??
Thanks
On Wed, 2010-09-01 at 09:09 -0400, Barry Fawthrop wrote:
Has any advancement been made to get 3102 operational in either a SIP or
H323 asterisk environment.
A post back in time mentioned a downloader service.
From the posts and articles I have read, the NCP is acting like a bootp
On Wed, 2010-09-01 at 09:09 -0400, Barry Fawthrop wrote:
Has any advancement been made to get 3102 operational in either a SIP or
H323 asterisk environment.
A post back in time mentioned a downloader service.
From the posts and articles I have read, the NCP is acting like a bootp
and tftp
Has any advancement been made to get 3102 operational in either a SIP or
H323 asterisk environment.
A post back in time mentioned a downloader service.
From the posts and articles I have read, the NCP is acting like a bootp
and tftp server which uploads the configuration to the phone??
Am I
How does one redirect calls based on incoming number or caller ID or the
lack thereof?
current I have for number 123-4567 that it redirects all 800 , 877 and
866 numbers to Voicemail directly.
If the primary area code is 352 then accept this and pass it to
extension
exten =
In trying to get the asterisk and faxing working
I had to resolve to using iaxmodem and hylafax.
I have incoming working, but outgoing the other fax rings
but it would appear from web searches that the fax signals
are too low to be heard
I can read about rxgain and txgain for zapata.
my fax setup
Barry Fawthrop wrote:
Hi All
I have been looking around and haven not been able to find a working example
I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri
1.4.10.2
I use a sangoma A200 card so I am using wanpipe 3.4.7
If I use zaptel which I read I need
Hi All
I have been looking around and haven not been able to find a working example
I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri
1.4.10.2
I use a sangoma A200 card so I am using wanpipe 3.4.7
If I use zaptel which I read I need for app_rxfax then asterisk crashes
Kevin P. Fleming wrote:
Barry Fawthrop wrote:
I have been looking around and haven not been able to find a working example
I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri
1.4.10.2
I use a sangoma A200 card so I am using wanpipe 3.4.7
If I use zaptel which I
Hi All
I have a simple network with 2 IP phones, 1 Std. Phone (using a Siptura
2000) and a CG-410 Gateway to connect the the two PSTN lines that I have.
I have a odd hassle that for no apparent reason, the calls will quite
working.
but quite I mean the phones will ring but their is no voice
line through the TDM card to the fax
machine?
Right, this is possible also or use an ATA if not a TDM card ?/
Thanks All
Barry
Barry Fawthrop wrote:
Hi All
Curious will this work
Std. PSTN line ---x-- X100p
|
-- Fax Machine
Using a standard
Hi Gordon
Following you dial plan
How does Asterisk know to move from s,2, to either incoming,1, or fax,1,
The only jump I recognize it Goto(internal,incoming,1) which should take
all calls to incoming,1, and not fax,1,
OT: is spandsp rxfax handled by astlinux ?
Thanks again
Barry
Gordon
Hi All
Curious will this work
Std. PSTN line ---x-- X100p
|
-- Fax Machine
Using a standard home phone pstn line with a splitter connecting a fax
machine and X100 Asterisk Box
Incoming Line: Can I have in the dial Plan
[incoming]
exten =
,..
and what make it go to exten = fax,
How does this logic work??
Thanks again
Barry
Barry Fawthrop wrote:
Hi All
Curious will this work
Std. PSTN line ---x-- X100p
|
-- Fax Machine
Using a standard home phone pstn line
Hi All
Could a VPN be used to help with SIP Tunneling and QoS issues.
State 1:
Two IP Networks Connected via the Public Internet transmitting VoIP Traffic
Say a VoIP User and VoIP Termination Provider.
Each side can put QoS onto their part, but if QoS does NOT exist between
them
then call
Hi David
Care to share how you approached using Diffserv and VLANs with the FSM7326P
We are considering the same switch. But I'm unsure about the configurations
required.
Thanks in advance
Barry
David Coulson wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Typically we deploy the
will be relevant to OpenVPN, which might be a bit
different from IPsec, PPTP or other solutions.
Am Montag, den 11.12.2006, 17:26 -0500 schrieb Barry Fawthrop:
Hi All
Could a VPN be used to help with SIP Tunneling and QoS issues.
State 1:
Two IP Networks Connected via the Public Internet transmitting
Hi all
Is the use of a VPN between IP-PBX and VoIP Provider a useful tool?
Since the QoS and general traffic of the Internet can never be
predicted, would the implementation of a VPN between Client and VoIP
Provider increase voice quality and/or security or is the converse true ?
Thanks
Hi all
How much does configuring a network with VLANs improve or effect quality ?
