As mentioned yesterday, i made an attempt to write documentation to get
NAT + SIP to work on http://www.asteriskguru.com/natut.php
If you send me the info for those phones, firewalls i will include them.
(I was planning on adding some Linux/BSD firewall rules but i dont have
a pix,).
/Z
Irakli Nats
Hi Peter!
FYI: Yesterday i put Asterisk between a Hicom 350E and a Telekom Austria
(TA) PRI. Both use TON=unknown for called number, but Hicom always uses
TON=international for calling number whereas TA uses a dynamic TON for
calling number. Thus, for incoming calls (PSTN->PBX) the presented
ca
Does anyone know of a Linux SoftPhone that will play nicely with ESD?
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Hi Chris,
I've had A LOT of experience with the cheap X100Ps in the last few weeks. I
myself bought two of them off ebay. $6.95 special! I also had problems with
them and within the past 12 hours have replaced them with a TDM22B that so
far(1 phone call) has worked great. I would suggest turning v
Joseph wrote:
Can anybody explain me why IAX is called proprietary protocol?
In some places IAX is refereed as "open protocol".
How can proprietary protocol be open protocol?
Since the source code is available to anyone and GPL'ed it is an open
protocol.
However it's not a standard and there i
On Mon, 25 Apr 2005, Andrew Kohlsmith wrote:
> It has absolutely nothing to do with what "economically suits them best" --
> it
> has everything to do with the fact that when you buy a clone X100P you DO NOT
> KNOW what you're getting. The chipset may be the same but as you can clearly
> see
Hi
I am currently running [EMAIL PROTECTED] version 0.9 and have a few questions,
which i hope someone on this list might be able to answer.
1) I am trying to setup incomming fax support, but however i never manage
to receive the faxes, getting a signal 15. As per handbook, there isn't
too much u
Jean-Michel Hiver wrote:
Hi List,
I have seen this:
http://www.convergence.com.pk/iax2/trunked.html
According to this table, using trunking, you can have 16 channels with
171.7 kbps bandwith using g.729 + IAX2 trunking? Sounds too good to be
true...
Any comments on this?
If I'm reading the litt
Hi Ken,
> Can't seem to find it anywhere, and my cisco login works, but says
> there's no longer any downloads available for the ATA186.. anyone know
> where I could find the MGCP version of the firmware via download?
Log in. From the main page, click the dropdown list for
Downloads and select
The user is reporting a problem with IPSwitchboard.
Steve Kann wrote:
youssef ouadou wrote:
Aucune connexion n’a pu être établie car l’ordinateur cible (server)
l’a expressément refusée at System.Net.Socket.Connect(EndPoint
remoteEP) at IPS.listener.Start()
Hmm, I think you have two incorrect la
Hi,
Please include "tT" options in your Dial statements in extensions.conf.
Example:
extensions.conf
[default]
exten => ,1,Dial(SIP/u0001&SIP/u0004,20,tT)
exten => _0XXX,1, Dial(SIP/u${EXTEN},20,tT)
exten => 828112070,1,Dial(SIP/u0001,20,tT)
exten => 828112071,1,Dial(SIP/u0004,20,tT)
Two questions.
If there is a VoIP-VoIP call, how do I see from a console what codecs are in
use by peers?
Second question: if there is transcoding going on, how do I see detailed
information about it - peers involved, extensions, IP addresses, ports,
codecs from/to and so on?
Thanks,
Irakli
_
Hi guys,
I have a TDM400 with 4 fxo ports installed in my
IPPBX box. When I call in my IPPBX through this card
and after it answers I hangup, IPPBX still keeps going
to timeout. It cannot recognize the hangup signal from
PSTN.
Anyone knows the solution.
Thank you so much.
jintwo
__
I think its a win win situation. Cisco has tons of money to throw at
them to get a better product with more features. I dont believe they
would aquire them and not put money in them to make a better product.
