An ordinary T1 (non-ISDN) doen't have a separate channel for signalling.
See: http://www.voip-info.org/wiki-T1
ISDN T1's have a separate signalling channel.
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: Dean Collins [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
Yes in some cases you would if the 800 number restricts the call via a
geographic area. But you are in no way REQUIRED to set a callerid before
you place a call.
bkw
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, July 10, 2004 10:21 PM
Subject: Re:
Yep I sure did, damn upstream pipe gets so congested I had to drop it to
about 75% to keep from dropping packets... Seems to be working excellently,
I tried downloading a large file and doing some interactive SSH with no no
noticeable degradation... I'd say we have a winner. Installing and running
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Shaw
Sent: 11 July 2004 08:35
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IVR Menu and VoiceMail quality
Yep I sure did, damn upstream pipe gets so congested I had to drop it to
about 75% to keep
When I call a SIP user, the phone should ring in more
(Bthan one
(Bextentions. Also more than one phone should be able to
(Bregister with
(Basterisk. Right now it is not the case.
(B
(BThere is no issue here. You seem to be confused, that's
(Ball.
(B
(BA SIP account is a SIP account and
I accept your views.
(B
(BI have a specific requirements, can you help to attain the same.
(BIn our business we have 25 employees handling customer service.
(B
(BI want to add or remove employees in the customer service so does the
(Bdevices connected to it.
(BI don't want to make any
I was going to keep out of this (was interesting to read, as I have dealt
(Bwith simmillar situation) however I would like to add just this one commnet.
(B
(BTry to better understand asterisk than to throw about your money. What you
(Bwant to do is perfectly possible with asterisk there is no
I explained him a sample need.
(BI don't think asterisk does whatever i want in sip. It is an good PBX.
(B
(BPlease help me to understand. Anywhere am I wrong ? Or as you say is that
(BSIP feature is written?
(B
(B-Kannaiyan.
(B
(B
(B- Original Message -
(BFrom: "usedcanon"
Hi,
we intend to use innovaphone's IP200 full standard compliant H.323 hardware phones
(great!) with asterisk.
Everything works perfect (calls, signalling, talking) until we want to perform a call
transfer (H.450).
Upon pressing the R-key the first connection is hung up immediately. As
Well, there was a problem with my Fedora installation.
Thanks to everyone.
Xavier Olivella i Rigol
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
The patch is actually in plain text at the end of the post. You can
either scrape it off of the page or just get them from here...
http://sremington.zapto.org/weblog/2004-07-04_14.52.21.html
Then you will want to apply the patch in /usr/src/asterisk/contrib. If
you have a problem with the
Daniel Jimenez wrote:
http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry
From the WIKI:
Contributions
Manager: Daniel Jimenez (cuban)
Bounty: $50 USD
Date opened: July 10, 2004
Contributors: cuban ($50)
Detail
Yes, Yes I know you could do all
Please let me know if you find a solution, I'll do the same. I received
this from the FISPA list, and have ordered a 6400R which should arrive
Friday for testing.
*** begin paste ***
If you are using asterisk and digium cards, they are very sensitive to
CPU/IRQ speed. We had this issue until we
You was correct, I just upgraded to the lastest CVS and the g.726 codec is
working.
But, when I use the ATA-186 the call works but I received the NOTICE below,
with sipura all is ok.
-- Executing Dial(SIP/2007-ca45, Zap/1/2132979) in new stack
-- Called 1/2132979
-- Zap/1-1 answered
I love the Adtran 850T and it does have echo cancellation built into it.
I've used them for years, and now for a month or two w/ asterisk. No
echo at all, ever, period.
W. Kevin Hunt
CCIE #11841
MCSE, Linux+ SME
www.huntbrothers.com
-Original Message-
From: [EMAIL PROTECTED]
Paul is right. For voice, there are two varities of T1's. The straight
channelized T1 provides 24 pots lines and no custom services, like caller
ID. The other is a PRI ISDN, which is 23 voice and 1 'D' channel. The
telco can now provide more services, including caller id over the D channel,
Caller-ID can be sent over Channelized T1's, as well as a few of the
other class-features (stutter dial tone, etc...)
Setting outbound ANI is something that can't be done on Channelized T1's
which may be what you were referring to.
The csu's you reference are usually called Drop and Insert
[EMAIL PROTECTED] wrote:
You was correct, I just upgraded to the lastest CVS and the g.726
codec is working.
But, when I use the ATA-186 the call works but I received the NOTICE
below, with sipura all is ok.
