Yeah you could use just a few analog ports and patch the phones to a patch
panel and then have several phones on each FXS extension... However if you
want them to all have their own ... I am not sure... Ill think about it, but
right now the Cisco ATA or other FXS port at 1 FXS port / extension
Samy Antoun wrote:
Sorry I'm late. How about a shameless plug for my
distro - AstLinux.
Kris,
I was taking a look at your site yesterday, great
work. One day I'll get a Soekris and try it out. I'm
downloading the PC distro now to give it a try.
Appreciate your work
Samy
ps. Why
Jeff Heath wrote:
I checked out CVS HEAD today and tried to compile it with no luck, so
then I checked out the stable version and compiled it successfully. I'm
99% sure that I'm not missing anything and that I'm following the
instructions correctly (I'm no guru, but I've compiled lots of
Does anyone have an estimate for the pricing on the DS3000P DS3 PCI card by
Digium? How about a timeframe?
Thanks,
Nathan
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To
For bridging VOIP with PSTN Lines
Which one is giving better performance SIP or IAX ?
I am looking at a result without NAT in picture ?
Can some body give details from experiance ?
Also with single SIP/IAX channel can I use more than one call at a time ?
Thanks
Sandeep
Sandeep A.S wrote:
For bridging VOIP with PSTN Lines
Which one is giving better performance SIP or IAX ?
I am looking at a result without NAT in picture ?
Can some body give details from experiance ?
I can't say anything based on experience, but guessing that IAX2
trunking will
I'm now uprunning with
- mISDN with avmfritz driver for Fritz PCI card
- chan_capi from debian recompiled with a patch (see below)
- EuroISDN with Point-to-Point (ptp) mode (Austria)
- With Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k from debian sarge
But am having some problems:
1) I needed to patch
In article [EMAIL PROTECTED],
Samy Antoun [EMAIL PROTECTED] wrote:
It really depends on what kind of load that cpu is
going to have. There's no
technical problems with doing the above. Except I
don't see the point with
having a dhcp server, unless you are an ISP.
Steve,
Thank you
Another alternative is to use H.323 FXS IAD in combination of H.323
channels. I bought a 4-port IAD of US$50 per port. It works for me!!!
Let me know if you will be interested in the product.
On 6/2/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
I'm looking for an inexpensive way to connect 20
On Wed, Jun 01, 2005 at 08:27:31PM -0700, Samy Antoun wrote:
Hi,
I'm planning to get my Asterisk box out of the LAN,
get rid of my router and make the box acts as a
Router, Firewall, DHCP Server (with Shorewall).
Regarding the DHCP part:
On Rapid we added dnsmasq as an the dns/dhcp server
Hi,
I assume that you are talking about fixed line MMS like it is
implemented in Germany. Some time ago i already played a little
bit with a Gigaset SL74 (and an ISDN dect base). So far as ISDN
is concerned the basestation uses a PPP connection to connect
to a HTTP Server for sendind/retrieving
Dear All,
I was trying to enable call forwarding, following the steps of the link
on voip.org regarding this issue it doesn't work and the phone I am trying
to implement on is still ringing. below is my conf in extensions.conf and
the CLI output during the process.
My configuration is :
exten =>
7910's with Chan_sccp are running well except some cases.
As usually Transfer, Msgs, Conf,Forward,Speed buttons are not working!
Infact I have already knew that, those buttons will not work (from
mailing list).
It seems like the * console is very busy with messages constantly on
it from the sccp
On 02/06/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote:
Dear All,
I was trying to enable call forwarding, following the steps of the link on
voip.org regarding this issue it doesn't work and the phone I am trying to
implement on is still ringing. below is my conf in extensions.conf and
Hello!
I'm new to asterisk and linux, so please don't blame me if I write silly
things :-)
I'd like to setup a system with IVR only.
