I would not recommend the 3Com phones for use with Asterisk.
3Com 3100 series phones do not support SIP with non-3Com systems. They have
a basic boot loader which must download code from a 3Com NBX or a 3Com VCX
system. If you don't have either of these, then you won't get runtime code
on the
Hi there,
Thanks for the tip ! I am happily using this service now.
One question though : I cannot get DTMF to work. Is there anything I can
do in my asterisk setup to fix this ?
Thanks,
Lukas
trixter aka Bret McDanel wrote:
http://www.trxtel.com/index.php?page=Tollfree_Termination
I'm looking for some cli output at the very moment a call on hold got
dumped.
[]'s
MM
-Original Message-
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Sat, 25 Mar 2006 22:55:28 -0800 (PST)
ok got it tnx guys!On 3/26/06, Dovid Bender [EMAIL PROTECTED] wrote:
Yes,There are issues witht he latest kernal release. Search the list archives. Dovid
Mark Quitoriano [EMAIL PROTECTED] wrote:
Hi Guys,Im having a problem compiling zaptel
1.2.4 on CentOS 4.3, anyone encountered this
What sort of call path are you trying to get working?
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: Rudolf Ladyzhenskii [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, March 26, 2006 10:18
Hello list,
Another newbie question,. if I put disallow=all and allow=g723
my sip.cof does it mean that extension could only communicate using
g723 ?
bellow is one of my extension example
[10112]
username=10112
type=friend
secret=x
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
Hi Guys,
I'm writing an app that receives a call on an incoming channel (A), the
caller
negotiates through a series of prompts and is transferred to an outgoing
channel (B) using the Dial cmd. That part works perfectly!
For billing I'd like to be able to charge for the time that the first
On 25 Mar 2006, at 19:15, Douglas Garstang wrote:
Why do I need a username at all if I am doing rsa authentication?
Why doesn't it match against the key?
So you want the receiving asterisk to take an incoming key and
speculatively see if it
matches _any_ of the keys mentioned in it's
Maybe you are better off with dundi ?
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.info
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Why is it drug addicts and computer afficionados are both called users?
___
You should put something between answer and dial, to let * have time to
get the fax tone: with your dialplan, the call is immediatly bridged to
SIP/3000, so no fax detection happen at all.
A thing like:
exten = s,1,Answer
exten = s,2,Playtones(ring)
exten = s,3,Wait(3) ; if fax tone comes here, *
Look, you don't have to necessarily specify a username when Dial(.).
It's sufficient ti specify the username in the peer declarations:
On pbx1:
[pbx2]
type=friend
username=pbx1 ; this is user for OUTGOING connections
host=w.x.y.z
inkeys=pbx2
outkeys=pbx1
.context=
[pbx3]
On Sat, 2006-03-25 at 09:41 +0100, Dave Cotton wrote:
On Sat, 2006-03-25 at 11:52 +0330, Paradise Dove wrote:
hi,
i've just upgraded to latest trunk. everything compiles fine but when
starting this message appears and fails to start.
WARNING[3990] loader.c: module chan_zap.so error
why should I? i thought in 2.6 kerneles that was not necesary when you dont have physical internfaces on the system.On 3/26/06, Jonathan Augenstine
[EMAIL PROTECTED] wrote:
Have you verified that ztdummy is loaded?On Sun, 2006-03-26 at 01:06 -0500, Erick Perez wrote: Hi, using asterisk 1.2.5 with
Hi gang. Just put an FXS port on a Zap interface for the first time. I
can't figure out which parameters in zapata.conf are global and which
ones can be channel specific nested. I have mucked around with it but
I can't seem to make any effect on the gain levels on a per channel
basis.
For us it takes about 6 seconds to detect fax tones, so you should change your
dialplan to either play and audio while detecting fax tones
(NVBackgroundDetect) or Wait for at least 6-7 seconds if using a Zaptel
channel.
On Sat March 25 2006 08:56, Thys de Wet wrote:
Hi There.
