We're currently running a PSTN - SIP gateway with Asterisk. We also run
IAX/SIP - PSTN.
We have performed a test where the call is routed
UK PSTN - Digium E1 card - Asterisk GW - SIP G.711 - FWD - X-Ten
softphone
There is no echo at the softphone end, but severe echo on the PSTN
Hi,
I don't know whether only we are experiencing this
problem but it seems that if authentication is
used on a couple of phones, and then the authentication is removed (i.e. remove
the secret parameter from each of the extensions), then this isn't reflected in
asterisk after a reload.
My musiconhold.conf is as below:
;
; Music on hold class definitions
;
[classes]
default = quietmp3:/var/lib/asterisk/mohmp3
loud = mp3:/var/lib/asterisk/mohmp3
random = quietmp3:/var/lib/asterisk/mohmp3,-z
This has been copied from the working system as has the mp3 file into
This is a recent CVS checkout and show version reports Asterisk
CVS-07/19/03-22:42:04
What's alsa. I have not come across that yet.
Stuart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: 21 July 2003 02:46
To: [EMAIL PROTECTED]
Hello all,
I am a newbie to this list - and so far very impressed with the
functionality of Asterisk. So far I have setup a simple soft phone
running on a windows PC making calls to other SIP soft phones.
Later this week I hope to get UK ISDN2e up and running with it!
My question is I
Don't use E-1 channel banks. Pick up the new Digium card, TE410P, run
your E-1 connection to the telco and run T-1 channel banks on the other
spans.
Jeremy McNamara
Anton Tinchev wrote:
Need to buy 2-3 channel banks for some asterisk deployments...
what's the shortcoming of E1 channel banks?
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 21, 2003 2:36 PM
Subject: Re: [Asterisk-Users] Best E1 channel bank?
Don't use E-1 channel banks. Pick up the new Digium card, TE410P,
price, price, price, price
On Mon, 21 Jul 2003, johncn wrote:
what's the shortcoming of E1 channel banks?
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 21, 2003 2:36 PM
Subject: Re: [Asterisk-Users] Best E1 channel bank?
I bought second hand E400P for around $450.
Jeremy McNamara wrote:
Don't use E-1 channel banks. Pick up the new Digium card, TE410P, run
your E-1 connection to the telco and run T-1 channel banks on the other
spans.
Jeremy McNamara
Anton Tinchev wrote:
Need to buy 2-3
Title: Message
Does anyone have any
views on the best software base SIP client to use that normal users could use
with Asterisk without being too techie ?
I have tried the
X-Lite client with varying success. The first version worked OK but music on
hold broke the voice paths and the
Should I be able to see a process starting using mpg123 ? Because I
don't !
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stuart Hirst
Sent: 21 July 2003 09:02
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Music on hold Read error on sound device
Hi
MOH seems to work fine for me now, however, one thing I did spot by reading the source
when it wasn't :-)
MOH MP3Player call mpg123 from '/usr/bin' and by default a mpg123 source install
lives
in '/usr/local/bin' these days.
Might not be this affecting you, but try a: ln -s
Unsubscribe
Ta wiadomosc zostala sprawdzona na obecnosc wirusow komputerowych przez system antywirusowy na serwerze IT Form.
Hi All,
I seem to be having a problem with calls from Asterisk into the AS5300, I am sniffing
the session between the AS5300 and the Asterisk server and I see the Asterisk server
send a SIP INVITE and the AS5300 responds with a SIP 100 TRYING but then I do not see
any more SIP signalling
Darren,
You are a diamond. That worked a treat. Thanks for taking the time to
reply.
Stuart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren Smith
Sent: 21 July 2003 12:23
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Music on hold Read
Hi,
I am looking for call termination in the UK so that I can place calls via my internet
line instead of buying more PSTN lines.. anyone know of amy providers in the UK..
somthing like nufone.net in the UK would be perfect..
Later..
--
__
Hi Nick,
You'll probably run into quality problems making calls over the ISDN from
Xten via *. We did which led us to try several other softphones which were
better and worse, e.g. Pingtel was great from a quality point of view but
the interface wasn't.
