Does Asterisk support pager notification of new voicemails out of the
box? Or do I need an AGI script to do that?
Yes, asterisk supports both email and pager notification out of the box.
This is actually pretty flexible, since you can customize the content of the
emails to be whatever you
Can anyone tell me how (and for how long) asterisk remembers the IP address
for an IAX2 peer? Voicepulse has been going up and down for me, and it seems
to have something to do with the IP address changing. Is there a way to
force asterisk to run gethostbyname() again for the peer? Or do I just
Ditto here. I can ping but not log in.
--Ernest
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, June 30, 2004 8:46 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 10:10am CST - VoicePulse
appears to
Title: Message
Does anyone know of a solution where I would be able to
setup some sort of permanent connection to the asterisk server via IP?
I
can't have a dial tone in their ears constantly and I need to find a phone or
solution which is $150 or less (preferably under $100) per
If you don't have CVS, then you probably also don't have the kernel source,
the development tools, etc. What Linux (hopefully) distro are you using?
--Ernest
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hermann Wecke
Sent: Thursday, May 27,
PS
Someone mentioned about some other problems with 2.05e. What kind of
problems are they ?
For me it would be important to know.
The biggest one I know of relates to the speakerphone. When you have the
phone set to ring on speakerphone but use headset to talk, an incoming call
will bump the
First, try moving back to 2.05c or earlier. 2.05e has a few problems
(remember, it's beta quality) that could be causing this. Second, are you
sure that the disconnect on hook or transfer on hook settings are the
way you expect them to be. That caught us for a while since we were putting
people on
to be an
improvement...
CS
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger
Sent: Thursday, May 13, 2004 7:35 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] 2.05a firmware
They also made a bad (for me) change
You know what would be cool? A Show Variables command in the cli. It could
return something like this...
VariableScope Channel
=
CallerIDC ZAP/1-1
EPOCH G
EXTEN C ZAP/1-1
This is pretty obvious, but have you logged into the phone to make sure that
the CWI is turned on?
--Ernest
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of nicolas
Sent: Friday, May 14, 2004 1:29 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
This happened to me as well. I resolved it by logging into
the web interface, going to the Advanced Networking screen and turning off
automatic updates. Then, I manually entered the firmware URL and updated through
the website (on the Updates screen). It took a few tries, but I think the
Does anyone know what kind of file needs to be uploaded for the custom ring
tone?
--Ernest
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Justin Huff
Sent: Thursday, May 13, 2004 10:09 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
They also made a bad (for me) change. In 2.05a the phone would ring
normally and I could press OK for headset or pick up the handset for
handset. Now, when headset is enabled the phone only rings in the headset
(i.e. not through speakerphone).
--Ernest
-Original Message-
From: [EMAIL
The newest snom firmware (2.05a) resolves this issue. It's not yet freely
available, but it is in the pipeline.
--Ernest
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Justin Carlson
Sent: Wednesday, May 12, 2004 10:29 AM
To: [EMAIL PROTECTED]
My overall impression with the SNOM 200 phones is quite good. Snom (or the
people at ABP) have helped me resolve most of the issues that I had with
them.
Good:
Five lines
Headset support for both 1/8 and RJ11 cables
Attended transfer and conference calling
Address book
Multiple rings based on
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mike Machado
Sent: Monday, May 10, 2004 5:10 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] alternative FXO gateway to
Mediatrix 1204?
On Mon, 2004-05-10 at 12:37, Ernest W. Lessenger wrote:
We use an AudioCodes MP-108
I see that your line signalling is set to kewlstart... Are you sure that
your telco provides this? Also, I found that I was having similar problems
when there were other devices on the line (like fax machines). The problem
usually occurred when someone tried to make an outgoing call on the same
Assuming that you have 1 - analog lines - 4 and that you want your phones
to be 100% VoIP (i.e. no Analog handsets): You should just need the new
Digium TDM04B bundle and the granstream phone(s). If you have 1 - analog
lines = 2 and 1 = analog phones = 2 then you can use the TDM22B bundle.
Can anyone recommend a FXO gateway product that does behave
in this more
correct manner?
We use an AudioCodes MP-108 and have been quite happy with it. NOTE: Make
sure you get the most recent software build, the one that came installed on
ours was REALLY old.