I have the
Erick Perez wrote:
why should I? i thought in 2.6 kerneles that was not necesary when you
dont have physical internfaces on the system.
ztdummy is still required for timing if you are using applications like
meetme, even under the 2.6 kernel.
Doug
--
Ben Franklin quote:
Those who would give
On 3/26/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I'm pretty sure I already know the answer to this, but...
Is there a way to copy/transfer/replicate sip subscriptions from one asterisk
system to another, for the purposes of HA? You coudln't even write a script
to do it I don't think.
On Saturday 25 March 2006 14:15, Douglas Garstang wrote:
Why do I need a username at all if I am doing rsa authentication? Why
doesn't it match against the key?
I agree that it's suboptimal, but the IAX2 spec (at least as I understood it)
REQUIRES a [EMAIL PROTECTED]
I think it's silly too,
Hey guys,
I've been struggling with hangup detection on a Centrex system for a bit
now, I was on site Saturday and I took my DMM with me.
It would appear that this Centrex service provider uses polarity
reversal at the beginning of a call and several times after hangup.
I've been reading
Thanks to SER, each of our Asterisk servers knows the address of every phone.
Any asterisk system can terminate a call to any phone. So, all we would need
would be for the asterisk system that terminates the call to also have a copy
of the subscription, and voila... it sends a NOTIFY back to
Hello,
I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1
FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5
dialplan.
I have connected an analog phone to TDM FXS port, but when I pickup the
phone to make a call, Asterisk hangs up the call. Let me
Hi,
using [EMAIL PROTECTED], with quadBri from junghanns.net I am facing a
strange problem:
I have set incoming routes for some extension / DID:
[ext-did]
include = ext-did-custom
exten = 23,1,SetVar(FROM_DID=23)
exten = 23,2,Goto(ext-local,23,1)
exten = 57,1,SetVar(FROM_DID=57)
exten =
Drat, because the 3Com phones looked pretty good for the price. :)
Is there somewhere that has a compatibility list for Asterisk with
all the phones that are known to work/not work with Asterisk; since
apparently VoIP phone companies incorrectly state that they support
the SIP protocol (I
Hi Joseph,
With iax servers dispersed across the internet, you could still use the
below setup, it would work but it's not as secure as you would want it.
I would then have a context for each server and use the IP address deny and
permit statements.
Also, you can have 1 server with a public IP
Hi Daniel,
If you are not locked in to an asterisk solution, I have a friend I
have done a couple of network/phone systems with. I am also looking at
Asterisk but have not gotten into it that far.
Rich Radcliffe
Kondor Waffenamt
(760) 240-4728
[EMAIL PROTECTED]
[EMAIL
samsung, and others, I have a list, email me to remind me to dig it
out and post to the list
On 3/24/06, James Harper [EMAIL PROTECTED] wrote:
Now that I actually try and google for it, I can't find any dual mode
GSM/DECT handsets, only pages telling me that they exist without any
actual
Look at the Linksys SPA942, it's a great phone for the price.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Radcliffe
Sent: Sunday, March 26, 2006 10:21 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] 3Com Phones
Hi
I found a solution... I just has to enter an Answer
line and now it behaves as I wanted. Here is the
working code:
[inbound]
exten = 1234567,1,Set(GROUP()=limit)
exten = 1234567,2,GotoIf($[${GROUP_COUNT()}2]?103)
exten = 1234567,3,Dial(Zap/5Zap/6,25,tT)
exten = 1234567,4,Voicemail,u110
exten =
3Com is one of the few that lie about it. Many Cisco phones support
SIP, but not all of them. I think Nortel also lies about SIP on some of
their phones.
Daniel Hazelbaker wrote:
Drat, because the 3Com phones looked pretty good for the price. :) Is
there somewhere that has a compatibility
If you can find yourself a local Asterisk consultant, they should be able to
let you see some phones and maybe even try them out.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: Daniel Hazelbaker [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
If you find anything out, I would like to know.