I guess the prices will go up like a rocket
Not necessarily, When Cisco acquired
- Original Message -
From: "Isamar Maia" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, April 27, 2005 7:54 PM
Subject: Re: [Asterisk-Users] Linksys/Cisco buys Sipura
I guess the prices will go up like a rocket
Not necessarily, W
Robert,
It looks like you're dialing 447733322998, 44 for UK, then the area
code, etc. I have sipgate.de setup to dial local numbers (any German
number) as 0+AREA CODE+NUMBER. Always dial the area code, even if you
sipgate number is in the same city. For international numbers you need
to dial 00+C
Yall' (being a southern Yankee!) should checkout the app_dictate app in
the Mantis, It allows you to replay and gives you better control for
something like this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Wednesday, April 27, 200
OK a simple AGI can do this, I know you didn't want one, but its only
three lines
Cat findzone.agi
#!/bin/sh
zone=`cat $1.dat`
echo "SET VARIABLE zone $zone \"\"\n"
Put the above script in your agi-bin (usually /var/lib/asterisk/agi-bin)
chmod 755 findzone.agi
Then in your dialplan do th
So now that they are done how about you post the files for us? Share the
wealth.
Mark
Paul Redstone wrote:
To add something to a post of a few days ago on this:
We're just putting in an asterisk system and wanted to have our own
messages. We're Asterisk and are not yet live but the following
Hello-
Before you all get bent out of shape, let me give you the background
on what I have -- what I need help with should be rather easy, but I
can't seem to find it on the Wiki.
All of my business cell phones are subscribed to a "tracking" feature
which uses GPS and/or tower location to send au
Steve Kann wrote:
Something *proprietary* is something exclusively owned by someone
nobody "owns" the IAX2 protocol.
Although, Digium have trademarked "IAX"
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> I have a local telco line hookup to my FXO port.. receive calls just
> fine…but when I try to dial out, it ring the phone connected to the FXS port
> of the same card….not sure where to start looking to fix the problem, thanks
> in advance for your time…
You have other ports on this cards, at le
Most probably your server was busy starting up when asterisk loaded and
calculated the table.
Next time, just issue show translation recalc without after the server
settles down.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Irakl
We can always buy from China, not to mention some of they will support
IAX2 protocol.
#Joseph
On Thu, 2005-04-28 at 09:54 +0900, Isamar Maia wrote:
> I guess the prices will go up like a rocket
>
> Isamar
>
>
> On Wed, 27 Apr 2005, MF Hulber wrote:
>
> > Have you seen this story? Cisco d
Hello.
I just set up * on a new server, MOH does not work.
Musiconhold.conf has:
[classes]
default => quietmp3:/var/lib/asterisk/mohmp3
Extensions.conf has:
exten => 000,1,Answer
exten => 000,2,Musiconhold(default)
Dialing 000 gives the correct MOH.
However, when I receive a call and place i
On what trascoding time depends on?
I started server, run * and issued command show translations
--
sipsrv1*CLI> show translation
Translation times between formats (in milliseconds)
Source Format (Rows) Destina
> >http://www.voip-info.org/tiki-index.php?page=Asterisk%20protocols
> >
> >I think that should be corrected!
> >
> >
> >
> Happy now? :)
>
> Jonathan / denon
Much better :-) (thank you!)
--
#Joseph
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Hmm. How did you do that because I went to the support site, filed a
'question' and i've had no response..
On 4/26/05, Spencer Nassar <[EMAIL PROTECTED]> wrote:
> I saw the same thing. Filed a support ticket with Voice Pulse Connect
> and they cleared it up within an hour.
>
> SN
>
> On Apr 2
I guess the prices will go up like a rocket
Isamar
On Wed, 27 Apr 2005, MF Hulber wrote:
> Have you seen this story? Cisco definitely wants to own the VoIP
> market. I wonder what effect this will have on Sipura products.
>
> http://story.news.yahoo.com/news?tmpl=story&am
How can proprietary protocol be open protocol?
Proprietary means it came from a proprietor - Digium in this case. This
is a completely unrelated issue to whether it is open. Marketing
departments try to confuse the issues. :-)
So if the protocol is not encumbered by any patent or copyright (only
Joseph wrote:
How can proprietary protocol be open protocol?
Proprietary means it came from a proprietor - Digium in this case. This
is a completely unrelated issue to whether it is open. Marketing
departments try to confuse the issues. :-)
Even WIKI is confusing the cause calling it
I am very new to Asterisk, running [EMAIL PROTECTED] .06..I
have a problem, everything on my box seems to be working ok, except:
I have a local telco line hookup to my FXO port.. receive
calls just fine…but when I try to dial out, it ring the phone connected
to the FXS port of the same car
> >How can proprietary protocol be open protocol?