I'll use a SIP gateway to receive calls from the outside world, and I'll
install asterisk on a dedicated linux server placed in another location
that will be
Dear Peter,
here is my 777 conf in extensions.conf:
[Internal-sip]
exten = 777,1,Dial(SIP/777,7,tr)
exten = 777,2,Dial(SIP/777SIP/888,10,tr)
exten = 777,3,voicemail,u777
exten = 777,104,voicemail,b777
As for the stdexten macro I really don't know what you mean by using it do you
mean
by doing
Tony Mountifield [EMAIL PROTECTED] uttered the following thing:
In article [EMAIL PROTECTED],
Samy Antoun [EMAIL PROTECTED] wrote:
It really depends on what kind of load that cpu is
going to have. There's no
technical problems with doing the above. Except I
don't see the point with
Hi.
I have connected my Asterisk (inside my NAT) to IPTEL.org.
I can call others users of iptel, but anyone can call me. They listen a busy
tone
Someone can help me???
my sip.conf:
register = prueba22:[EMAIL PROTECTED]/phone2
[iptel]
type=friend
host=iptel.org
port=5060
fromdomain=iptel.org
On 02/06/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote:
here is my 777 conf in extensions.conf:
[Internal-sip]
exten = 777,1,Dial(SIP/777,7,tr)
exten = 777,2,Dial(SIP/777SIP/888,10,tr)
exten = 777,3,voicemail,u777
exten = 777,104,voicemail,b777
As for the stdexten macro I really
Hi
Michael,
I
installed the asterisk-oh323-0.6.6-pre4 and is working nice as I registered to
the gatekeeper zone I wanted the problem is when I try to make an outside
number or even a registered sip user in asterisk the h323 tries to make the call
but I get the an error in the console
Hi ;
Have two handytone 486 and want to use them as digium TDM400
fxo-fxs card...
I mean is it possible to direct pstn calls from astersik
(extensions) to handytone line port directly and
vice versa ?...
Thanks in advance
Betul
Onemli not : Bu e-mail iletisi, sadece
Betl Gzlkolu wrote:
Hi ;
Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card...
I mean is it possible to direct pstn calls from astersik (extensions) to
handytone line port directly and
vice versa ?...
The answer to the same question a few days ago was,
Two asterisk, one in a 2.4.x and another one in a 2.6.x are connection ok
using IAX, before i upgraded one of them (the one with 2.6.x) Trunk was
working ok. Since i upgraded the one that its now 2.6.x, i m getting this
message:
chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only
C F wrote:
On 6/1/05, Joseph [EMAIL PROTECTED] wrote:
Is there updated spot on the wiki that shows how then new setgroup and
checkgroup functions are to be done?
For example, if we want to make sure a sip phone can only take one call
from the queue, how would you check the current call count?
I have an Adtran 600 (12 x FXS, 12 x FXO) connected via Sangoma A101 to an
Asterisk server which is in the lab for testing. The channel bank is
disconnecting every few hours without apparent reason and no-one can tell
why, not even Sangoma who have worked very hard to determine the cause of
the
Hi,
Some background:
I would like to call my girlfriend over the internet. We are both behind a nat
router and I want to avoid portmapping.
I've heard that you can call someone behind a firewall (nat router) with the
IAX protocol, but I'm not sure.
The questions:
Do I have to set up my own
At this point we have close to 200 registrations for Astricon Madrid and
new attendees are registering all the time. Make sure you register now
in order to get a seat at this first community meeting in Europe,
arranged by IPsando and Digium - the company behind Asterisk.
The detailed tutorial
Hi Peter,
You are totally right it worked, and I really loved the macro idea I
have mostly grasped it now and will use it more extensivley in the future.
Thx
MAG
Peter Bowyer wrote:
On 02/06/05, Mohamed A. Gombolaty [EMAIL PROTECTED]>
wrote:
> here is my 777 conf in extensions.conf:
>
I'm planning to get my Asterisk box out of the LAN,
get rid of my router and make the box acts as a
Router, Firewall, DHCP Server (with Shorewall).