I have tried to find a gsm/wifi phone in the past (in melbourne) and failed.
later,
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: James Harper [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Alright, I've come across a really strange issue and I've been banging
my head trying to figure it out.
I have 3 machines. 1 Dell Dimension 4100, Pentium 3. 1 Dell 400SC,
Pentium 4. 1 Dell 1600SC, Xeon. I run mpg123 0.59r on each machine.
Using RH9 with a 2.4.20-8 kernel, each machine plays
Hi,
Can someone explain to me how to set up the sip_buddies
table from 1.2.5 properly so my users can authenticate correctly
without using the name field ? (if it's possible)
First I was assuming that it would be possible for a user
to connect and dial just providing username,secret,host and
i wrote previous about a setup i thought might work with asterisk and
the tsu-600. no one replied, so i thought i would ask if anyone is using
a tsu-600 with asterisk and if so how do you have it setup ??
___
--Bandwidth and Colocation provided by
What about using system(echo) to push stuff into a text file, or the mysql
plugin to push stuff over to a database?
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: Clint Tevlin [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Melcon Moraes napisał(a):
supertones=pl inside your unicall.conf
Ok, done.
something missing, isn't? What are you trying to do?
Maybe that will be better (thz Marcin):
# cat call
Channel: Unicall/1/363
Application: playback
Data: demo-thanks
klaudia*CLI !cp call
I'm using a Digium TE411P connected to a UK switch (EuroISDN).
Everything is working, but if I dial a busy number (from SIP) is seems
to stay busy until I hang up, even though the dial-plan drops through
some other stuff using CALLSTATUS variable (i.e. S-BUSY), none of the
timeouts come into
Daniel Hazelbaker wrote:
Drat, because the 3Com phones looked pretty good for the price. :)
Good for the price? You can import an atcom AT-320 EE for $40 +pp
(although they are hardly fantastic phones, at least they support IAX2).
They have a few faults (the speed-dial keys aren't really
On Sun, 26 Mar 2006, mike webb wrote:
i wrote previous about a setup i thought might work with asterisk and the
tsu-600. no one replied, so i thought i would ask if anyone is using a
tsu-600 with asterisk and if so how do you have it setup ??
The Adtran TSU-600 can be made to work like a
Ina previous message I described how Linksys 942 phone users who dialed in to a meetme conference at their site heard severe jitter. This was also experienced with Sipura SPA-2002 ATAs. Users of other IP phones like Polycom and Snom had no such problem. Also, the Linksys and SPA users had no
On Sat March 25 2006 18:06, Nick Hoffman [EMAIL PROTECTED]
wrote:
Hi guys, I've been using a Polycom IP 301 for a couple of weeks now
and find that it's extremely slow for configuring. For instance, it
takes several minutes to boot up, apply any changes via the web
interface takes at
mike webb wrote:
i wrote previous about a setup i thought might work with asterisk and
the tsu-600. no one replied, so i thought i would ask if anyone is
using a tsu-600 with asterisk and if so how do you have it setup ??
___
I have three working.
Easy to configure, lots of options and features, excellent quality speaker for
hands free. Although the 301 is nothing to get excited about. The 501 and 601
are much better.
Doug.
-Original Message-
From: Nick Hoffman [mailto:[EMAIL PROTECTED]
Sent: Sun
Now we're talking. :)
I don't know anythig about Alcatel boxes, but can you make a simple call
to your regular phone number from some SIP/IAX2 softphone/hardphone?
What do you have in zaptel.conf file? For instance, I have this on my:
span=1,1,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
And those
Nick Hoffman wrote:
Hrm, well that's disappointing. If they're so slow, why are they so
popular?
They may be slow to startup, but they're great phones. :) Once the phone
has started up, it works like a charm and the sound/call quality is
fantastic.
