Hey Gurus,
I have a very simple asterisk setup that basically lets me share a PSTN
line from one location to another. I would like to have the phones at
both locations ring when the PSTN # is dialed(inbound calls from PSTN
to asterisk).
I tried something like:
exten =
On Jan 28, 2006, at 12:54 AM, Ronald Wiplinger wrote:
Martin Joseph wrote:
snipI tried something like:
exten = 2020,2,Dial(SIP/2005,25,trIAX/2010,25,tr)
I thought this might cause both 2005 and 2010 to ring when 2020 was
dialed, but only 2005 rings?
Below works for me:
PHONE_LOCAL
On Jan 28, 2006, at 6:50 AM, Mark Adams wrote:
x-tad-smallerHi Everyone,/x-tad-smallerx-tad-smallerI know this may be off subject but I am not sure who to ask. I am currently looking for voip termination that is closest to replicating U.S. pots service. I run I.V.R. systems and I want to point
On Jan 28, 2006, at 7:15 PM, Vic wrote:
Hi, Zoa,
yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them.
We will also need an IVR function as well.
I am not up to speed on Asterisk yet, so, I am a
On Jan 28, 2006, at 2:13 PM, Rene Kluwen wrote:
Is somebody here using a RoadRunner/Time Warner connection and able to
successfully with SIP (or IAX2)?
We are experiencing high latency up to the point that the voice
conversation
is not understandable anymore. This goes for both SIP and
On Jan 29, 2006, at 11:24 AM, Rich Adamson wrote:
Anyone tried to muck around with using the 488 for both fxs and fxo
with asterisk?
I've been playing with one for the last couple of days, and it looks
like its a little lower quality then the spa3k. No gain settings,
echo
canceller is
On Jan 29, 2006, at 10:30 AM, Warren Burstein wrote:
I took a look at the asterisk-1.2.3 Makefile, seems to me that the
WARNING is just a list of all the .so files found in the modules
directory that aren't also found in a subdirectory, it isn't checking
that they were built with the current
On Jan 29, 2006, at 1:24 PM, Michiel van Baak wrote:
On 13:09, Sun 29 Jan 06, Martin Joseph wrote:
I removed the following to get it starting up again:
app_enumlookup.so
app_groupcount.so
app_md5.so
app_txtcidname.so
func_cut.so
Both the README and the UPGRADE listed that those functions
I have a strange problem with echo.
My setup includes a Grandstream HT-488 which is both an FXO and a FXS.
I noticed last evening that if I called the FXS through my asterisk box
from my cell, the resulting connection was fine for me at the cell end,
but produced dramatic and conversation
On Jan 31, 2006, at 1:19 PM, Brent Torrenga wrote:
Has anyone had problems getting their preffered codecs on the Teliax
web
interface taking effect?
I have two accounts, two separate yet similarly configured * servers.
On one
account the settings took right away - on another server I am
On Jan 31, 2006, at 12:08 PM, Fran Sedano wrote:
Hi!
No one can help me with this??
x-tad-smaller- Original Message -/x-tad-smallerx-tad-smallerFrom:/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerFran Sedano/x-tad-smallerx-tad-smaller
Why?
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On Feb 1, 2006, at 4:24 AM, Olle E Johansson wrote:
Cosmin Prund wrote:
As the subject line says: Is PING a good indicator of network
latency? If
not, how can I measure latency?
Using Asterisk is a good way. If you define a phone in sip.conf and
turn on qualify=, we will measure the latency
On Feb 6, 2006, at 5:08 AM, ammar Ali wrote:
Jose,
There are No open source IP phones, I was only joking, I assumed you
should know what an open source is.
The AG-168V is an open sourced ATA. Although the idea that Walmart
would give something (useful) away for free, was funny to me.
Any feedback on this brand and in particular on doing business with
WelltechUSA?
I am looking to the Wellgate 3701A which is a 1FXS-1FXO arrangement. I
am hoping to replace the near worthless Grandstream HT-488.