--Ernest
-Original
Any good ideas would be appreciated!
We use a package called Nagios to monitor our servers, which works quite
well. It has the ability to track service and host dependencies so you don't
get flooded with a bunch of service down alerts when the real cause is a
bad switch (or similar).
It would
What about using NFS or AFS for this?
--Ernest
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
NaySent: Monday, April 12, 2004 10:35 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Voicemail
storage in DB
Hey
all,
Quick Question. I
Does anyone have the physical spec sheet for the T100P from Digium? The one
on the website doesn't have what I need. Things like 3.3 or 5v operation,
uses n IRQ channels, requires 32-bit PCI, must be installed while standing
on one foot and reciting the GPL, etc. Also, if anyone is selling a used
having problems with snom phone installstion
Please tell us what's up. I recently installed several SNOM phones and
worked through many minor issues. Let me know and I'll tell you what I can
:)
--Ernest
___
Asterisk-Users mailing list
[EMAIL
What version of asterisk are you using, and what version of
the SNOM firmware?
--Ernest
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
jcSent: Tuesday, March 30, 2004 10:20 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Exception
flag set - snom200
At 11:39 AM 3/22/2004, you wrote:
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
At 09:52 AM 3/16/2004, you wrote:
I need to setup asterisk so that users can dial into asterisk using normal
phone lines and and enter a number when prompted then this number should be
accessable to a backend app. is this possible in asterisk. any pointer would
be helpfule
Yes, this is possible.
At 10:16 AM 3/16/2004, you wrote:
and what would i need to connect asterisk to 2 normal phone lines
You would need two FXO cards from Digium to connect to two Telco lines.
You would need two FXS cards from Digium to connect to two telephones.
Telephone - FXS + Asterisk + FXO - Wall (Telco)
If you
At 08:37 AM 3/11/2004, you wrote:
Music on hold works if the environment is noisy.
But in case of silence the sound goes off.
If I scratch continuously on the mikrofone, then the replay works without
any interruption.
Q: is there a parameter which influences this behaviour?
Whatever phone or
At 07:07 AM 3/2/2004, you wrote:
Ahhh, you must have upgraded to firmware version 4.2. I had the same
problem because
I didn't find the new parameter that they added in this release for
broken RTP connections.
Here is how I fixed it:
BROKENCONNECTIONEVENTTIMEOUT = 36
That did it, thanks!
: Ernest W. Lessenger [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 01, 2004 10:25 AM
Subject: Re: [Asterisk-Users] Tiny install with Solid State Storage
At 08:01 AM 3/1/2004, you wrote:
Hello All,
I was wondering if anyone is successfully running asterisk on a
system
with solid
card. There are
some other tweaks (like using tempfs for the /tmp partition) but otherwise
everything worked like a charm.
--Ernest
Thanks
-Matt
- Original Message -
From: Ernest W. Lessenger [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 01, 2004 10:25 AM
Subject: Re: [Asterisk
At 08:51 AM 3/1/2004, you wrote:
I have 5
BT phone lines coming into my office. We use four for international
calls, and one for local/mobile calls. We have just obtained another call
carrier, and now we would like to be able to make calls from any phone to
any carrier, without having to
We have an Audiocodes MP-108 that keeps dropping connections to voicemail
after exactly ten seconds. All other calls are normal, and voicemail works
fine from SIP devices other than the gateway. The reason given for dropping
these calls is RTP Connection Broken. I suspect that the gateway is
We have a situation where voicemail coming in (i.e.
FXO-Asterisk-Voicemail) through a Mediacodes MP108-FXO are getting cut
off a couple of seconds early. I recall a thread about this quite a while
back where this was happening due to silence detection on ZAP channels...
Has anyone experienced
At 05:26 AM 2/25/2004, you wrote:
I am in the middle of getting my self some hard phones. Anyone care to
comment on the *voice* quality of the following phones:
Cisco 7960
Siptone II
SNOM
Budgetone
I have seen a few reviews, but none go to deep into the voice quality
issue.
I have not received
At 05:49 AM 2/25/2004, you wrote:
The Snom 200 phone mostly functions well, however the phone's logic is more
oriented to european telephony and several of the functions do not work in
a manner that one might consider 'standard' in the US. It's light-weight,
pulls across the desk when the handset
At 09:21 AM 2/25/2004, you wrote:
Ernest W. Lessenger wrote:
At 08:15 AM 2/25/2004, you wrote:
This may sound silly but how can I say to asterisk that new number have
been dialed and that it has to treat these as a new extension ?