-- Executing Dial(SIP/2007-ca45, Zap/1/2132979) in new stack
-- Called
hallo all,
i am new to asterisk, just started my first tests and tried to build up a
phoneserver with serveral iaxcomm clients running over iax2.
my question is:
now, it is possible to reach all phones over the asterisk server, but the
media stream is always routed through the server.
is it
You are confused about what a SIP session is and what a SIP session does.
SIP, session initiation protocol, controls an RTP, real time protocol,
session between two IP endpionts. The end points have to have unique IP
addresses for the session to run. The unique SIP registration is how * finds
a
hi,
yes this is ok .. when you have dial,Zap/g1 this problem is solved,
becouse you have many channels in some group.
But my problem is on x100p card so only one channel and is not busy with
nothing and i can't dial out .. even if i ring line connected to x100p
asterisk does not report
Searched the wiki and googled extensively to no avail.
How can the Directory App be used by a VoiceMailMain user to forward a
voice mail to another voicemail user and do it by a directory 'dial by
name' lookup?
Ex. A voicemail system with 1000+ subscribers. One user wants to
forward another
I have a lot of PC bits and pieces that I no longer need. I am moving to new
premisses soon so I thought I offer them up for sale. Usually I would sell
stuff through eBay but I dont really have much time for that right now.
Anyway what I would like to do is offer it here and donate ½ of the
On Sat, Jun 26, 2004 at 07:57:21AM +0900, Isamar Maia wrote:
I installed a TDM04b and a TDM40b with aggressive echo suppression
and it's working almost perfectly.
The problem is that all extensions are fax machines and people uses it for
both purposes, voice and fax. AFAIK, I cannot use
Thank you, now all is working good.
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RE: ATA 186, firmware SIP 3.1 and codec g.726
Date: Sun, 11 Jul 2004 15:42:02 +0100
Reply-To: [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
You was correct, I just
Paul Mahler wrote:
If you want to be able to more easily recognize what extension the traffic
if for, you can add additional extensions to the 7960. For example, if you
have two staff the admin monitors, add two additional extensions to the
7960. The admin can tell who is being called by the
Soren Rathje wrote:
Eh... Sort of like shadow lines ???
Remember that Dial(SIP/1 H323/1 ZAP/1,[timeout],[options],[URL]) will dial all 3
extensions simultaneously (regardless of channel choice) and with a little tinkering in your dialplan you can even
activate/deactivate this from the Manager
Kannaiyan Natesan wrote:
I hope this helps.
Since I feel this is a great feature, I will topup up to $100/-
-.Kannaiyan
http://www.goods2world.com -- Your Only VoIP
Thank you, I updated the wiki with your $25 addition.
--
Daniel Jimenez djimenez[at]pobox[dot]com
start it with asterisk -vvvgc
bkw
- Original Message -
From: Arjan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 11, 2004 12:27 PM
Subject: [Asterisk-Users] Asterisk on FreeBSD 4.10 dies
Hi All,
I'm pretty green to Asterisk. I'm trying to work towards a basic setup
Paul Mahler wrote:
If I have the requirement right, you could accomplish this by ringing the
staff extension and the admin extension at the same time. The Dial command
allows you to ring multiple extensions simultaneously.
Paul
Did you even read the bounty?
Yes, Yes I know you could do all sorts
Hello All,
Just a very simple example. I'm trying to make a call to
a busy phone number using Dial application.
-- H.323 call to 12345 with codec ALAW
-- Called 12345
-- OH323/L5663 is ringing
-- H.323 call 'ip$localhost/5663' cleared, reason 18
(Remote endpoint is
On 11/07/2004 at 08:42 Paul Mahler wrote:
You are confused about what a SIP session is and what a SIP session does.
SIP, session initiation protocol, controls an RTP, real time protocol,
session between two IP endpionts. The end points have to have unique IP
addresses for the session to run.
As Daniel Says, Bounty stands.
I cannot explain to you anymore. I'm sorry.
Please read more what SIP can do with SER.
-Kannaiyan.
- Original Message -
From: Paul Mahler [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 11, 2004 4:42 PM
Subject: RE: [Asterisk-Users] New
Does asterisk provide quality of service(QoS)? If it does, how do I use
it? The reason why I ask is that I need to switch to use POTS should the
internet connection becomes poor?
Thanks,
Jim
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I use an extra call to do this.
Write an .call file for example and let it connect through a local
extension to these 2 contexts. the macro's simply issue a Monitor
Command and do some management stuff. so it seems that you can't
really monitor a meetme conference.
this solution just adds another
Does asterisk provide quality of service(QoS)? If it does, how do I use
it? The reason why I ask is that I need to switch to use POTS should the
internet connection becomes poor?
Asterisk 'participates' in the qos process by allowing you to set TOS
bits in the IP header. For example:
In
The whole point of a SIP registration is to identify a UNIQUE device. You
CAN'T HAVE multiple devices registered as the same SIP device. That's WHY
the last device that registers gets the traffic.