This company is telling me that I need to wire $ directly into there
bank
On Feb 8, 2006, at 9:22 AM, Ariel Batista wrote:
I normally don't like talking bad about products. But I would like to
say that the Welltech/Wellgate are not products that are support to
work with asterisk. I have invested many hours of work in getting
there device to work with Asterisk.
I am wondering if the instructions for hard wiring a Tellabs canceler
are applicable to a regular old two wire loop?
Or is this only something that works for people with T1?
Any comments from people that have tried this are appreciated.
___
On Feb 12, 2006, at 1:05 AM, Nitin Gupta wrote:
Sorry for re-posting this message -
I am trying to run the latest stable Asterix version 1.2.4. on 64 bit
amd procesor.
Things are working but the playback sounds that I hear when tring to
connect over IAX are of very high frequency.
i.e a
On Feb 13, 2006, at 10:20 AM, Eric ManxPower Wieling wrote:
snip
The nearest CO my POTS line goes to is 11 miles away.
snip
i take it you aren't a DSL customer?
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On Feb 13, 2006, at 2:45 AM, Simone Cittadini wrote:
C F ha scritto:
Am I the only one having trouble with this list?
Since the begining of the week I have not been receiving mail from the
list like I used to, is this a gmail problem? or is it subscription
problem? or is something wrong with
On Feb 16, 2006, at 1:11 PM, Adolfo R. Brandes wrote:
turby wrote:
is there recomended source files for t.38 pass? latest cvs does not
work for me.
is it possible publish working src?
You mean T.38 passthrough? I've just uploaded an asterisk-1.2.4
backport of the lastest svn
On Feb 16, 2006, at 3:40 PM, [EMAIL PROTECTED] wrote:
http://www.zoomtel.com/products/voip_products.html
anyone using these? they look very interesting in that they support
ilbc, and they offer a separate cheaper model without g729 license.
i'm wondering if their EC is better than the
On Feb 17, 2006, at 6:36 AM, Robert Webb wrote:
Sorry, this is off topic to asterisk itself, but is about the list
server.
I had a power failure lastnight at home, where my email server
resides, and my network was down for about 20 minutes, that was after
45 minutes of uptime on UPS.
On Feb 19, 2006, at 6:06 AM, Phil Blundell wrote:
Overall, I'm happier with the SPAs than the handytones, though neither
of them are entirely perfect. Oh well.
Thanks for the update...
I am being told by the freaks at Grandstream that there will be a
firmware update forthcoming to try to
On Feb 19, 2006, at 9:41 AM, Dovid Bender wrote:
Some people have to stap on others to make them selves
feel good. Very unfortunate.
Some people have no sense of humor. Very unfortunate.
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This is also very dependent on where you are and who your ISP is...
I used Teliax and there setup instructions and support are excellent.
Unfortunately for me, my ISP (frickin comcast) has a very poor route
to Teliax's servers. This seems to be somewhat changeable, but is
consistently
On Feb 22, 2006, at 10:24 AM, Rusty Dekema wrote:
On 2/22/06, Matt [EMAIL PROTECTED] wrote:
Yes.. there are provisioning tools that you have to get.
Unfortunately it's this catch 22 loop. You have to prove that you can
offer 200+ ATAs to customers, or you can't get the tools, but yet, you
On Feb 22, 2006, at 3:44 AM, Jean-Marc Salsa wrote:
Thanks,
But, I do not have phones connected to Asterisk ...
but only one peer : my softswitch ...
So call flow is Phone - Softswitch - Asterisk - Voicemail
I can force the link Sofswitch - Asterisk ( Codec and DMTF Mode )
Codec is PCMx
On Feb 23, 2006, at 4:58 AM, Adam Robins wrote:
Thanks,
We already have a cron reboot of all of our Asterisk servers every
night. We've been doing this for over a year due to memory leak
issues.
??? What do you think this is windows 95??? I had a problem like that I
would be looking at
On Feb 23, 2006, at 6:11 AM, Tele Cost Price Reducer wrote:
hi ,
i have some options we are working with at vast deployement with no
problems:
www.tigernetcom.com - type 102 is a nice ATA, like GS 486 but far away
better.
we import directly from the producer at great prices so if anybody
Hi again,
Kind of sheepish about asking for help, as I have only spent a day
banging my head off this...