> >
> >
> Proprietary means it came from a proprietor - Digium in this case. This
> is a completely unrelated issue to whether it is open. Marketing
> departments try to confuse the issues. :-)
So if the protocol is not encumbered by any patent
Hi there,
> There are plenty of good documents on Asterisk, SIP and NAT on the
> voip-info.org wiki. Please look them up. There are also information
> within the configs/sip.conf.sample file within Asterisk.
Folks, let's face it - documentation on Asterisk sucks big time. This is
the reason why
Have you seen this story? Cisco definitely wants to own the VoIP
market. I wonder what effect this will have on Sipura products.
http://story.news.yahoo.com/news?tmpl=story&u=/nf/20050427/bs_nf/33554
MARK.
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Aste
If there is another MoH source what is the correct way to use it with
extensions?
Thanks,
Irakli
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Hello everybody,
I'd like to know was there any load tasting done with *? Let's imagine 500
SIP clients on a server, 80 simultaneous calls. No transcoding, G711 or G729
codecs are used between endpoints.
How asterisk performs with 80 simultaneous calls when it sits on a media
stream? Is there any
Stefan de Konink wrote:
On Wed, 27 Apr 2005, Joseph wrote:
How can proprietary protocol be open protocol?
If the protocol is fully documentated and this documententation is
available to anyone you can speak of a open protocol. It is not an open
'standard', because it is only supported by Di
Joseph wrote:
Can anybody explain me why IAX is called proprietary protocol?
In some places IAX is refereed as "open protocol".
How can proprietary protocol be open protocol?
Proprietary means it came from a proprietor - Digium in this case. This
is a completely unrelated issue to whether it is
Can't seem to find it anywhere, and my cisco login works, but says
there's no longer any downloads available for the ATA186.. anyone know
where I could find the MGCP version of the firmware via download?
Thanks!
Ken
smime.p7s
Description: S/MIME Cryptographic Signature
Title: Normal
Well, going into a
CO port on the PBX means that you can just grab a trunk & dial out,
which is nice for the outbound side. Unfortunately, on the inbound side,
the call would just appear on that line - you would have to route the call to an
auto-attendant, or use DISA to allo
IAX is an abbreviation for Inter Asterisk Exchange.
So IAX was a proprietary protocol for interconnecting Asterisk servers,
it was only used with 2 asterisk servers. IAX has always been open for
the community. So some may say it's proprietary, while it is open. At
the current time, the IAX proto
On Wed, 27 Apr 2005, Joseph wrote:
> How can proprietary protocol be open protocol?
If the protocol is fully documentated and this documententation is
available to anyone you can speak of a open protocol. It is not an open
'standard', because it is only supported by Digium, thus proprietary.
http
--- "Kanuri, Seshu (Company IT)"
<[EMAIL PROTECTED]> wrote:
> If you want to save on the bandwidth cost, the only
> way you can do that
> is by adopting a low bandwidth codec like G729 or
> G723, which consumes
> much less bandwidth and can give equal or better
> quality calls.
I pretty much tr
Can anybody explain me why IAX is called proprietary protocol?
In some places IAX is refereed as "open protocol".
How can proprietary protocol be open protocol?
--
#Joseph
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I am trying to do the exact same thing. Do you still have the SIP image and
if so could I borrow it?? I need to convert my 7940G to SIP.
Thank you,
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael West
Sent: Wednesday, April 13, 2005 07:27
T
Do you still have that image for the 7960? I bought a 7940 on ebay and it
doesn't have the SIP firmware. I can't find it anywhere but Cisco's website
and they require that I have an account with them. Did you happen to save
that binary file?
Paul
-Original Message-
From: [EMAIL PROTECTED]
I sit corrected
Thinking of an Ericcson BP250 Config..
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan & Company, LLC
Sent: Thursday, 28 April 2005 9:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Thanks J
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jltaylor
Sent: Thursday, April 28, 2005 11:29 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] CDR
Billing Question.
Really looks like ha
This is incorrect, David - Pins 4 and 5 are the correct pins, and are
not reversed.
David Phelan wrote:
I think you will find it is pin reversed.
So flip the RJ45 Over
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory Junker
Sent: Thursd
I think you will find it is pin reversed.
So flip the RJ45 Over
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory Junker
Sent: Thursday, 28 April 2005 4:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [As
Hi Guys,
Sorry to be a pest but does anyone have any ideas? its really strange.