I'll do that to be able to use some SIP clients
remotely.
Does anyone doing the same with the Asterisk box, is
it a good idea, is there any
[EMAIL PROTECTED] wrote:
This comes in with a price tag of £56 ( $100 ). It has been 6 weeks since
I purchased the contract. Still no news! and no access!
The UK vendors don't seem to like dealing with these contracts.
I waited two months. It took legal threats before the vendor (Lanway)
Samy,
Sorry I'm late. How about a shameless plug for my distro - AstLinux.
It sounds like it may work well for your needs:
http://www.kriscompanies.com/modules.php?name=Contentpa=showpagepid=3
--
Kristian Kielhofner
Hi,
I like the look of this, however I'm not ready to go the
He is right Karl. Without the ship-to being on file with the bank..
the company can be held responsible for fraudulant purchases.
On 5/31/05, Karl J. Vesterling [EMAIL PROTECTED] wrote:
I'm amazed that this thread keeps going...
About the claim of Ship-To being on file with bank...
Hendrik Wouters a crit :
Hi,
Some background:
I would like to call my girlfriend over the internet. We are both behind a nat
router and I want to avoid portmapping.
I've heard that you can call someone behind a firewall (nat router) with the
IAX protocol, but I'm not sure.
The questions:
Dear All,
I was trying to make call confrence available but all the asterisk documents
use the meeting room concept, where those who wanna meet have to dial an
extension corresponding to the meeting room, while call conference actually
means that I am on exten 100 I can dial exten 200 and add it
Yes.. you should have spent 16 hours browsing the small print and
Terms of Service (if it took you that long). Good Grief man! You
must be one of those people sueing Vonage because their 9-1-1 setup
procedure isn't clear!
On 5/31/05, Karl J. Vesterling [EMAIL PROTECTED] wrote:
Garrett,
Hi
Has anyone heard of solutions for implementing call-hunting
over a bank of gsm lines / sim cards?
We wan to have a single gsm dial in number for access to
asterisk.
Only solution I know about is HP's opencall which is very
expensive.
Apologies if this is somewhat off-topic.
Thanks
Eric
Hello!I'm new to asterisk and linux, so please don't blame me if I write silly things :-)I'd like to setup a system with IVR only.I'll use a SIP gateway to receive calls from the outside world, and I'll install asterisk on a dedicated linux server placed in another location that will be
I have tried numerous versions of asterisk from asterisk at home to
compiling it myself through the cvs server. I don't
understand it works
fine with the intel p2 box but not the faster via cyrix box.
Is it the
processor or something?
Have a look in the makefile. It might be
I'm looking to connect Asterisk with three (four in the future) PSTN
lines, and would like to get some opinions on the TDM400 Digium card, vs.
sip gateways like the Mediatrix 1204, vs. other hardware solutions I'm not
yet aware of.
I need the ability to prioritize which PSTN lines are
On Thu, 2 Jun 2005, Mohamed A. Gombolaty wrote:
I was trying to make call confrence available but all the asterisk
documents use the meeting room concept, where those who wanna meet have
to dial an extension corresponding to the meeting room, while call
conference actually means that I am on
On Thu, 2005-06-02 at 11:16 +0530, Adnan Ahmed wrote:
On 5/31/05, Anton Krall [EMAIL PROTECTED] wrote:
because SugarCRM is far
sophasticated and robust than anyother CRM package
Just to add some choice, check out CentricCRM at www.centriccrom.com.
Similar functionality as SugarCRM.
On Thu, Jun 02, 2005 at 01:56:09PM +0600, Yusuf Iqbal arranged a set of bits
into the following:
7910's with Chan_sccp are running well except some cases.
As usually Transfer, Msgs, Conf,Forward,Speed buttons are not working!
Infact I have already knew that, those buttons will not work (from
Well, I'm looking for each phone to have its own extension.