--
National Manager - Special Projects
Not GSM/DECT but GSM/Wifi phones are available - This is not a
recommendation, I don't like what I've seen.
try www.imate.com (to start with) .. they have at least three types of GSM
phones that do Wifi .. They run windows so there are several sip softwares
and one IAX software that work with
I think the main issue for James and myself is that we can't buy anything in
Australia.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: AR Tarzi [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday,
And the fact that rebooting a phone is a fairly rare occurence.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: Avi Miller [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Cc: [EMAIL
Hi Guys,
I'm writing an app that receives a call on an incoming channel (A), the
caller
negotiates through a series of prompts and is transferred to an outgoing
channel (B) using the Dial cmd. That part works perfectly!
For billing I'd like to be able to charge for the time that the first
Melcon Moraes napisał(a):
but can you make a simple call
to your regular phone number from some SIP/IAX2 softphone/hardphone?
Right now? It's not working.
I could use (i have needed hardware).
Data: SIP/extension-number
Or even capi/zap channel.
When e400p is configured as a E1/PRI and
Hi,
This is a weird request, but does anyone have a Snom 360 monitoring
extensions for BLF on several Asterisk servers accross a network?
Alternatively, can anyone give me a pointer as to how to setup a Snom
360 to monitor an extension not on it's own server?
My scenario is that I have a
I have a bad feeling that getting a phone with 160 lights is not going to
happen anytime soon.
From memory, the snom360 is limited to way less than that.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: Stuart Elvish - Dallas Delta Corporation Pty Ltd
[EMAIL
I installed 2 Snom360's a few months ago, and 'at the time' only 1 expansion
module could be added.
(also the fact that the modules draw so much current that it got the POE
switch upset!)
Have you tested a snom360? I should have one in the lab soon enough.
Paul Hales
Technical Manager
The worst thing on all Polycom IP phones is the speaker phone's poor
quality. You could not have a conference call using the speakers,
only the head phone.
Denis.
On 26 de mar de 2006, at 21:17, Avi Miller wrote:
Nick Hoffman wrote:
Hrm, well that's disappointing. If they're so slow,
I had a look at the snom website - and the manual for the expansion module
read that only one module can be attached 'currently'.
So maybe this has changed. Any ideas?
Personally, I like snom phones a lot. I used a snom 200 at my desk at a
previous job for almost 2 years.
Paul Hales
Technical
Now that's an interesting comment - most people think the speakerphone on
the Polycom is quite good.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: Denis Galvão - iSolve [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
AR Tarzi wrote:
Not GSM/DECT but GSM/Wifi phones are available - This is not a
recommendation, I don't like what I've seen.
try www.imate.com (to start with) .. they have at least three types of
GSM phones that do Wifi .. They run windows so there are several sip
softwares and one IAX
Denis Galvão - iSolve wrote:
The worst thing on all Polycom IP phones is the speaker phone's poor
quality. You could not have a conference call using the speakers, only
the head phone.
WHAT! The Polycom phones that have speaker phone features (the 50x/60x)
are great speaker phones. The 301
Nick Hoffman wrote:
On Sat March 25 2006 18:06, Nick Hoffman [EMAIL PROTECTED]
wrote:
Hi guys, I've been using a Polycom IP 301 for a couple of weeks now
and find that it's extremely slow for configuring. For instance, it
takes several minutes to boot up, apply any changes via the web
interface
I'm looking for a good web based voicemail client that can use mysql or
realtime drivers. I can't seem to get vmail.cgi to work with realtime.
Thanks for any help you can give.
___
--Bandwidth and Colocation provided by Easynews.com --
Not GSM/DECT but GSM/Wifi phones are available - This is not a
recommendation, I don't like what I've seen.
It strikes me as really strange that GSM/Wifi would be available while
GSM/DECT is not so much. DECT is a voice technology, while wifi isn't.
Still... there's a lot about the world I
Ari?
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
-Original Message-
From: Hall, Eric M. [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 26, 2006 9:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Web based voicemail client
I'm
Polycom is king as far as speakerphones IMHO.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 26, 2006 9:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Understanding..is not required. ;)
PaulH
- Original Message -
From: James Harper [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 27, 2006 12:23 PM
Subject: RE: [Asterisk-Users] GSM/DECT handsets (was
If your talking about Asterisk Recording Interface this is what I found
on the web site
Submitted by dan.littlejohn on Wed, 12/28/2005 - 5:34am.
