterious effect on VoIP
traffic).
Under Linux, check out http://lartc.org/ which deals with configuring
routing under Linux with traffic
Hi all,
Let me first describe to you our environment -
Asterisk Server (xx.jacksongrp.com)
---
in non-natted environment w/ public IP
Office 1 - Millersville, MD
---
- Natted environment
- Cisco 7960 telephone, registered with
The mpg123 homepage is at http://www.mpg123.de/ Either follow the
instructions there for downloading and building mpg123 or use whatever
installation tool your Linux distro uses.
On Mon, 2003-10-27 at 14:25, [EMAIL PROTECTED]
wrote:
I would appreciate it if anyone can give me some instructions
Hi,
I have two OpenLine4 boards, and would like to test with *.
But I see in vpb.conf that only V6PCI/V12PCI is mentioned. It means that
OpenLine4 does not work?
Thanks for your time
Jorge
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MPG123 is not included in Asterisk...
Download the package:
http://www.mpg123.de/cgi-bin/sitexplorer.cgi?/mpg123/precompiled/
Install using:
rpm -ivh mpg123-0.59q-1.i386.rpm
Copy the file mpg123 from /usr/local/bin to /usr/bin
You no longer need to copy it from /usr/local/bin to
From what I've heard and learned, SS7 appears to be a meta meta signalling
protocol.
First we had analog lines. Then ATT started grouping 24 analog lines to
form a T1. Inband signalling was used in each channel. Time studies
indicate that these channels can be more effectively used if the
It works in /usr/local/bin/ now also.
On Mon, 27 Oct 2003, CW_ASN - Gus wrote:
MPG123 is not included in Asterisk...
Download the package:
http://www.mpg123.de/cgi-bin/sitexplorer.cgi?/mpg123/precompiled/
Install using:
rpm -ivh mpg123-0.59q-1.i386.rpm
Copy the file mpg123 from
Some notes can be found at
http://www.oneunified.net/support/asterisk/index.html
Regards,
Ray Burkholder
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: October 27, 2003 15:25
To: [EMAIL PROTECTED]
Subject: Re:
Look in /etc/asterisk, for a zapata.conf file. Rename it, for now, since you don't
have a line card.
You probably only need these files to get started:
sip.conf (for voip phones)
extensions.conf (dialing rules, extensions)
asterisk.conf (basic this is where things are file)
voicemail.conf
Does anyone have any recommendations on implementing Answering Machine detection for
call generation programs?
What I would like is * to determine what picks up the other line (Answering Machine,
Voicemail, or Human) to determine which action to take. For example:
If * detects Answering
I would appreciate it if anyone can give me some instructions on how to
install mpg123.
One more time for those running RedHat v9 ... ;)
Well... we found the problem. Redhat guys replaced mpg123 for mpg321.
Asterisk only works with original
The VPB4 worked for me. The vpb.conf needs to be updated to reflect that vpb4
works.
On Mon, Oct 27, 2003 at 04:34:38PM -0500, Jorge Mendoza wrote:
Hi,
I have two OpenLine4 boards, and would like to test with *.
But I see in vpb.conf that only V6PCI/V12PCI is mentioned. It means that
Olle E. Johansson [EMAIL PROTECTED] writes:
On the BSDs, your friend is ktrace (or ktruss, depending on
flavor). It will tell you what system calls your process is executing
while it is doing this.
ktrace on FreeBSD generates a file filled with this:
And some signals caught here and
See mbranca's patch at:
http://bugs.digium.com/bug_view_page.php?bug_id=441
On Mon, 27 Oct 2003, WipeOut wrote:
I guess the subject says it all.. :)
I am running the CVS from right now.. +- 12:25 GMT
MySQL CDR logging is installed and working..
Anyone got any ideas?
On Mon, Oct 27, 2003 at 04:34:38PM -0500, Jorge Mendoza wrote:
Hi,
I have two OpenLine4 boards, and would like to test with *.
But I see in vpb.conf that only V6PCI/V12PCI is mentioned. It means that
OpenLine4 does not work?
The author of chan_vpb.c was expecting a patch from someone who
Hi,
I install * and is working fine. I have 3 FXO cards w/ 3 phone lines.
All the phones are SIP phones (Grandstream). The SIP phones from the same LAN
w/ Asterisk are working but on the external phones (from the Internet) I dont
have sound. All the Grandstream phones from the Internet
Hi,
A brief 6-step guide on how to hardcode a change in the Asterisk source that
will allow it to work from behind a nat device. I know its messy, but it
may prove useful to some people.
1. First punch a whole in your nat device. I just forwarded the port 5060
(for sip) and all ports between
Ray,
You are correct in regards to what SS7 is and does, although you can lose the
meta meta in your description.
What I was inquiring about was Gus' comment about a PRI treated as route on a 5E.
I'm also trying to find out what types of SS7/AIN features may be available over
a PRI D channel.
As there are still some of us that are grappling with the config files
and what the different things mean, could you possible post a working
example of this file with comments as to what the options are, what they
do, and why they are set the way they are.
Thank you,
Chris Tooley
On Mon,
Chris,
I install * and is working fine. I have 3 FXO cards w/ 3 phone lines. All
the phones are SIP phones (Grandstream). The SIP phones from
the same LAN w/ Asterisk are working but on the external phones (from the
Internet) I dont have sound. All the Grandstream phones
from the Internet
You are correct in regards to what SS7 is and does,
I thought it would be helpful to bring other users on the list up to speed.
;-)
Some additional SS7/VoIP integration info from a 3Com perspective can be
found at:
http://www.c7.com/ss7/whitepapers/3com_ss7_intelli.pdf
What I was
Hi,
I'm trying to put together an * system with extremely high up time.
The system as it stands now is 1 dual p4 with raid, redundant power, etc..
and a T100P card. I would like to get a second similar or identical box
with another T100P card. I have 1 PRI, how do I get the second box to take
Howdy again,
Now that I have my ATA-186 fixed, it seems my Cisco 7960 is dropping
it's registration. I can call from ATA-186 (both lines), to each
other. I can call from the Cisco 7960 to the ATA-186 (both lines.) I
cannot, however, dial from the ATA-186 (either lines) to the Cisco 7960
-
OpenSS7 project mentions Asterisk also. I think project will bring something
what we all really need - SS7 support for Asterisk
Take a look : www.openss7.org
Regards,
Alexander
Unofficial Asterisk Forums
URL :
look for a T1 failover switch.
(cheap as dirt on ebay, mine was $7.50, - yes really, I got the decimal in
the right spot, hard to find an empty rackmount box that is cheaper.)
Basically it looks like a Y in the T1. It contains a csu/dsu on each
interface. It decodes the t1 signals, and then
Jon -
Can you please post the name of the product and model number? Some
cursory searches don't find appropriate results on eBay or Google.
JT
look for a T1 failover switch.
(cheap as dirt on ebay, mine was $7.50, - yes really, I got the
decimal in the right spot, hard to find an empty
On Mon, Oct 27, 2003 at 08:12:05PM -0500, Jon Pounder wrote:
look for a T1 failover switch.
(cheap as dirt on ebay, mine was $7.50, - yes really, I got the decimal in
the right spot, hard to find an empty rackmount box that is cheaper.)
What is the manufacturer and model number. I searched
At 08:46 PM 10/27/2003, you wrote:
Jon -
Can you please post the name of the product and model number? Some
cursory searches don't find appropriate results on eBay or Google.
verilink-1558a_cg
The manufacturer was very helpful and sent me the software for it for free
as well.
I bought from
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