Waldo
On Jun 2, 2005, at 2:02 AM, Matthew H. Franz wrote:
Yeah you could use just a few analog ports and patch the phones to
a patch
panel and then have several phones on each FXS extension... However
if you
want them to all have
I would be interested. However, from what I've heard of the problems
of H323 and NAT, I don't know if that would be a viable solution.
These are to be used for a remote office where the phones will be
behind NAT and potentially, the asterisk server will be behind
another NAT.
Thanks,
-
I am about to perform this same installation Scott so I'll be wathcing this
closely.
What type of T1 card do you have in the Mitel? Does it take a 15 pin serial
or an RJ 48x? The one I need to install to has the RJ 48x and we are trying
to figure out if it needs to be straight through or
Laurent Lesage wrote:
thanks for the aswers but I forgot to say that I would like it work
with Linux. I think all of you use it with Winxx? And DIAX works just
for Winxx? What would be the best, that's to use a BT fixed on the
Asterisk server, so that you do not have to use another computer
On Thursday 02 June 2005 02:48, Sandeep A.S wrote:
For bridging VOIP with PSTN Lines
Which one is giving better performance SIP or IAX ?
I am looking at a result without NAT in picture ?
Can some body give details from experiance ?
Also with single SIP/IAX channel can I use more than one
I don't know, but pricing it per line whould be safe. Say $100 per
line that would be $67,200.00. So anything less than that would be
great. I think it will be about $20 bucks a port.
672 * 20 = 13,400
On 6/2/05, Nathan [EMAIL PROTECTED] wrote:
Does anyone have an estimate for the pricing on
Have I missed something, or why not use a channel bank?
I use an Adit 600 CMG along with the MGCP protocol and has 40 FXS lines.
There are really cheep ones at ebay. Take a look!
--
Daniel
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T1 card and Rhino Channel bank. That should be about $84 bucks a port.
On 6/2/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
Well, I'm looking for each phone to have its own extension.
Waldo
On Jun 2, 2005, at 2:02 AM, Matthew H. Franz wrote:
Yeah you could use just a few analog ports
On Thursday 02 June 2005 03:14, Tony Mountifield wrote:
I find DHCP on my LAN extremely useful for both my and visiting laptops.
Any machine that will be using my LAN regularly gets a static entry in
/etc/dhcpd.conf so it will always get the same IP address. It also gets
an entry in my local
On Thu, 2005-06-02 at 10:07 -0400, steve szmidt wrote:
On Thursday 02 June 2005 03:14, Tony Mountifield wrote:
I find DHCP on my LAN extremely useful for both my and visiting laptops.
Any machine that will be using my LAN regularly gets a static entry in
/etc/dhcpd.conf so it will always
Ditto That:)
Thats what i use!
What's wrong with :-
host W2K {
hardware ethernet 00:30:1B:AC:39:E3;
fixed-address 192.168.1.130;
}
this box always gets the same IP and I know who's got what.
___
Try ATCOM's AG168V available from US
Distributorhttp://www.iareaphone.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
DineshSent: Wednesday, June 01, 2005 9:40 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] IAX2 analog
telephone adapter
Hello
All,
Hi:
I' have one TDM400P and need configure my linux box like modems.
It's posible give my users with PC/modem/windows/diul up line access to
internet with my linux/asterisk/tcm400p??
Thanks in advance
Edgardo
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For * behind another NAT, if I am not wrong your NAT has to open up a
range of ports. From security point of view, this is not a good idea.
I am not sure if there is any better solution available. For a NAT
environment that was NOT mentioned, my setup is rather costly that may
violate your
On Thursday 02 June 2005 10:17, Dave Cotton wrote:
What's wrong with :-
host W2K {
hardware ethernet 00:30:1B:AC:39:E3;
fixed-address 192.168.1.130;
}
this box always gets the same IP and I know who's got what.