We're using snom 100s at the moment which
We use our own gateway for h323 and sip shortly. Contact me offline.
Tan
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 21, 2003 1:43 PM
Subject: [Asterisk-Users] UK call termination..
Hi,
I am looking for call termination in the UK
Hello, * newbie here,
I'm designing a setup that is to eventually be used in a production
virtual PBX/VoIP service.
Customers need to be able to change their setups over the web - I want
them to be able to do simple things like setting up call forwarding, as
well as more intricate stuff that
Hi Steven!
Small world isn't it?
On Mon, 2003-07-21 at 15:52, Steven J. Sobol wrote:
Hello, * newbie here,
I've been lurking on the list for a few months now.
I'm looking at DynExtenDB (and have played with it). I love that it reads
the dialplans out of a MySQL database - that is a
Hey!
On 21 Jul 2003, Armand A. Verstappen wrote:
I've been lurking on the list for a few months now.
I'm looking at DynExtenDB (and have played with it). I love that it reads
the dialplans out of a MySQL database - that is a critical issue for me.
But it has some issues.
I haven't
Has anyone had X-Lite Build 1016 working with Asterisk and if so what
settings within the X-Lite client did you use ?
Rgds,
Stuart
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I'm just curious if anyone has the X100P Callerid receiving working
outside US.
Replies are appreciated. Also if it's not working for you in a certain
coutry you can respond too.
regards
Martin
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Can you have them try with just a headset?
Mark
On Mon, 21 Jul 2003, Linus Surguy wrote:
We're currently running a PSTN - SIP gateway with Asterisk. We also run
IAX/SIP - PSTN.
We have performed a test where the call is routed
UK PSTN - Digium E1 card - Asterisk GW - SIP G.711
Might enter it in the new Asterisk bug tracker
Mark
On Mon, 21 Jul 2003, Tan Aks wrote:
Hi,
I don't know whether only we are experiencing this problem but it seems that if
authentication is used on a couple of phones, and then the authentication is removed
(i.e. remove the secret
On Tue, 22 Jul 2003, Jeremy McNamara wrote:
Karl (klasstek) and myself (mainly Karl) has spent a few clock cycles
figuring out how to make dynamic extensions happen, but we had no real
motivation to finish the task.
Well, I'd certainly be willing to pick up the project from you. I think it
One can use the retrieve_extensions_from_mysql.pl script and then issue a
extensions reload command to asterisk. The pending calls are unaffected
and the final substitution of the new dialplan is done in a very short
time.
regards
Martin
On Tue, 22 Jul 2003, Jeremy McNamara wrote:
DynExtenDB
A brandly new E400P 128 channel PRI, you can find
more information on the Digium's site. The card was not used before, I sell it
cause our company just don't need, we use Cisco AS5300. I can offer you the PCI
PRI at $940 if bought in the next 2 days, I can send it to you via FedEx or DHL
Hi,
I would like to know how suppress number for outside dialling in
CDR table. For example, if I need press 9 key to make an outside call, I
would like that the number in dst field in cdr table was the outside
number without 9 key. It's possible?
Thanks in advance,
srsergio
Try to install the new codec code that is available in
ftp://ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary/codec_g729b.so
place it in /usr/lib/asterisk/modules and restart asterisk (or try to
start it).
There is also a new command available g.729 show license usage and a few
fixes
How can I use ztmonitor to figure out the caller id sent by the telco?
Because it is not working for me in Chicago.
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 21, 2003 9:03 PM
Subject: Re: [Asterisk-Users] anyone with X100P
I don't know yet. However you should be able to hear some wierd signal
that is callerid codec in FSK mode.
regards
Martin
On Mon, 21 Jul 2003, Tamas Levente wrote:
How can I use ztmonitor to figure out the caller id sent by the telco?
Because it is not working for me in Chicago.
-
Yes, you can contact over the manager interface (you need to setup a
user/pass in /etc/asterisk/manager.conf). I've sent a short perl script
how to do that some time ago.