Is there much reason to justify the configuration of VLANs ( I know
networking, but not VLANs at all)
Would it not be better to find high traffic users and determine why?
Your Thoughts
Thanks
Barry
what is the sip.conf for 1239
which I'm going to assume is a extension on the TNT
Barry
JR Richardson wrote:
Hi All,
I have a lab setup with two asterisk servers and a MAX TNT in the
middle like this:
asterisk sip sip TNT pri pri asterisk
The TNT is running 11.0.6 and the asterisk
Hi Kristian
How many are you going to have at tonight's meeting ?
Thanks
Barry
Kristian Kielhofner wrote:
Matt Florell wrote:
Hello,
We will be having another Tampa Bay Area Asterisk Users Meetup on
Monday, November 6th at 7:30 PM.
Asterisk users from gurus to new users are welcome.
Along
Hi All
Has there been problems with Caller ID and Asterisk 1.2.10 ???
I have a Phone Number from Teliax, I get this
chan_sip.c:10468 handle_request_invite: Failed to authenticate user
3523029577 sip:[EMAIL PROTECTED];tag=as03efd979
And the call does not go through.
I have a CG-410 FXO
Hi Kristian
What else is in the 0.4.4 release ??
Any news on the Sangoma A200 or Faxing ?
Thanks
Barry
Kristian Kielhofner wrote:
Hello everyone,
I have released AstLinux 0.4.4. Thanks to all of the testers on
astlinux-users, AstLinux 0.4.4 now includes mISDN support (again). Grab
On 10/24/06, *Barry Fawthrop* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi All
I'm trying to understand how I would send my fax ?
If I use Word or what ever word processor or even an email
client to
create what I want faxed.
I have *asterisk setup with and FXO
Hi All
I'm trying to understand how I would send my fax ?
If I use Word or what ever word processor or even an email client to
create what I want faxed.
I have *asterisk setup with and FXO Gateway that will make the call to
the fax number I dial
SIP extension 320 is the FXO gateway.
Hi all
Anyone got a Clipcomm CG410 working with Asterisk ???
Where incoming calls are passed on directly to asterisk and
outbound calls are passed directly to PSTN ??
I have tried configuring the gateway and I just can't get it to register
no matter what I try
If any could share
Hi all
I'm having a problem getting usable quality from my Asterisk setup.
*SETUP*
2 Ghz PC with 1 GB Ram with TDM 400p 1 x FXS to route to analog phones
in the house and 2 x FXO to receive calls and in the future faxes.
Gentoo Linux
Here is what I've done so far
(1) Moved theTDM 400p (FXS,
Hi all
Anyone using a sonicwall firewall ?
I have been and then suddenly it drops UDP packets because SIP is no
longer on port 5060 but some random assigned port ?
Why ?
Thanks all
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this nothing was changed it blink I'm
not going to work now
Thanks all
Barry
Dr. Michael J. Chudobiak wrote:
Barry Fawthrop wrote:
Hi all
Anyone using a sonicwall firewall ?
I have been and then suddenly it drops UDP packets because SIP is no
longer on port 5060 but some random assigned port
all the
VoIP SIP setting are enabled/configured and SIP is packet forwarded to the
Asterisk Box due to Sonicwall NAT
Now I'm trying to find out why and how to correct this.
Thanks all
Barry
Rich Adamson wrote:
Barry Fawthrop wrote:
Hi all
Anyone using a sonicwall firewall ?
I have been
Hi Francesco
Yes it is
SonicWALL Enhanced has an option called 'Persistent NAT'... Is it turned on?
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Thanks for the input
Yes I have nat=yes and qualify=yes I know in the SIPMacAddress.cnf file
I have
# NAT/Firewall Traversal
nat_enable: 1
nat_received_processing: 1
nat_address: phone's public IP Address
Do I still need to set it again in SIP Configuration ?
Thanks all
Barry
Hughes, Sam
Hi All
Yes I know double Nat is a problem
But I have a Cisco 7960 which is remote from the * PBX ad connected via
the Internet. Each side has NAT
(1) Sometimes it will work often it won't. And when it decides to work
is random
Always
(2) The Register side works fine. SIP SHOW PEERS has
If the phone already had the SIP image running.
Check the SIPDefault.cnf file there may be a phone_password= string
this is the phone's current password use it
remember to change to number or uppercase if need be
Ferguson, Michael wrote:
Maxx,
Thanks much for the feedback. I will check
behind NAT. You can try setting port forwarding on the phones
side as well as look at a better router. Some routers will make you
pull your hair out while others will work almost perfectly (this
explains my now bald head :) )
Dovid
- Original Message - From: Barry Fawthrop
[EMAIL
Hi All
I have a Cisco 7960 which is connected remotely to an Asterisk server.