I mean: I have received a call, and now I want that asterisk
we went ahead and purchased an
additional 12 phones and the MP-108 gateway on the recommendation of our
salesperson. Our experience with them has been very good, and the
(uncompleted) RMA process of one defective unit has gone smoothly so
far.
--Ernest W. Lessenger
OACYS Technology
OACYS TECHNOLOGY
At 11:48 AM 2/25/2004, you wrote:
No matter what I put in parking.conf for parkpos, I find that the first
call is always parked on 701. Is this a bug?
With recent CVS builds I've been able to specify 7000 and 7001-7200 as the
call parking lot. I haven't tried any other numbers.
--Ernest
At 12:34 PM 2/25/2004, you wrote:
On Wed, 25 Feb 2004, Ernest W. Lessenger wrote:
With recent CVS builds I've been able to specify 7000 and 7001-7200 as the
call parking lot. I haven't tried any other numbers.
The parking lot is assigned by the user or by the system?
I found that my
I'm writing an application for asterisk (really just a set of access
commands to the builtin API), and I notice that a lot of existing
applications are not thread-safe. Should they be? Should mine be?
Thanks,
--Ernest
___
Asterisk-Users mailing list
At 08:31 AM 2/23/2004, you wrote:
On Monday 23 February 2004 10:15, Ernest W. Lessenger wrote:
I'm writing an application for asterisk (really just a set of
access commands to the builtin API), and I notice that a lot of
existing applications are not thread-safe. Should they be? Should
mine
At 09:14 AM 2/23/2004, you wrote:
Why would you program something that isn't thread safe? From what I can
tell, it isn't much extra effort to do things the right way instead of
debuging crap later.
I wouldn't, and generally don't. But sometimes (rarely) you need to include
functions that aren't
At 05:15 PM 2/19/2004, you wrote:
I usually use
[EMAIL PROTECTED]
they do eventually get back to you.
We operate a call centre and have offered them an inbound package, but
it seems they are not interested.
Matt
P.S. Our DID line hasn't been working for around a month nowin the
process of
At one point I had Asterisk running on a Fedora Core 1 based embedded
system using a Soekris embedded device. Once the OS is running, the only
hard part is finding a source of timing for the MOH and conference calling.
However, I think the new Soekris units have a timing source on them (USB).
At 06:53 PM 12/28/2003, you wrote:
Side note, and probably not related, but what's the SB live card for? You
don't actually use this computer, do you? It's a server, let it be one...
Asterisk requires a timing source to play music on hold and conference VoIP
channels. The SB performs this
At 11:10 AM 12/24/2003, you wrote:
Skinny phone functionality is 'richer' than SIP phone functionality. First
off, on a skinny phone, in hands free mode, you can start dialling and the
phone will automatically go off hook. Sip requires you to manually hit the
speaker button, hit new call, or
At 09:20 AM 12/17/2003, you wrote:
Hi,
I am trying ti install an asterisk system on fedora core 1. During the
make of asterisk I got the folowing problem:
bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
make: *** [ast_expr.c] Broken pipe
Does anybody know how to solve this?
At 08:45 PM 12/9/2003, you wrote:
Ok - Here is where I am at. I know this topic has been discussed
before, but never a solid answer was set in place. Is anyone aware of
any phones that can put a caller on hold and the caller hear MOH by the
user pressing the hold button. I understand most
At 12:23 PM 12/8/2003, listas iPfone
[EMAIL PROTECTED] wrote:
I updated
my snom200 to 2.02t and now MOH from * don´t works anymore... only the
MOH from snom server and if i clear the MOH server field in the phone i
have no MOH at all..( with the transfer button, moh plays using a
extension).
At 01:32 PM 12/8/2003, you wrote:
Greetings,
I have been experimenting with Asterisk for a few weeks and finally
decided to
take the plunge and purchase a few DIDs for inbound calling. Our attempts at
IAX/IAX2 connectivity with VoicePulse have been less than successful. We get
Registration
At 07:17 AM 12/5/2003, you wrote:
I guess for the XBox you would need some external gateway. Audicodes or
Mediatrix come to mind but they start at $500.