ARI does not support realtime yet. It is coming
Nice app but just can't do what I need it to.
-Original Message-
From: [EMAIL PROTECTED]
Dr. Michael J. Chudobiak wrote:
I'd like to be able to use my Snom 360 LEDs to view the status of
parking slots, so I'm trying to install the metermaid patch
(http://bugs.digium.com/view.php?id=5779). Can someone help an svn
newbie figure out how to install this patch? I've done the following:
Well well,
This is not right at all. You should have like 1001 for TxABCD and 1011
for RxABCD.
That's why you are getting blocked. Something called my attention:
Bipolar Violation: you have some of it, which confirms that there's
something wrong.
Indeed, your call file is ok. That's why I
Should be posted to the -biz list?
Steve Ducat wrote:
CHINA DID
I am once again in search of China DID's. Either Shanghai (021) or
Guangzhou (020).
Please advise if you can supply.
Steven Ducat.
___
--Bandwidth and Colocation provided by
Great! Thx a lot Paul, I guess this applies to all sipuras right?
Anybody knows if this can also be done with polycoms?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Paul Hayes
|Sent: Wednesday, March 15, 2006 5:34 AM
|To: Asterisk Users Mailing
Release Notes for recent snom360 beta firmware:
Release 5.5.1:
o GUI: fixed consultative Xfer with fkeys
Release 5.5:
o GUI: fixed cursor handling (scrolling, backspace) in edit number state
o GUI: put last active call on hold on top in holding/transfer
Release 5.4:
o GUI: added shared line LED
On 27/03/06, James Harper [EMAIL PROTECTED] wrote:
Not GSM/DECT but GSM/Wifi phones are available - This is not a
recommendation, I don't like what I've seen.
It strikes me as really strange that GSM/Wifi would be available while
GSM/DECT is not so much. DECT is a voice technology, while
Hi John, yes, Im using native transfer. What I do is use Monitor on the
dialplan of the extension that picks up the call coming from PSTN, so after
that, if the extension forward or transfers the call, monitor keeps
recording all thru the end of the call no matter where it is been
transferred to.
To answer my own question :)
Yes, it can be done by way of using polycoms magic config files:
On phone1.cfg for all phones or MAC ADDRESS.cfg for a specific phone:
call.autoOffHook.x.enabled=1
And
call.autoOffHook.x.contact=EXTENSION TO CALL
|-Original Message-
|From: [EMAIL
Man! I love those phones Great speakerphone, great functionality, works
great with asterisk
kiss ass mode
Any polycom reps here?
/kiss ass mode
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Anton Krall
|Sent: Sunday, March 26, 2006
Denis Galvão - iSolve wrote:
The worst thing on all Polycom IP phones is the speaker phone's poor
quality. You could not have a conference call using the speakers, only
the head phone.
Huh. Polycoms have the best speakerphone I've ever used on an IP phone. :)
--
National Manager - Special
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Marty,
But with the same 128 bit upstream circuit, directly connecting the SJPhone
the Stun server and using ulaw, everything is perfect. The problem comes
when i am putting Asterisk in the picture.
I have used SJ Phone softphone. His
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
i can confirm that this exist on 1.2.5, and the last time i said this, the
original poster was supposed to file a bug on bugs.digium.com.
OK. Can anybody else confirm this? I don't wona report it if it isn't bug.
--
Tomislav Parcina
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
It's a toll free number. You can call it from anywhere and the costs of the
call go on the callee not the caller.
Thank you.
--
Tomislav Parcina
tparcina#lama.hr
___
--Bandwidth and
I have running Asterisk 1.2.5 with addons 1.2.1. on Fedora Core 4. I have
installed ooh323 from 1.2.1 addons.
How to upgrade addons to 1.2.2 version and install new ooh323 driver? Do I need
to install addons 1.2.2 if I only need new ooh323 driver?