This doesn't have ANYTHING TO DO WITH ASTERISK. This is a SIP issue, not an
Asterisk issue. You
It's not what SIP does with SER, it's what SER does with SIP.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I'm really glad you found it useful, I'm just trying to do my part since you
guys have been very helpful to me. Asterisk has been soo cool, I just want
to make sure everyone thinks so :)
-Chris
- Original Message -
From: usedcanon [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday,
This is a very complex question.
First, you have to ask about VoIP and QoS. This is because * uses VoIP
protocols like UDP and RTP. In general, the QoS of VoIP is not as high as
with the PSTN. Even so, call quality can be generally very good.
Second, * does support features that support QoS,
Daniel Jimenez wrote:
Soren Rathje wrote:
Eh... Sort of like shadow lines ???
Remember that Dial(SIP/1 H323/1
ZAP/1,[timeout],[options],[URL]) will dial all 3 extensions
simultaneously (regardless of channel choice) and with a little
tinkering in your dialplan you can even
Did you even read the RFC? Section 10.2.1 clearly talks about adding
multiple bindings to the same address-of record.
On Sun, 2004-07-11 at 12:31, Paul Mahler wrote:
The whole point of a SIP registration is to identify a UNIQUE device. You
CAN'T HAVE multiple devices registered as the same
On 11/07/2004 at 12:31 Paul Mahler wrote:
The whole point of a SIP registration is to identify a UNIQUE device. You
CAN'T HAVE multiple devices registered as the same SIP device. That's WHY
the last device that registers gets the traffic.
WRONG!
This doesn't have ANYTHING TO DO WITH ASTERISK.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arjan
On Sun, 11 Jul 2004 at 15:39 -0500, Dr. Rich Murphey wrote:
You might check login class in login.conf for the user that invokes
asterisk. Setting cputime=unlimited may help.
This
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arjan
Hi Brian,
On Sun, 11 Jul 2004 at 12:38 -0500, Brian K. West wrote:
start it with asterisk -vvvgc
Been there, done that. Nothing, no coredump. A lot of verbose
output while starting
On Sat, Jul 10, 2004 at 05:55:21PM +0100, Kevin Walsh wrote:
Richard Airlie [EMAIL PROTECTED] wrote:
First things first. Scrap the ports and build from the latest
CVS source. 0.9 is far to old and buggy, and suspect the same of
the Zaptel driver you have, although I don't use *BSD myself.
I'm toying with adding a feature request to provide some sort of
gain setting for voicemail when accessed from certain interfaces.
Maybe something like voicemail=6.0 (db) within a specific channel
section of zapata.conf corresponding to a pstn line.
Situation:
1. Someone calls into asterisk and
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello
On 12/07/2004, at 4:24 AM, Arjan wrote:
43676 root63 0 10244K 7628K RUN 2:44 99.05% 99.02%
asterisk
This is covered in the asterisk FreeBSD section:
http://www.voip-info.org/tiki-index.php?page=Asterisk+FreeBSD
extract:
CPU 99.9 %
Let's see you have 11.2 db of loss from the phone you are using to call in
on and the FXO interface on Asterisk. Retreiving voice mail would add
another 11.2 or a total of 22.4 db. But your measured tone level was 36db.
In other words the FXO interface and Asterisk introduced about 14 db of
Mike Machado wrote:
On Sun, 2004-07-11 at 12:31, Paul Mahler wrote:
The whole point of a SIP registration is to identify a UNIQUE device. You
CAN'T HAVE multiple devices registered as the same SIP device. That's WHY
the last device that registers gets the traffic.
This doesn't have ANYTHING
Both the question and the answer are not talking about QoS.
From the Q, qos does not provide a measure of quality, it provides a system to
allow you to request your data be handled according to priorities.
From the A, qos is confused with the pstn.. qos is a feature of IP, that has
nothing to
Hello guys,
I need your help related to a mediatrix 1204
configuration. I readsome of the messages that you posted in the asterisk
users mailing list about the mediatrix 1204 and decided to contact you. I know
thatthecommunity is not related to Mediatrix devices, but so far I
have not
At Astricon, I plan to cover QoS on FreeBSD using the pf
firewall's class based queuing. This includes implementing
classes to prioritize each of RTP, IAX, SIP, FTP, and others.
Within each class packets can be prioritized based on whether
TOS is set.
I'm wondering whether this should be a
Well, the question may not have been about QoS, but my answer certainly was.