I got my new Welltech 3701a, 1FXS,1FXO gateway.
I flashed it with what is seemingly the appropriate firmware (SIP
V1.04). This seems to have gone ok, and it is now registering both
On Feb 25, 2006, at 10:04 PM, Anton Krall wrote:
Nice!
Did you ever think about trimming your messages?
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On Feb 26, 2006, at 12:57 AM, Alexander Burke wrote:
Hello, list!
After Googling and checking out the voip-info wiki, I haven't had much
luck in locating a decent web-based voicemail system for Asterisk to
check your VM while you're away from the office without using a phone.
Can anyone
On Feb 26, 2006, at 5:37 AM, Wooi Koay wrote:
I have a POTS and a sip incoming into my asterisk server. When I call
the POTS number from outside (cell or landline) and trying to
authenticate myself when enter #, 8 out of 10 times I got an
authentication incorrect. If I call in to the sip
On Feb 26, 2006, at 5:43 PM, hugolivude wrote:
Say, thanks to all you for your time in responding.
I hope I don't sound unappreciative (I have no time for flamers) but
I don't understand how changing from SIP to IAX would make any
difference. I don't have any problems with the signalling
On Feb 26, 2006, at 11:59 AM, Paul wrote:
snip
Unless you have good QOS routing be sure that mail server is somewhere
where you don't have voip phones. I had a mail server at an office with
400k sdsl. I would be on a call and let an incoming call go to voice
mail. The incoming email with wav
Short version:
Flash device with latest SIP firmware (currently 1.04)
Set Network (I am using the LAN port only) and SIP config as
expected.
Set Line configuration so that the FXO is hotline to the asterisk
extension you want to ring with incoming PSTN calls (mine is set to
2020).
Set System
On Feb 27, 2006, at 6:09 AM, Dr. Michael J. Chudobiak wrote:
I find that DTMF does not work reliably if jitterbuffer=on for certain
IAX providers. For instance, DTMF tones are missed entirely about half
the time when I dial into an exgn.net account. However, it always
works fine for an
On Feb 26, 2006, at 8:03 PM, Joseph Blake wrote:
We're setting up asterisk at the office (really doing some testing
right now) and it is going to be hosted on a dual G5 XServe running OS
X.
I love it. Glad to hear it. Should be a monster.
We're an apple certified solutions provider, etc.
On Feb 27, 2006, at 2:55 AM, Dr. Michael J. Chudobiak wrote:
Can someone recommend an IAX provider for US DIDs who will:
snip
3) Have great audio quality
This is somewhat a meaningless question, as the route from you to the
call terminating service can make or break the quality.
You
On Feb 28, 2006, at 1:22 AM, [EMAIL PROTECTED] wrote:
I am having problems with a Zoom 5801 and *.
It does not appear possible to route voip calls out the FXO, all voip
calls get routed to the FXS no matter what.snip
If there is a routing function of some kind on the modem setup,
perhaps
On Feb 28, 2006, at 6:50 AM, Ash Thakrar wrote:
Hi Mark,
Thanks for your reply.
For the phase you have indicated the time it took was immediate, no
delays
there.
I have seen on the list several discussions of how additional delay on
ringing can be due to Asterisk trying to get caller ID
On Feb 28, 2006, at 7:14 PM, Anton Krall wrote:
Anyway the phone can compensate? I don't think it works that way but
worth
asking..
If the phone has an input gain (for phone users voice) then adjusting
it down can help echo that is being generated at the far end. ie if
it's too loud
On Feb 28, 2006, at 6:43 PM, James Harper wrote:
What about the fonebridge (http://www.red-fone.com/fonebridge.html)?
It uses POE, so you could hack something together to supply 48V @ 15W
if
you don't have access to a power point, and it appears to have a 2 port
switch built in so you could
On Feb 28, 2006, at 2:35 PM, [EMAIL PROTECTED] wrote:
On Tue, 28 Feb 2006, Cory Andrews wrote:
Here is a link to some additional resources which may be helpful in
configuring the 5801 and other Zoom products
http://www.zoom.com/salessupport/index.php?pd=ATA_Documentation/
I just found out,
On Feb 28, 2006, at 9:51 PM, Andres wrote:
Ed Greenberg wrote:
I need to set up an office full of Cisco 7960 phones behind NAT with
the server out in Colo.