Nothing, that's really how they
I am connecting to a Nortel CS 1000. I can place calls out to an extension
so we are half way there. When calling into the box I get the following from
sip debug ip X.
I get dead air when calling into the box.
In my sip.conf I have a context of nortel and in extensions.conf the nortel
context
Hi,
we've purchased new Beronet Octobri card and have problems loading modules
for stock Asterisk for Debian Sarge (it has bristuff patches in it).
when loading qozap it says that no multibri card was found although lspci
shows it... There were quite some rumours about bristuff not liking
I' have one TDM400P and need configure my linux box like modems.
It's posible give my users with PC/modem/windows/diul up line access to
internet with my linux/asterisk/tcm400p??
Probably not. The TDM card (or drivers) seem to have an issue with
missed data across the pci bus which will
Raul Elizondo (wizardteam) wrote:
Two asterisk, one in a 2.4.x and another one in a 2.6.x are connection ok
using IAX, before i upgraded one of them (the one with 2.6.x) Trunk was
working ok. Since i upgraded the one that its now 2.6.x, i m getting this
message:
chan_iax2.c:5067 socket_read:
On Thu, 2005-06-02 at 10:27 -0400, steve szmidt wrote:
On Thursday 02 June 2005 10:17, Dave Cotton wrote:
What's wrong with :-
host W2K {
hardware ethernet 00:30:1B:AC:39:E3;
fixed-address 192.168.1.130;
}
this box always gets the same
Hello - thanks for your reply,
I think the T1 card and channel bank will be a bit to expensive for
this time around (5 phones and 3 PSTNs); do you have specific
gateway devices you have used and are happy with?
Thanks - Ivan.
On Wed, 1 Jun 2005, C F wrote:
In my experience if you can get
Hello,
I am new to asterisk (i started to try tree days ago) and i have managed
to setup asterisk to fit my needs so far exept one thing.
I want to setup asterisk as an SIP ISDN Gateway. As I said things are
going well so far. I can make calls from SIP phones to other SIP phones,
and incoming
Dave Cotton wrote:
On Thu, 2005-06-02 at 10:27 -0400, steve szmidt wrote:
On Thursday 02 June 2005 10:17, Dave Cotton wrote:
What's wrong with :-
host W2K {
hardware ethernet 00:30:1B:AC:39:E3;
fixed-address 192.168.1.130;
}
this box always gets the
On Thu, 2005-06-02 at 10:09 -0500, Kristian Kielhofner wrote:
Dave Cotton wrote:
Have another look at it because this only scratches the surface. I love
DHCP.
I second this completely. ISC DHCP allows you to do some crazy
things... Start doing diskless clients with
With the addition of web based CRM into this machine, is there a plan to
implement traffic shaping in some way?
I know my users will be busy with the CRM side of this but I would not
want it to affect the RTP side fo things if too many users are connected
to the web.
Thanks,
Wiley
Hi,
I already have a G729 license and the voice is OK
also I tried to use G729 codecs on SIP and AS5400 side, without any g729
license and works fine with a pass-trought configuration, so you only
need to check out voip-info.org to solve that issue.
on the stun and NAT part, I have not reached to
I have a SIP trunk coming into my asterisk box.
the error is Unable to create/find channel.
How do I define that a DID number is handled in a certain manner???
Asterisk cannot find how to handle an incoming call from a number.
I have in sip.conf my definition to the PBX. That seems good as I
Can anyone tell me the correct settings for a kpn e1 line in a digium Te110p
card in zaptel.conf and zapata.conf specifically
In zapata.conf
Span=1,0,0,ccs,hdb3,crc4 (is this correct and does kpn use crc4?)
In zapata.conf
Switchtype=euroisdn (pretty sure this is right just checking)
I agree with you as well as with Daniel Nystron (from previous post).