My 7290 (sccp) can not dial out to my SIP phones. The SIP phones ring but
when I answer them the 7290 does not recognise it and keeps making the
ringing noise. Even when I press hangup on the 7290, the display cha
> How can I have asterisk ignore incoming rings so it doesn't answer a
> specific line. I tried setting up an empty context section but that didn't
> work.
What I did is this
[incoming-line-noanswer]
exten => s,1,Hangup
Works perfectly
Or, if you want your CDR to have the callerid, do it like
I'd given up trying to get sipgate working with my * server, but given some problems I've had with a voip provider I
need to revisit this again. I've tried to set things up as per sipgate's website, I've read the info on voip-info and
from several other postings from other lists (and this one).
Hello,
I have had asterisk running nicely for months. Now I have a need to
intergrate with ser and rtpproxy. I have read all the literature I can
find, but keep having the same problem. rtp proxying.
sip (public) --> ser:5060 (public and private) --> asterisk:5061 and rtp
(public and private)
Really
looks like having a central SQL server is the best way.
Run it
on a separate, dedicated machine with an IP address all of the others can
see.
James
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Paul
DracevichSent: Wednesday, April
youssef ouadou wrote:
Aucune connexion n’a pu être établie car l’ordinateur cible (server)
l’a expressément refusée at System.Net.Socket.Connect(EndPoint
remoteEP) at IPS.listener.Start()
Hmm, I think you have two incorrect languages here; first, asterisk is
written in C, and your errors seem t
Title: Normal
To expand upon my original question, does
anyone know of any devices that would make connectivity between the Panasonic
system and Asterisk possible? What are opinions of using FXS ports in Asterisk
going into to CO ports on the PBX? Or if I’m putting money into the
problem,
Wanting to track all calls made to
(through) individual servers and bill a single customer based on ANI
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Dracevich
Sent: Thursday, April 28, 2005 10:08 AM
To: 'Asterisk Users Mailing List -
No
Yes that’s it.
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jltaylor
Sent: Thursday, April 28, 2005 10:28 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] CDR
Billing Question.
Are you talkin
Hi,
I’ve a problem with the capi channel. I’ve a SIP
phone and a CAPI card configured in asterisk.
CAPI -> SIP : working
SIP -> SIP : working
SIP -> CAPI : It’s ringing on the called party
but I’ve no sound. I’ve tried codec ilbc and ulaw with no success.
Does anyone have an i
Are
you talking about tracking a single call through three servers, or are your
wanting to track all calls made to (through) individual servers and bill a
single customer based on ANI?
James
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Pau
Hi Paul,
You can specify central
SQL database with cdr_. Application and put all records into one table or one
database.
The records will be
stored imediately after call, you could do even on-line billing
R.
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Sorry - still forget to clear the HTML flag - thought I'd better post again.
To add something to a post of a few days ago on this:
We're just putting in an asterisk system and wanted to have our own messages.
We're Asterisk and are not yet live but the following works.
Our PA simply has a list
I have three servers and I want to be able to bill a call
going from one through all of the serves.
The problem is that I am unable to link or pull the
data from each server cdr record and have a common
bill.
I have been looking on google, but anyhelp would be great
Regards
Pa
Dan Morin wrote:
Normal
My company has an old
Panasonic KX-TD1232 phone system
that they are using. I want to interface my Asterisk box with this
system for
a good conferencing solution. I have two X100P clone cards in my
server for
testing. They are hooked up to the analo
Is there anything I can do to reduce/remove the amount of errors I get when a
fax comes in for a SIP/Cisco ATA186 port? I'd like to make use of the fax
detection, but haven't seen an example usage of the fax extension that works.
Here's what I see when a fax comes in now:
Apr 27 16:38:
How do you tell asterisk to make a data call? I need to dial a
Pipeline 50 ISDN modem to setup a ppp connection, but it wont take
voice calls. Needs to be a data call. I have a pri line.
I'll give you a dollar if you can let me know by Thursday.
Thanks,
Jason
_
My configuration (./ configure) went smooth on Solaris 10 x86.
Then I got an error after running make command.
# pwd
/opt/home/srao/pbx/cvs-1.11.20
--
# make
make all-recursive
Maki
FYI To All, I fixed my problem by doing a Linux upgrade by typing Yum
Update and taking the CentOS updates. My problem is solved. I have no
clue what was going on but now I have Audio Now.