Can I just untar addons, and run make clean;
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
That's good to know... this only affects 8.2, right?
As far as I know, yes. I have been using 7.5 and now I use 7.4 on 7940 and 7960
and I didn't have those issues.
--
Tomislav Parcina
tparcina#lama.hr
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I have setup several queues for a customer. Their periodic announcement says
please wait for the next available agent, or press * to leave a voicemail.
This does not work when the message is playing. The message stops, but the
user is
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Sorry for being a bit of a newbie here but I find the
docs or README for downloading the G.729 codec from Digium
are not as detailed as I would like or just don't really
break down the different versions to a point that I am clear
on
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
now, i just edited the Makefile that comes in zaptel directory to disable
any usb, as i am not going to use any usb device in my asterisk, and it
compiles and work ok.
Hi Raul!
Please send us what lines did you comment. Does it work with
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hello,
I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323
module and g729/g723 free codecs too.
Can you send us more information about this free g729 codecs?
--
Tomislav Parcina
tparcina#lama.hr
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Can someone tell me how I can configure ooh323.conf to accept call
from h323 gateway (only the authorized h323 gateway) to my asterisk.
Sorry, this is not answer to your question, but I need to ask you something.
Are you using ooh323
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I'm looking for a web GUI to offer my end-users (Hosted PBX), and I thought
I'd pick a few brains first.
I'm not looking to configure the Asterisk server itself, VI works adequately
for that. But I want to give Web access to as
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi !
What is now the difference between a:
reload - (cli command reload).
restart - (I assume the application asterisk is restarted. o.k starting
from new)
sip reload - (cli command sip reload). Is sip reload part of the
reload
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You have to set up a dialplan.xml file in your tftpboot directory for the
phone to pull:
DIALTEMPLATE
TEMPLATE MATCH=9,59. Timeout=0/
TEMPLATE MATCH=9,29. Timeout=0/
TEMPLATE MATCH=9,832... Timeout=0/
Please stop send me email
Best Regards,
Mr.Peeramate Rochanasmita
Project Manager/General Manager
SIPphone (Thailand) Co., Ltd.
644/19 Moo 1 Klong Kum,
Bung Kum Bangkok Thailand 10230
SIP No.100888
SIP Call Center No.888
Tel. 0 2690 3999
Fax. 0 2690 3535
Mobile. 0 1423 1423
Email : [EMAIL
Hi Aaron!
Can you tell me what , and \ stands for?
. changes only one number from 0 to 9, right?
* changes unlimited number of numbers from 0 to 9?
--
Tomislav Parcina
tparcina#lama.hr
Absolutely right :)
\ escapes the next character, so if you wants *69 to go through
immediately, you'd
Actually I've got five, but the first one I have received
around Xmas and I don't have these problems with it.
I use spa3000 as FXO and the gsm gateway works
seamlessly inbound, outbound, DISA, no annoying sounds,
no DTMF problems. There is one problem however, the gateway does
not transfer
Title: 7940 with Asterisk?
I just picked up a Cisco 7940 from an Auction and would like to use it on an Asterisk box.
Can anyone give me a pointer where I should start so I can get it working?
Skeeve
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Skeeve Stevens, RHCE Email:
Tomislav Parčina wrote:
Can you send us more information about this free g729 codecs?
There is no such thing as a 'free' G.729 - The DSP Group has claimed and
defended the Patents they hold against the algorithm and process.
Please do not use Asterisk/Digium related resources to exchange
Melcon Moraes napisał(a):
Well well,
Question: if you have PRI, why are you using R2?
Sugestion: check Alcatel's setup for this card. On Alcatel side, do you
have any alarms or errors?
I would like to build simple IVR soulution, whitch allows me to do
free B-channels. Now I could
My TZ is GMT-3. Just waiting for the lucky/magic numbers.
[]'s
MM
-Original Message-
From: Krzysztof Drewicz [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Mon, 27 Mar 2006 09:39:52 +0200
Delivered:
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