QoS is defined as The performance specification of a communications channel
or system. (188) Note: QOS may be quantitatively indicated by channel or
system performance parameters, such as signal-to-noise ratio (S/N), bit
QoS is most certainly an issue when making the decision to move off the
PSTN. Is the performance of your VoIP system going to be comparable to the
performance of your PSTN system? Sounds like a reasonble question to me.
Not trying to get in the middle of whatever argument you're trying to
Paul Mahler wrote:
You should spend your money on getting a copy of each of the
two books that are now available and learn *. Then it will be clear to you
that you don't really want what you are asking for.
Shameless plug? It's offical you are trolling.
--
Daniel Jimenez
Good catch. I've got two pages full of test results for various items
and copied values from the wrong page. Regardless, it's still an issue
of very low volumes when voicemail involves creation and access from
a pstn location.
Let's see you have 11.2 db of loss from
OK... so I'm not sure what I'm looking at. I've had the good old echo
problems on my Rev C FXO again this morning, so I thought I'd attempt
some debugging, though I'm not sure what I'm looking at.
This call has echo.
Channel: 2
File Descriptor: 20
Span: 1I
Extension:
Dialing: no
Context:
Please forgive me for sending that last message to the wrong list. It was
supposed to go to the Dev list.
Sorry,
Steven
Steven Sokol
Owner/Manager
Sokol Associates, LLC
Phone: 816.822.1807
IaxTel: 700.613.9004
Web:http://www.sokol-associates.com
Hi again,
I can now say that the info below has nothing to do with the echo
issues.
I'm noticing a new connection too - it seems that numbers are being
left out of the dial at times. It's on those calls especially, that I
see echo. (The exchange answers and tells me the number isn't correct
Well, this is certainly getting exciting.
Andy, I took your advice and re-read the RFP. Andy--I don't think you are a
good candidate for a beginner's book on *, but if you send my your address,
I'll send you a copy on me. :-)
So, gentlemen, help me out here. The spec says:
The Address of
On 11/07/2004 at 20:00 Steven Sokol wrote:
Please forgive me for sending that last message to the wrong list. It was
supposed to go to the Dev list.
Sorry,
Steven
LOL, for me at least - this message arrived before whatever message you
accidentally sent...
:D
Andy
Andrew,
What type of motherboard/system are you running?
Hi again,
I can now say that the info below has nothing to do with the echo
issues.
I'm noticing a new connection too - it seems that numbers are being
left out of the dial at times. It's on those calls
On Fri, 09 Jul 2004 13:58:30 +1000
Dear Gonzalo Servat,
I'm successfully using your wake-up script, but found 1 problem. Other than that it
works perfectly good. Thanks man. ^_^
anyway, my problem seems to be the timezone or date problem.
I'm using time zone WIT/JAV,
But when I run the wake up
At the moment, it's running Fedora Core 1, with the 2.4.22-1.2197.nptl
kernel.
My Asterisk hardware is a TDM04B.
The motherboard in this machine is a Gigabyte 8S661FXM-RZ. I had
similar problems on an Intel D865GLCL board. Both machines are running
256MB DDR, and using Celeron 2.4/2.6GHz
At 5:00 PM -0600 on 7/11/04, Rich Adamson wrote:
I'm toying with adding a feature request to provide some sort of
gain setting for voicemail when accessed from certain interfaces.
Maybe something like voicemail=6.0 (db) within a specific channel
section of zapata.conf corresponding to a pstn line.
Hi All,
I'm using DISA on my * server to avoid overseas toll charges when
making calls to Western Europe from my cell phone. I have DISA working
with a DID from a VoicePulse Connect account. The outgoing call to
Europe is also made via Voicepulse Connect.
I see that the IAX media path is
notransfer=yes
bkw
- Original Message -
From: Michael Graves [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 11, 2004 9:09 PM
Subject: [Asterisk-Users] Stopping reinvite with IAX2?
Hi All,
I'm using DISA on my * server to avoid overseas toll charges when
making calls
i'm having problems with s100u (wcusb) on Via EPIA-V-8000
running RedHat Linux 9 (2.4.20-31.9). when i try to dial,
asterisk won't respond (still hear dialtone) or get a fast busy.
seldom asterisk would process the dialed number. i was able to
dial to echo test and i noticed that the line is
per peer
bkw
- Original Message -
From: Michael Graves [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 11, 2004 9:25 PM
Subject: Re: [Asterisk-Users] Stopping reinvite with IAX2?
Is this set on a per peer basis, or in the general section?
Michael
On Sun, 11 Jul 2004
Here's an update on my progress for all who are interested.
After carrying out many more hours of testing, the only thing that made
a significant difference was changing the mainboard/CPU of my asterisk
server.
My original Asterisk server was a Celeron 533 with 128mb ram. Now, keep
in mind, even
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