Michael D Schelin wrote:
Asterisk 1.0.7 built by [EMAIL PROTECTED] on a i686 running Linux
[EMAIL PROTECTE
Title: Normal
Arrrgh. I'm
hoping to do the same, except in my case I have 2 Toshibas that I plan to
connect too. The Toshibas can at least send a DTMF "D" upon disconnect,
though Asterisk can't do anything with this. Yet. I was hoping
that the Panasonic wouldn't have the same problem, bu
Chris, Try upgrading Lenux. I did mine with Yum Update and now I got voice.
Chris Stinson wrote:
Has anyone else had this issue?
Original Message
Subject: [Asterisk-Users] Voicemails stopping
Date: Tue, 26 Apr 2005 13:04:55 -0500
From: Chris Stinson <[EMAIL PROTECTED]>
Reply-To: A
To add something to a post of a few
days ago on this:
We're just putting in an asterisk
system and wanted to have our own messages.
We're Asterisk and are not yet live but the
following works.
Our PA simply has a list of the
extract from extensions.conf as shown below,
calls a number e.g
You can plug an RJ-11 in that very carefully.
:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Shiflet
Sent: Wednesday, April 27, 2005 2:03 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RJ45 to RJ11?
I just received my TDM400
On Wed, 27 Apr 2005, Wiley Siler wrote:
> Actually, I was recently told that the callwaiting disable feature in
> SIP 1.4.1 was not working.
> Are you using 1.4.1? If your method works, it would be useful to the
> other Polycom users I am sure.
> I love the conf scripts for Polycoms but it is f
Any idea on what this error message means?
Don't know what to do if second ROSE component is of type 0x6
I get that when I dial into my asterisk box. I'm using CVS HEAD of
ZAPTEL, LIBPRI and ASTERISK, though I tried it with v1.0.7 of ASTERISK
and I still get the same error.
Thanks,
Jess
On 4/27/05, Paul Shiflet <[EMAIL PROTECTED]> wrote:
> I just received my TDM400 card from digium with 2 fxo and 2 fxs
> interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS
> phones. How do i interface my POTS phones with this; can i just crimp an
> RJ45 connection on the end of the
Tom Ivar Helbekkmo wrote:
Leo Ann Boon <[EMAIL PROTECTED]> writes:
My prediction: 2 years down the road, they'll leave again and set up
SipZilla to make another low-cost ATA to compete with the SPA-.
...and then they'll sell *that* to Cisco, too. :-)
Or, 2 years down the road, VOIP will be
My question pertains to having people from random IP addresses log
in to my Asterisk system using SIP.
I would like to be able to have a SIP phone log into my system using
nothing but a username and password.
Based on the username the phone would be placed in a context and be
asigned an exten
I have the WIP-5000 phone. working great.
Anyone know how to put it in 12 hour mode?
The time on hte screeen is showing int 24 hour mode.
Cant get used to it...
I've looked all over the pdf files. have not found it.
Thanks,
Jerry
___
Asterisk-
On 4/27/05, Guy Boehm <[EMAIL PROTECTED]> wrote:
> Hello,
>
> I want to call a peer over the Asterisk Manager with this php-script:
>
>
>
>
>
> $socket = fsockopen("192.168.204.44","5038", $errno, $errstr,
> $timeout);
> fputs($socket, "Action: Login\r\n");
> fputs($socket, "UserName: te
If you need to, just go to Radio Shack and buy the 2 RJ-11 to RJ-45 adapter.
They're $5.99.
Justin Newman
Newman Telecom, Inc.
---
Message: 29
Date: Wed, 27 Apr 2005 14:40:26 -0400 (EDT)
From: "Jon Pounder" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] RJ45 to RJ11?
To: "Asterisk Users M
Title: Normal
My company has an old Panasonic KX-TD1232 phone system
that they are using. I want to interface my Asterisk box with this system for
a good conferencing solution. I have two X100P clone cards in my server for
testing. They are hooked up to the analogue phone ports on the ba
Jorge Mendoza wrote:
Mark,
Could you please post the models of your first and second mobo?
Thanks
The first, that didn't work correctly the the TE400P was an Asus P4 2.4
Ghz. The model that does work correctly is an AOpen P4 2.0 Ghz.