The first test phone registers fine, but the second one does not
register.
The first phone's registration looks like so:
On Mar 1, 2006, at 1:46 AM, [EMAIL PROTECTED] wrote:
On Tue, 28 Feb 2006, Martin Joseph wrote:
On Feb 28, 2006, at 2:35 PM, [EMAIL PROTECTED] wrote:
On Tue, 28 Feb 2006, Cory Andrews wrote:
Here is a link to some additional resources which may be helpful in
configuring the 5801 and other
On Mar 1, 2006, at 5:20 PM, kevin ling wrote:
Hi,
I have another model 3702a (2FXS 2FXO) voice gateway. You can
implement
one-stage dialing on this device.
1. using sip show peers to make sure two ports (1fxo/1fxs) was
registered
to asterisk.
2. login 3701 web and change the defualt
On Mar 1, 2006, at 3:03 PM, [EMAIL PROTECTED] wrote:
On Wed, 1 Mar 2006, Arsen Chaloyan wrote:
The inbound PSTN DTMF works excellently, e.g. people
calling from PSTN
into the * box are able to pick IVR items with DTMF
reliably.
exactly!
There are some other problems with DTMF and spa3000.
On Mar 2, 2006, at 3:46 PM, Wai Wu wrote:
You can really mix G729 encoded frames. So I would guess that licenses
are not needed for non-G279 devices. BTW, there is a difference
conference app (forgot the name) that only mixes the two parties that
have the loudest volumn. It sounds more
On Mar 2, 2006, at 12:32 PM, Anton Krall wrote:
Looks very nice.. Is it GPL, GNU?
Maybe if you trimmed you posts and pasted relevant quotes, we could
have some idea what this question means...
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On Mar 2, 2006, at 9:46 AM, Matt wrote:
Hi,
Occassionally Asterisk will go down and I have to restart it.. not
often.. but sometimes. When it does the manager interface stops
working, as does the CLI.
My thoughts was to poll the manager interface once every 5 minutes for
a value. If I don't
On Mar 3, 2006, at 9:48 AM, Anton Krall wrote:
Guys.
I have a te100p with unicall and an E1 and Im having problem with DTMF
tones
but the weird thing is, I only have problems sending the tones to
certain
phone numbers, anybody seen this behavior?
Asterisk shows on the console the dtmf
On Mar 3, 2006, at 11:42 AM, Michael Sampson wrote:
It is my understanding that when you hear echo the problem is on the other end. So if a caller complains they hear echo that is something you should be dealing with, but if you hear echo that is the phone companies fault. Now with a normal
On Mar 3, 2006, at 9:49 AM, David Thomas wrote:
I'm doing an install for a client with the following requirements.
- 1 Million minutes of outbound calling
Per what?
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On Mar 3, 2006, at 9:57 AM, Bartosz Jozwiak wrote:
Hi guys,
I am trying to register IP IAX2 phone to our Asterisk server.
this is what I see on traffic debug between the asterisk server and IP phone.
I do not see anything in asterisk console.
Can somebody give me hints what could be the
On Mar 3, 2006, at 11:35 AM, Tom Vile wrote:
I have had the same issue with Sipura as well, they gave me quite the
run around on 1 bad 2002 ATA and decided to just forget about it and
bought a new one to save time.
So they gave you shitty support and you bought more?
What are you a microsoft
http://grandstream.com/BETATEST/HT488_496_386/
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On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Mar 3, 2006, at 11:35 AM, Tom Vile wrote:
I have had the same issue with Sipura as well, they gave me quite the
run around on 1 bad 2002 ATA and decided to just forget about it and
bought a new one to save time.
So they gave you shitty
I realized today that my call waiting isn't working properly. If I am
using the FXS attached phone and a call comes in the FXO, it just goes
directly to voicemail, with no indication (call waiting beep).