Both are unfortunately behind NAT.
The Phone registers and is show in sip show peers, with the correct
public ip for the phone and a 100ms qualify time
(1) I can dial the phone from another phone, it will ring but
Thanks David
But what I was more looking for was storing the configuation file eg
extensions.conf as a database file in MY SQL and then have asterisk load
the table from MYSQL as opposed the text file extensions.conf ?
Is there any benefit in this ?
Thanks all
Barry
Hi All
Is there a benefit to using a database to hold the extensions and sip
.conf information/configurations
or is using the standard Text file just as good and no benefit is received?
Also how does one go about converting the text .conf files to a
database, and the have asterisk read it
Tried to install and get this
extension_dir does not exists
/usr/lib/php/extensions/no-debug-non-zts-20020429
when entering index.php
Any Ideas ?
Barry
El Flynn wrote:
Hello,
We've just released our Libero Management System application, a
web-based interface to configure and manage your
Thanks All for your replies
A couple have mentioned backup routes, as I'm clueless on backup routes
How do I program the dialplan in extensions.conf to:
(a) try multiple provider to make an outgoing call based on current
latency between my * box and the different providers ?
(b) have if
Thanks All
First off I never mentioned Teliax (but yes correctly ASSUMED they are
my provider) and this is not a Teliax issue per se
My issue is more the fact that I have Qualify = yes in sip.conf but
repeatedly get REACHABLE and UNREACHABLE
as can be seen below. even when I set Qualify =
Hi All
I am repeatedly getting a UNREACHABLE and then REACHABLE about 10 sec
apart most of the time and then sometimes for about 45 - 74 minutes
I have tried a reload and sip reload but neither bring the provider back ?
What else could I try and how do I prevent this
Thanks in advance
Hi There
I know of wiki there is a list of VOIP providers, but is there a list or
can we create / suggest one
that will list VoIP providers, their location and quality of service ?
Too me this will be very valuable, plus looking at some of the requests
of late I'm sure others would like that
I'm in the US, using cards bought direct from Digium.
I have lowered the rxgain and txgain to -8 and that seems to be helping
futher.
I wish I could understand why?
The problem with more time is that I can hear myself in the headset of the
std. phone
as well as the party on the other end. The
I'm having with an echo or delay
I connect to the PSTN with a x100p and then connect a std. phone
to a FXS module on a TDM10B.
The std phone is only 2-wire so I know this is not helping.
(yes I have read the 2-wire 4-wire issue)
I have tried many echocancel values. The best thing to help was
In having configured my first * server there are a few questions I could not
understand or find answers to
1) How does one use ztmonitor to adjust the rxgain and txgain. I have set
mine to -1.0 each to get rid of echo on std phones connected on the TDM10B
FXS module
2) Is it best to use a TDM
I'm also trying multiple lines
on my phones I have mapped P2 - P5 numbers 2502 - 2505
I have 2502 - 2505 in my sip.conf andextensions.conf
snom200 is using 2.05e
1] the LCD for p2 - p5 are always lit. When I dial 2503 the MWI flashes
none of the p1 - p5 flash :(I was
Greetings
I was getting bad static crackle on a phone, so I
reload from the latest CVS and did
a make clean ; make install on zaptel, libpri and
asterisk
Now I get this error
[chan_skinny.so] = (Skinny Client Control
Protocol (Skinny))May 28 13:59:42 WARNING[16384]: chan_skinny.c:2541
canreinvite=no
[general]
port = 5060
srvlookup = yes
nat = yes
tos = lowdelay
disallow = all
allow = ulaw
allow = gsm
allow = alaw
context= INVALID
Currently my IP phones haves this in the sip.conf
[4403]
type= friend
username
The problem is probably that both of your SIP phones are using the same
port. I played with two 7960's behind a Linksys on Saturday and finally
got them playing right when I changed the following:
In Phone 1's SIP[macaddr].cnf:
voip_control_port: 5061
In Phone 2's SIP[macaddr].cnf:
- Original Message -
From:
Christian
Stredicke
To: [EMAIL PROTECTED]
Sent: Thursday, March 25, 2004 10:05
AM
Subject: RE: [Asterisk-Users] IAX and
Snom200
We thought about
this option. I guess the IAX2 is not the problem. We believe the real
NAT traversal is a huge issue I agree with Michael and Brian
what with the latest viruses etc... security is and will be
more and more of an important issue, many SOHO and small corps.