Now notice that extensions reload only renews extensions without
touching other modules.
regards
Martin
On Mon, 21 Jul 2003,
Hi,
I don't know whether only we are experiencing this problem but it
seems that if authentication is used on a couple of phones, and then
the authentication is removed (i.e. remove the secret parameter from
each of the extensions), then this isn't reflected in asterisk after
a reload.
I don't know if 911 uses caller ID or BTN (Billing Telephone Number)
900 calls, operator calls, and 800 calls use the BTN not the Caller
ID...
Anyone
3. Re: E911 and asterisk (Martin Pycko)
Message: 3
Date: Mon, 21 Jul 2003 12:05:38 -0500 (CDT)
From: Martin Pycko [EMAIL PROTECTED]
To:
I have seen this, along with other strange SIP auth issues. I just
thought that you HAD to stop and restart * for the changes in the
sip.conf file to be reread. I also have not been able to get auth to
work. If I put a password in the Windopws Messenger field asterisk does
not authenticate. I
I'll try to set up a retest and report back.
Linus
Can you have them try with just a headset?
Mark
On Mon, 21 Jul 2003, Linus Surguy wrote:
We're currently running a PSTN - SIP gateway with Asterisk. We also
run
IAX/SIP - PSTN.
We have performed a test where the call is
Quoting Myself :(
Hi,
I'm having a problem with DTMF tones from my SIP client
apparently crashing
the chan_capi driver. However I'm not sure whether this is a bug or
misconfiguration on my part: if I set softdtmf=1 in
/etc/asterisk/capi.conf the problem goes away. Does the AVM B1 not
I'm trying to debug a problem with robbed bit signalling on a T1
coming into an Asterisk box on a T100P card. Specifically, I need
to look at the signalling timing. Is there a way to turn on this
kind of debugging in Asterisk, similar to what 'pri debug' does?
On Mon, 2003-07-21 at 19:25, Martin Pycko wrote:
I'm just curious if anyone has the X100P Callerid receiving working
outside US.
It does not work in the Netherlands. The Netherlands does not use FSK
signalling, but DTMF signalling:
1) polarity reversal
2) DTMF: DNumberC
3) ring signal
where
Dear Pals
One customer has a Panasonic
PBX KX-T336 with 60 ext. and a E1 (R2) for Trunks working perfectly now,
This customer
has 10 wireless links to his
branches, wireless working great now, no voice at the present
MY IDEA :
T1Card into the Panasonic (additional
to the
Has anyone had any experience using asterisk for a 911 call center? Does anyone know
of any reason why it would not be suitable? As far as I know all 911 call routing
takes place at the CO switch so a regular T1 line should work fine. I understand that
there is support for ACD in asterisk and
911 trunks are usually delivered to public-safety answering points (PSAP) on
analog reverse-battery facilities. (The PSAP provides battery toward the CO).
ANI is provided using MF tones. The PSAP equipment must take the ANI and use it
to submit a database query to lookup the caller's address (ALI
Daryl, thanks for the info! I'll checkout those companies.
Gene
-Original Message-
From: Daryl Jones [mailto:[EMAIL PROTECTED]
Sent: Monday, July 21, 2003 11:39 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Using asterisk for a 911 call center
911 trunks are usually
For the information of others (and Google) this is
the table structure I used to get cdr_mysql working:
CREATE TABLE cdr ( calldate varchar(255)
NOT NULL default '', clid varchar(255) NOT NULL default '',
src varchar(255) NOT NULL default '', dst varchar(255) NOT NULL
default '', dcontext
Hello,
Is Oh323 supports g729 codec from digium? I saw an g729 option in the
oh323.conf but I have also read some post in the mailing list saying that
oh323 doesn't support g729 codec from digium.
Foong
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Cant have been that obvious...
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 22, 2003 4:32 PM
Subject: Re: [Asterisk-Users] MYSQL Table Structure
On Monday 21 July 2003 23:21, Aaron Martin wrote:
For the information of others
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