If I flash there is a second dial tone, and I can initiate a second
call.
If I am
On Mar 4, 2006, at 7:54 AM, The Asterisk Development Team wrote:
Asterisk 1.2.5 is now available for download on the ftp. See the
ChangeLog for details about what has changed.
Reading the changelog I notice the following... I suppose it should
say incorrect?
2006-02-17 01:55 +
Probably just me being dumb, but I am trying to update my asterisk to
the latest (1.2.5)
When I go to my /usr/src/asterisk I type:
make upgrade
make install
This seems to be doing it's thing, but when I type show version from
the console (after a restart) I still get:
Asterisk
On Mar 6, 2006, at 8:39 AM, Colin Anderson wrote:
http://www.voip-info.org/tiki-index.php?
page=Asterisk%20MacOSX%20Support
It works but it's bitchy as hell to run because of root issues in
OSX.
I wonder what the above root issues means?
I run it on my Mini. Zaptel is not supported. You
On Mar 6, 2006, at 6:46 AM, [EMAIL PROTECTED] wrote:
Hi friend,
I am running asterisk in production and it is being used by many
people using h323. I cannot afford to change all their configurations.
Also, the newer asterisk dosenot support inband for h323 properly.
Thats why I want two
On Mar 6, 2006, at 6:46 AM, Giorgio Incantalupo wrote:
Hi,
I have an asterisk 1.2.1 (on a debian sarge) box with a TDM400P card.
I connected the TDM400P to a grandstream 286 to use a VoIP provider.
It seems all right except for a little problem: one call every 30 is
made to a wrong number.
On Mar 6, 2006, at 7:14 PM, Tom Vile wrote:
The firmware made thing much worse for me. Call waiting does not
work, I cant disable the local feature codes and DTMF transmission is
horrible as well. It has not locked up on me yet though because I
through it in the trash and plugged in my
On Mar 6, 2006, at 10:32 PM, bmw suzuki wrote:
Hello all ... mY first ever post in here.
I am bit or (full) confused on what this program does.is it useful if
i have a alcatel pabx system.And i can bill my guests for their call
charges etc..
You will need to decide that.
can i use it on
On Mar 7, 2006, at 2:38 AM, Giridhar Bandi wrote:
ya i found it it *1 to start recording from the caller end
Also pushing *1 again stops recording.
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On Mar 7, 2006, at 9:51 AM, Giridhar Bandi wrote:
Hey thanks for the prompt response ( that's what i liked about this
list )
i was not able to start recording
i have pap2 box as clients and the dial plan of pap2 is as bellow
(*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.)
On Mar 7, 2006, at 7:02 AM, artifex maximus wrote:
Hi all!
I have the following setup:
Phone lines - traditional PBX - Welltech 3802
- VPN -
Asterisk - Linksys PAP2/Welltech ATA-151 - phone
There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2)
PBX extensions. Asterisk is a
On Mar 9, 2006, at 9:30 PM, Marc Archer wrote:
x-tad-smallerHi All,/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerThis is probably a stupid question, but I’m trying to figure out if I Asterisk is in the middle of the media stream or not… Is there a command or something that indicates
Just a warning for you all that are using Nokia series E phones for SIP
function.
I updated my phones firmware today using the Nokia Updater, and now
the SIP functionality, which previously worked pretty well is
completely broken.
The phone no longer registers with asterisk, although it
On 2007-04-17 00:53:56 -0700, Dinesh Nair [EMAIL PROTECTED] said:
On Mon, 16 Apr 2007 20:14:40 -0700, Martin Joseph wrote:
The phone no longer registers with asterisk, although it displays the
little icon as though it has, and it doesn't even seem to try to pass
calls to asterisk...
So, I
On 2007-03-26 01:46:40 -0700, Salvatore Giudice
[EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
I opened up a ticket with them, but I'm not holding my breath. I think it's
time to start moving my DID's before the inbound stops working.
That seems like it was probably
Hello again gurus.
I have been using Asterisk with great results going on a couple of
years now.