Often don't have the know how or finanical backing to implement
standard/conventional security and internet
Greetings
I'm now really messed up::: setuptrying to by pass
NAT
*1 (bound to LAN addr)- Linux Gateway + *2
(bound to public addr)- Internet - IP Phone
A
*1 and *2 are registered and see each other fine
*1 has ext 13 SayUnixTime
*1 has ext 4403 an IP Phone on the LAN
From IP Phone A I
Greetings
What would it take to have a snom200 support
IAX,
what are the processes or having hardware
to
support a new codec? Can this be tested and
done
by a uesr or must this be done by the
manufacturer?
Thanks in
advance
Progress
It seems I can't hear the Say Time, due to RTP Double NAT
I'm guess this is both problem 1 and 2 really issue.
My config:
IP Phone - Router (Nat) - Internet - Linux (NAT) - * Server
ANyone know of work arounds the double NAT? or other methods
to route RTP with snom 200s, to work
Thanks to All who replied
I have tried all the steps above. and from the website given
I have two snom 200 next to each other
4403 and 4405
when I dial 4405 - 4403 nothing rings and
* CLI reports voicemail/default/4403/busy
when I dial 4403 - 4405 nothing rings and
* CLI reports
Thanks Willy and others
It seems I am not able to make myself clear on my two problems
I wish to try again, as I'm sure it is not the phones, but
some stupid config problem on my part. I repeat alot of
what I have said in order to try make myself clear.
I have a 4401, 4403 and 4405 in sip.conf
Greetings All
I'm busy trying out my new snom 200(s)
I have it connected and * CLI tells me registered
1) I pick up the handset and hear the dial tone
2) Dial and Ext, that says Date Time (13)
3) * CLI scrolls that the call is connected and time is being spoken
YET the handset is quite and
From: Olle E. Johansson [EMAIL PROTECTED]
snip
Check the CDRuserfield - it's a free field in the CDR you set in the
dialplan or from a script.
How would you set the CDRuserfield from the dialplan
exten = ?
Thanks in advance
B
___
From: [EMAIL PROTECTED]
Please include the sip.conf entry for the phone you have ..
SIP Configuration for Asterisk
;
[general]
port = 5060
bindaddr = 192.168.0.15
externip = 24.73.215.62
localnet = 192.168.0.0
localmask = 255.255.255.0
tos = lowdelay
disallow = all
allow =
Here's another funny
* CLI puts put
-- Registered SIP '4405' at IP.address Port 5060 Expires 3600
and within seconds the snomm 200 beeps the MWI goes on the LCD and the
light flashes a call from asterisk Not Found
Willy if you could let me see you sip and config files, if you have yours
working?
I would like to know how and where nufone and the other get their access
to provide termination, and yet only offer 2.9c /min
We are about to sign up with a termination provider and have read often
that people suggest nufone, yet for everyone who suggest them, they always
have had some sting in
check NAT setting try taking it out of sip.conf, that worked for me
Barry
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check context perhaps try include in the extensions.conf
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Greetings
What would it take (all hardware etc..) to setup *
on a
prepaid card server.
I have an * server a T1 and TDM10B card, thus
allowing 24
simultaneous calls I
guessing I need a VoIP Termination
Provider (eg: NuFone, etc..)
How do I print and create the cards, and what are
the
Even though it was 100, I'm also keen to hear
about large installs, what kind of experience
did you have setting it up, and what hardware for
the * server did you use?
Thanks in advance
Barry
Matthew wrote:
I've set up 75 extensions... I'm 100.
Sorry.
Sincerely,Matthew
Hi Matt
How do you reach a Iaxtel 1-700 from PSTN?
didn't think
this was possible ?
Barry
Greetings
Can a dial plan be set so that an * server
connected to a T1
can receive a fax and save it as a file in a
specific dir or have
the fax sent to a specific IP address, that will
handle the fax
and save it as a file?
Thinking along the lines of fax - email, but
within an * server,
What is the best or simplest method
to connect 4 fax
machines into a
* system?
Fax
-
ATA-186 -
Switch - * Server - VoIP or PSTN
Fax - * Fax Server with TDM 400P -
Switch - * Server - VoIP or PSTN
Would like a dedicated # on the T1
to do direct to the fax machine.
Would love your
- Original Message -
From: James Golovich [EMAIL PROTECTED]
James Golovich wrote:
I don't explicitly disable echocancellation on the channels I use for fax,
and zaptel always seems to detect the tone to disable echo cancellation
from the fax. I send/receive all my faxes over IAX2
Hi
How do I set up firefly to use the IAXTEL
network?
I have registered with IAXTEL I have a
username
and password.
I have tried IAX protocol and iaxtel.org ad the
network and nothing happens ?
Thanks
barry
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