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Asterisk-Users
Asterisk 1.0.7 built by [EMAIL PROTECTED] on a i686 running Linux
[EMAIL PROTECTED] version 0.9
CentOS release 3.4 (final)
Linux 2.4.21-27.0.1.EL
Hi All, I really need help on this. What would keep Asterisk from
playing out audio files using the (Playback command) but I can play the
busy tone . pl
Has anyone else had this issue?
Original Message
Subject: [Asterisk-Users] Voicemails stopping
Date: Tue, 26 Apr 2005 13:04:55 -0500
From: Chris Stinson <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Organization: ISDN-Net, Inc.
To: asteris
Robert Webb wrote:
On Wed, 27 Apr 2005 11:24:24 -0600
Andrew Elchuk <[EMAIL PROTECTED]> wrote:
Hi,
I have two of the above installed into a server running Asterisk on
Debian Linux. Currently, only two phone lines are connected to the
system. I had both phone lines plugged into the one card, an
Thanks for all the great replies everyone. I've plugged many RJ11's in
RJ45s...just one of those days i guess, my brain isn't working too well.
THanks for all the help guys.
Paul
> The RJ11 plug fits perfectly into an RJ45 socket and only cares about
> the center-most conductors, which are the
All,
After looking at the wiki and some sample sip configs, I'm a bit
confused as to put in the sip.conf for externip and localnet.
My SIP client (xlite) is connected to the Internet and behind a firewall
on LAN1 with no ports forwarded.
My * server is connected to the Internet and is NAT'd on L
Actually, I was recently told that the callwaiting disable feature in
SIP 1.4.1 was not working.
Are you using 1.4.1? If your method works, it would be useful to the
other Polycom users I am sure.
I love the conf scripts for Polycoms but it is fun chasing down fields
in the XML sometimes.
Thank
On Wednesday 27 April 2005 2:02 pm, Paul Shiflet wrote:
> I just received my TDM400 card from digium with 2 fxo and 2 fxs
> interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS
> phones. How do i interface my POTS phones with this; can i just crimp an
> RJ45 connection on the end of
> > Unfortunately the otherwise excellent Areski stat tool doesn't seem
> > to include the unique ID function and thus I can't pull a file back
> > directly from that tool
If you want Areski stat to store the unique ID, you have to modify the
makefile as explained in the installation document on th
> I have 2 Gnet SIP phones connected on the same switch as the Asterisk
> box. So far, our phones authenticate with *, because when I do "sip show
> users", I see our 2 phones there.
When you say that you see them, does it look something like this :
501/501172.16.1.201 D
Simply plug in your 6
position modular plug into the 8 position jack, and it will work. Only
the two center pins are used, and both 6 and 8 position ( incorrectly
called RJ 11 and 45 ) will connect through on those two pins.
If the TDM400 card will work is an entirely different discussion.
If y
The RJ11 plug fits perfectly into an RJ45 socket and only cares about
the center-most conductors, which are the ones with the connection to
the PSTN.
Mojo
Paul Shiflet wrote:
I just received my TDM400 card from digium with 2 fxo and 2 fxs
interfaces. They are all RJ45 ports as opposed to RJ11 li
Hi Paul,
An RJ-45 is designed to take an RJ-11 or RJ-12 connector as well. Just plug
them in.
Ian
>>> [EMAIL PROTECTED] 27/04/2005 14:02 >>>
I just received my TDM400 card from digium with 2 fxo and 2 fxs
interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS
phones. How do i inter
> I just received my TDM400 card from digium with 2 fxo and 2 fxs
> interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS
> phones. How do i interface my POTS phones with this; can i just crimp an
> RJ45 connection on the end of the phone cord?
either that or just plug an rj11 into
Hi
In sip.comf you should add
Externip=
Localnet=10.0.0.0/255.0.0.0
In the sample sip.conf these two lines are commented out
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On Wed, 27 Apr 2005 13:02:56 -0500 (CDT)
"Paul Shiflet" <[EMAIL PROTECTED]> wrote:
I just received my TDM400 card from digium with 2 fxo
and 2 fxs
interfaces. They are all RJ45 ports as opposed to RJ11
like my POTS
phones. How do i interface my POTS phones with this; can
i just crimp an
RJ45 co
RJ-XX connectors are self-centering. You should be able to just plug you
RJ11 connectors in.
On Wed, 2005-04-27 at 13:02 -0500, Paul Shiflet wrote:
> I just received my TDM400 card from digium with 2 fxo and 2 fxs
> interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS
> phones. How
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