However, putting a T1 card and a cheap channel bank is still somewhat
expensive, if you think about it. The Sipura SPA-2002 costs about $70
and that's 2 ports, which means $35/port. That's even better than $84/
port. I'm
Unless something about your setup is causing data corruption of some
kind, then I don't think the O/S or kernel has anything to do with this.
It appears that one side is using CVS-HEAD, and sending trunktimestamps,
and the other side is using stable, which doesn't understand them. So,
you need
Remove the Tthr options. You don't need any of them in the dial string for
AT320s
Seshu
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis
Sent: Wednesday, June 01, 2005 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Jeff Heath wrote:
I checked out CVS HEAD today and tried to compile
it with no luck,
It happened to me and I have seen it happen to many
people when they upgrade their Asterisk. I came to
conclude that it depends on the versions of each of
Zaptel, libpri, and Asterisk. So Asterisk might
I have been using an old DELL Dimension XPS as my asterisk server. It
has been using a Sangoma A101 for PRI.
I tried adding a TDM20B. Whenever I load wcfxs I get ProSLIC 3210
version 2 is too old.
Is this because my board is PCI 2.1, or a DELL, or interrupt issues, or
all of the above?
Just found out my self :-D. Problem was a syntax change in 0.4.0pre1
version of chan_capi.
All the resources I found on the web were written with the old syntax.
Matthias
Am Donnerstag, den 02.06.2005, 17:01 +0200 schrieb Matthias BXhm:
Hello,
I am new to asterisk (i started to try tree days
I know I'm running the risk of fanning the flames on an already belabored
thread here, but there is some misinformation flying around.
Credit card fraud is an unfortunate fact of life, and it costs everyone who
isn't perpetrating it money. There is no single universally agreed on
process that
On Thursday 02 June 2005 11:18, Dave Cotton wrote:
On Thu, 2005-06-02 at 10:09 -0500, Kristian Kielhofner wrote:
Dave Cotton wrote:
Have another look at it because this only scratches the surface. I
love DHCP.
I second this completely. ISC DHCP allows you to do some crazy
I have the sip trunking as below :
I tried with IAX Trunking .But no success
Can some one send IAX Trunking config for the below setup replacing
SIP ?
PBX1 (192.168.10.2)
==
sip.conf
--
[pbx]
type=friend
username=pbx
secret=pbx
host=192.168.1.2
extensions.conf
Hi,
Don't know if you've sorted this or not, but
zaphfc: empty HDLC frame or bad CRC received (framelen = 5,
stat = 0xff,
card = 0).
I've only ever seen when the signalling is wrong. For example if the line is
in PTMP mode when it should be in PTP or vice-versa.
this is the
I have been using an old DELL Dimension XPS as my asterisk server. It
has been using a Sangoma A101 for PRI.
I tried adding a TDM20B. Whenever I load wcfxs I get ProSLIC 3210
version 2 is too old.
Is this because my board is PCI 2.1, or a DELL, or interrupt issues, or
all of the
Hi all,
Is it possible to use 2 incoming fxo lines (one is for my company the
other for the family) with [EMAIL PROTECTED]
Best regards,
Francois
Random Thought:
---
Errors like straws upon the surface flow: Who would search for pearls must dive
below. - John Dryden, 1631 - 1700
Hi,
(I seem to be having some trouble getting messages to post on the list so
I may be duplicating an earlier post. Apologies if this is the case.)
I am compiling CVS tip Asterisk on a fresh CentOS 3.4 install. I got
this warning:
make ast_expr.a
make[1]: Entering directory `/usr/src/asterisk'
Details on IAX trunk can be found here...
http://www.voip-info.org/wiki-IAX
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sandeep
A.S
Sent: Thursday, June 02, 2005 9:15 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Replacing SIP
You can support as many as you want. You just need to update your
zapata.conf file and change this line...
channel=1-8
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francois
Meehan
Sent: Thursday, June 02, 2005 9:19 AM
To:
This is assuming you have problems with the autoconfig.
The latest seems to add the lines just fine.