A year ago, I installed Linux on Playstation 2. I had to purchase it with
the hardware for about $200. (40GB, keyboard and some network
At 02:34 PM 12/4/2003, you wrote:
However, considering the traffic volumes that you are talking about, is it
really true to say that the traditional telco cards are astronomically
priced, given the amount of revenue that can be generated per month on a
DS3?
Eight quad-span T-1 cards from Digium:
Correcting an idiot-math error (24/4 != 8 and 1000*3 != 1000) ...
At 02:34 PM 12/4/2003, you wrote:
However, considering the traffic volumes that you are talking about, is it
really true to say that the traditional telco cards are astronomically
priced, given the amount of revenue that can be
At 10:37 AM 12/2/2003, you wrote:
Does
asterisk support G.729a or do you have to add something (is there an open
source one)
Yes, Yes, and Maybe (i.e. it's not free, but you can license one through
Digium, and there is a reference source available but absolutely NOT
open-source).
Check out this
At 10:59 AM 11/22/2003, you wrote:
Is there a solution to have the hold button to play MOH. Or even some
type of ADSI function that allows for this?
What kind of phone do you have? MOH depends first on the phone, as it is
the phone that decides what to do when you press the hold button.
--Ernest
Sounds like a great idea! I'll gladly help if requested (I'm a technical
writer).
Comment: I don't see anything on echo cancellation. That's a big enough and
common enough issue that it deserves some discussion.
--Ernest
At 10:46 AM 11/21/2003, you wrote:
Hi Steven,
I think this is a great
At 07:26 AM 11/20/2003, you wrote:
Probably too late to ask for, but for us reversal polarity detection
(far end answer supervision) is very important for billing and pre-paid
purpose.
Don't the X100P cards already support this? I believe it's called KewlStart.
--Ernest
At 02:08 PM 11/13/2003, you wrote:
Now... for the self empowered type... You can go to http://lists.digium.com
and remove yourself... but I still would like to see what is meant by or
ELSE.
Presumably he means or else I'll have to actually look at the instructions
printed at the bottom of every
At 11:07 AM 11/10/2003, you wrote:
Thanks everyone for your help on this..
For those who are interested I have done some speed tests on these two
queries (below) on my server and the results are..
Test script of 1000 quieries..
Query1 (code field not indexed) = 47.183s
Query1 (code field indexed)
At 10:28 AM 11/11/2003, you wrote:
I'm sorry this is somewhat offtopic, but I do plan to use this to help
me create documentation for the * project.. so I guess it is somewhat on
topic :)
Anyways, I am looking for some sort of document control system. It
should act somewhat like a CVS where it
At 06:51 AM 11/11/2003, you wrote:
Ummm.. kind of. I mean, it says Enter the first 3 digits of the
persons last night and you enter them via the keypad, it then searches
for the names, and says, Calling so-and-so. I think I've seen this
feature on a phone system I called once, but I can't
At 10:33 AM 11/10/2003, you wrote:
Steve Underwood [EMAIL PROTECTED] wrote in news:3FAE487A.7000508
@coppice.org:
Hi,
I've kind of ported a DTMF text extry method I wrote some time ago for
Dialogic. It is now a semi-working Asterisk app. I've still got to clean
up some stuff in how Festival
At 01:08 PM 11/8/2003, you wrote:
So what do people think about adding the call rate to the CDR
structure??
Sounds great, but there's one problem. How does asterisk know what the
current rate in effect is? I can think of several ways to do this, but they
all involve some fairly significant C
At 07:01 AM 11/7/2003, you wrote:
Hi !
Now I can hear nice mp3 through my phone... Great :P
And many thanks for your posts. Now it's working fine... hmmm almost !!!
In fact, I m using DialenMP3.agi. It's a real nice agi script...
For those which would not know, you will find it here :
At 09:20 AM 11/7/2003, you wrote:
I though to it also, but really I don't know how can I get the pid of a
process ran by asterisk.
I mean, the only think I do it's :print EXEC MP3Player \$key\\n;
Then asterisk take the hand with mp3player applications that will launch
mpg123, etc...
You're right,
At 03:12 PM 11/7/2003, you wrote:
Is there a way to put a call on hold and play music on hold with out
using the park app?
There is a MusicOnHold extension that is like park, except that you can
never take them off hold.
Most SIP phones also have the ability to put a call on hold and tell * to
At 04:00 PM 11/5/2003, you wrote:
We are Mediatrix's US distributor and have used them with Asterisk in our
lab and have had several resellers purchase them to use with Asterisk. They
seem to work well with Asterisk, but I have to agree that the configuration
leaves a lot to be desired. Their
Hey all,
For those of you who are really worried about asterisk performance, I
thought I might alert you to a toy you might play around with. The Intel
Performance Primitives contain a number of optimized functions for use in
digital signal processing that could help with echo cancellation,
At 10:23 PM 10/31/2003, Bryan Nolen wrote:
System execute asterisk -rx reload
?
Yes, correct.
--Ernest
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Sent: Saturday, 1 November 2003 5:18 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
At 07:42 AM 11/1/2003, you wrote:
Hi All,
Is it possible to show which line a call has come in on in *.
Yes, absolutely. In asterisk each line is a channel. The channel
information is VITAL to the call and is available (and used) everywhere in
asterisk. Channels look like this: ZAP/1-1, which
At 08:54 AM 11/1/2003, you wrote:
P.S.: Looks like I have to post this once a day now.
You should post this (or I'll do it for you, with permission, as I already
have an account) on the Asterisk wiki at www.voip-info.org. You might still
have to post, but at least it will be out there...
a conversation with a compulsive interjector who
never finishes his sentences :) Do you have this problem? If so, do you
recall how you solved it?
Thanks,
--Ernest
in the makefile of zaptel and recompiling.
miklos
- Original Message -
From: Ernest W. Lessenger [EMAIL PROTECTED
At 05:15 PM 11/1/2003, you wrote:
Apologies if there is a cleanly written and searchable FAQ that I could be
directed to. I have no problem to RTFM if I can find the FM...
Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R
servers that do not support X windows under RH8.x and
At 05:03 PM 11/1/2003, you wrote:
Netfinity 4000R
servers that do not support X windows under RH8.x and I
prefer not to go
back to RH7.3...
I recall in the archives somewhere, and through someone's post earlier
today, that there is some sort of problem with RH9 with Zaptel (hardware)
drivers
I'm testing out my SNOM 200 phone by trying to call out through NuFone.
When I do so, I don't hear an echo at all (in fact I can't hear myself
through the phone) but the callee can hear an echo when she speaks. NuFone
tells me their network is totally digital and so can't be involved in an
Is anyone using Asterisk as the gatekeeper/proxy for videophone calls?
Thanks,
--Ernest
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Does anyone know of a command-line tool that I can use to mix my own MOH
tracks? Specifically, I want to be able to do this:
1) Record a Your call is valuable to us... advertisement
2) Specify a number of song files to be played randomly/in sequence/whatever
3) Insert or overlay the
Has anyone successfully used a Luxon VoIP gateway with *?
Thanks,
--Ernest
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Check out the software at
http://www.xten.com/.
Their XTen-Web and XTen.NET products may help you out. Allowing people to
dial a landline is actually quite simple, and can definately be done with
Asterisk.
--Ernest
At 08:00 AM 10/22/2003, you wrote:
Resending
this. Any help appreciated.
I'm using the SNOM 200 latest SIP beta (so that I can have the GSM codec,
etc). Everything seems to be working fine, but the music on hold doesn't
play when I use the HOLD button on the snom. Any suggestions?
Thanks,
--Ernest
___
Asterisk-Users
At 09:53 AM 10/21/2003, you wrote:
I know this is going to sound like a strange question, but here goes:
Does anyone know of a SIP softphone that has either a button or a
programmable soft-key to send the current call to VoiceMail?
Here is what I use with a SNOM 200...
exten =
Is anyone on the list using an Audiocodes gateway with asterisk and SIP?
I'm looking at that platform, but I have a couple of issues:
1) Echo cancellation. The echo that I'm hearing with an X100P is
unacceptable. Does the Audiocodes do better?
2) Line signalling. I'm using Kewlstart with the
At 12:33 PM 9/29/2003, you wrote:
Can you clarify any / find me on IRC? (irc.freenode.net/#asterisk/kram)
Just FYI: I had similar problems for a while, and then I completely
scrapped my CVS directory and did a CVS CHECKOUT (instead of an update).