When I started using 0.06, I had to do it manually.
W
-Original Message-
From: Wiley Siler
Sent: Thursday, June 02, 2005 9:29 AM
To: '[EMAIL PROTECTED]'; 'Asterisk Users Mailing List
I expect to have 8 concurrent calls maximum.
snip
I guess the disk, the RAM and the NIC will do, but I'll suggest a P4 instead
of a Celewrong, as 128k cache is a little low.
...
I surprised there are no enthusiast here trying to push things a their
limit a bit... if all he's doing is data
Look in modules.conf and you will see that wctdm is just an alias for
wcfxs.
On Thu, 2005-06-02 at 11:17 -0600, Rich Adamson wrote:
The TDM card uses wctdm (not wcfxs).
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I have been working with this for a wile and I have been watching
the list for about a month on this subject, to no avail.
I am wondering if anyone has successfully configured asterisk for
clients to connect to it when the clients are behind nat. I mean
successfully because I can do
Lance,
Have you configured your sip.conf to use these aprameters under General?
;externip=66.213.227.66
;localnet=192.168.1.0
;localmask=255.255.255.0
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lance
Grover
Sent: Thursday, June 02, 2005 9:39 AM
Karl has already stated more that once that this was NOT a credit card
purchase. If a credit card was not used for the purchase, why would you
need a Ship To address on file with the credit card company? Cory Andrews
from VOIPSupply has also admitted that the sales rep who took the order
sig type too fast and shift + enter will send the email early...
OK. Anyway... The parameters below are important for the issue you
have.
The wiki covers this under the sip section www.voip-info.org
W
-Original Message-
From: Wiley Siler
Sent: Thursday, June 02, 2005 9:51 AM
To:
Anyone know what exitstatus 127 for Asterisk is?
I can't find it on the wiki or anywhere.
I have an install being started by safe_asterisk but it loops because of
that error. However, if started w/o the script it starts fine.
Thanks,
Nathan
___
El mié, 01-06-2005 a las 22:27, Samy Antoun escribió:
I'm planning to get my Asterisk box out of the LAN,
get rid of my router and make the box acts as a
Router, Firewall, DHCP Server (with Shorewall).
I'll do that to be able to use some SIP clients
remotely.
Does anyone doing the same
And the point is that this should be a dead issue.
The vendor resolved the problem as quickly as possible and took
responsibility for the mistake.
This really should not be an issue that has to keep going for another
week
W
-Original Message-
From: [EMAIL PROTECTED]
El jue, 02-06-2005 a las 02:14, Tony Mountifield escribió:
Thank you for the valuable advice, I'll do exactly
what you are suggesting, No DHCP
I find DHCP on my LAN extremely useful for both my and visiting laptops.
Any machine that will be using my LAN regularly gets a static entry in
Thanks Wiley,
I was asking the question because, I don't have my card yet (TDM02B 2-Port
FXO) but did install [EMAIL PROTECTED] on a server and when looking at the
AMP-setup Incoming calls section, I see only one destination for all
incoming calls.
I assume then that once the card is installed,
Actually, it doesn't exactly work that way.
The number of incoming lines is not relevent to the incoming calls
section in AMP.
That section just allows us to decide what to do with calls coming in
from our PSTN side of the PBX.
Your second port should get grouped with the first creating a trunk.
All,
I am connecting to a CS 1000 nortel PBX. I can call out,
I have limited success with call in. I get debug traffic that a call
is coming in but I get the message Unable to create/find channel.
I was expecting that incoming calls over the trunk would
be handled from my sip definition and
The phone don't ask for this file, that's weird...
Regards,
--
Ing CIP Alejandro Celi Mariátegui
[EMAIL PROTECTED]
El mié, 01-06-2005 a las 22:49, Omar Sabek escribió:
Alejandro,
Do the phones ask for a file named CTLSEPmac_address.tlv? If so, you
will need to specify the firmware
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