That solved the problem.
--Ernest
Mark
On
Does anybody have a reasonable solution for an Outlook MAPI plugin that
works with asterisk? At very least, I would like Asterisk to push incoming
call information to the computer, which should then open an Outlook form,
launch a web browser, etc. Beyond that, it would be cool to have Outlook
can also make it more complicated with time-based
includes and gotos.
--Ernest
At 09:57 AM 9/16/2003 -0700, Ernest W. Lessenger wrote:
At 11:22 PM 9/14/2003, you wrote:
First -- Thanks to everyone who offered their help and tips on getting my
Cisco 7960 working with Asterisk -- this is great
At 07:52 PM 9/14/2003, you wrote:
Any chance of getting this feature added (preferrable as another option
on each mailbox setting in voicemail.conf (after the pager # maybe))? I
know it could be hacked, but I am trying to avoid those type of
improvements. :)
Asterisk already has an outgoing call
Can anyone tell me the features of the Cisco 7905 with SIP? I mean things
like number of lines, speakerphone, transfer buttons, etc. I've seen the
Cisco material, but all it told me was how nifty it is and how wonderful
the XML interface will be ;)
Thanks,
--Ernest
At 01:01 PM 9/15/2003, you wrote:
Hi all.
I have some questions:
1) Is there a way to get a full log of the calls (incoming and
outgoing)
You can get the Call Detail Records which show you the incoming user and
the initial dialed extension as well as the date, time, etc. They don't
At 11:24 AM 9/12/2003, you wrote:
Anyone configured X-Lite with asterisk? Any help on config, both on X-Lite
and asterisk?
Yep. Here's an acceptable SIP.CONF entry...
[551212]
type=friend
username=551212
secret=12345678
host=dynamic
qualify=1000
nat=1
mailbox=551212
context=default
We are trying to implement area-code dialing in our asterisk PBX.
Basically: we will have a number of customers, who may be in different area
codes, that want to direct-dial each other's extensions. We want this to
work like a real centrex, in that seven-digit numbers should try (1)
local VoIP
My S100U also gets quite warm. I haven't had any trouble with it though.
--Ernest
At 01:31 PM 9/10/2003, you wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jason A. Pattie wrote:
| I have another observation for you. Do your S100U's get warm? I've
| left this one plugged into the USB
At 11:26 AM 9/9/2003 -0600, you wrote:
What is the general consciences for the allow=all statement? Should it be
used, should it be specific towards those codecs supported, or removed?
My understanding is that you MUST have at least one allow and one deny, or
none at all. Just having one or the
At 02:38 PM 9/9/2003 -0500, you wrote:
That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint. Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
endpoints use the same protocol (SIP) AND the
At 07:57 PM 9/9/2003 -0400, you wrote:
Travis Johnson wrote:
I've called NuFone and was not impressed by their voicemail answering
system (choppy) and was unable to even leave a message before the phone
call was disconnected (in the middle of the
recording).
So your going to judge our system
At 07:52 PM 9/8/2003 +0200, you wrote:
Is there a way to configure Hylafax or sth one modem behind an ATA-186
to email faxes to different adresses depending on the called number ?
I've looked into this myself, and I think the answer is yes, with some
minor code changes. My thought is that you
At 03:17 PM 9/8/2003 -0400, you wrote:
We would like to setup in house SIP phones with numbered extensions for
demonstration purposes.
What is the syntax to associate a extension with SIP phone?
exten = 1234,1,Dial(SIP/username)
Does the Dial application have a SIP specific entry for example:
Is it possible to limit the number of lines provided by a given SIP/IAX
connection? For example: I want to limit SIP extensions to only a single
incoming line, even the phone itself can handle three. Or, I might want to
prevent extensions from making more than one outgoing call at a time. Or, I
In reading the source for the CDR_CSV module, I understand that it should
use the SIP username as the account code for calls made from SIP devices.
However, nothing is being recorded in the csv file for that field (i.e.
blank value). Is there any way to add an account code for SIP users? I can
At 12:21 PM 9/5/2003 -0700, you wrote:
In reading the source for the CDR_CSV module, I understand that it should
use the SIP username as the account code for calls made from SIP devices.
However, nothing is being recorded in the csv file for that field (i.e.
blank value). Is there any way to add
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