My primary box is running asterisk 1.42 built from a tar ball on Mac
OSX 10.4.9.
I have a very odd issue that I cannot seem to nail down, which is
related to my Nokia E60 SIP phone.
I use
On May 14, 2007, at 12:34 PM, Tim Panton wrote:
On 14 May 2007, at 17:50, Martin Joseph wrote:
Hello again gurus.
I have been using Asterisk with great results going on a couple of
years now.
My primary box is running asterisk 1.42 built from a tar ball on
Mac OSX 10.4.9.
I have
On 2007-01-17 10:29:43 -0800, Yelson Vivas [EMAIL PROTECTED] said:
Hi Guys
I'm conecting 2 astersk servers using this arquitecture
(Ext softphone)==sip==(asterisk 1)iax2 trunk(asterisk 2)
===alaw==(pstn)
If i call from the Ext to the asterisk 2 the sound is perfect, but if
i call
On 2007-01-28 08:37:43 -0800, Eric Germann [EMAIL PROTECTED] said:
We LOVE Teliax. We're on a Time Warner business class fiber connection and
avg 25ms latency from Ohio to Denver CO.
With that connection I would love Teliax also.
Marty
___
On 2007-02-14 22:12:23 -0800, jameson asterisk [EMAIL PROTECTED] said:
I'm currently looking to deploy an Asterisk server using an FXO media
gateway to connect to the PSTN and was looking for any user experiences that
may aid in selecting a gateway. Specifically i'm looking for a 4-port model
On 2007-02-22 04:22:20 -0800, Frederico Madeira [EMAIL PROTECTED] said:
Hi guys,
My asterisk is show me some errors on line registration.
This message appear on console: Request to schedule in the past?!?!
What it mean ?
Thanks.
I see this message all the time on my lowely powerPC mac
On 2007-03-24 01:53:16 -0700, Edoardo Serra
[EMAIL PROTECTED] said:
Hi Francois,
[EMAIL PROTECTED] ha scritto:
Hi men,
I have already encountered some issue like this with few switches (very
known great brand) which doesn't like VoIP traffic !
I also have switches of a very known
On 2007-03-23 14:37:18 -0700, Tom Lynn [EMAIL PROTECTED] said:
Now I know where they've been spending my remaining balance...
I still use Sellvoip as my primary terminator, and have found the call
quality to be superior to any other ITSP from my location (Seattle).
I agree completely
On Jul 5, 2006, at 7:58 AM, K Y Iyer wrote:
Hi
Hi!
Am a bit confused about the basic requirements for a simple, small,
test
Asterisk setup.
There are many options...
I want to setup a PBX with 8 PSTN lines and 50 extensions. For
argument's sake we'll assume all 50 extensions and 8
On 2006-08-24 18:10:20 -0700, kritikus Araklidas
[EMAIL PROTECTED] said:
So:
The FXO car is for the Pots lines (I.E. bellsouth line) so if you need
a analog phone cennected to asterisk you need a FXS card, so if you
gonna use a SIP Soft Phone (or a regular SIP Phone) you only need a
On 2006-08-23 18:02:52 -0700, El Flynn [EMAIL PROTECTED] said:
Hi list,
Just wondering -- has anyone used the SIP phone feature on the Nokia
E60/61/70 phones? We're trying to see if this would be an OK phone to
get for the company, particularly since we're already running Asterisk.
Not
On 2006-08-24 06:32:27 -0700, Jon Schøpzinsky [EMAIL PROTECTED] said:
I also have this phone, and have stumbled in to the same problem.
I just think that it isn't in nokia's interest to change this, as it
forces consumers to have some sort of local hardware, that (possibly)
only the telecom
Along the same lines as this question... Are there any Voip phones that
have dual gigabit ethernet ports?
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On 2006-08-25 10:35:36 -0700, Sam Tam [EMAIL PROTECTED] said:
Hello
WE can provide you with budget GSM Gateway if you are interested?
Sam
Hey Scumbag,
How many timed do you need to be told that this isn't the place to sell
your wares?
Please Stop it!
On 2006-08-22 01:59:09 -0700, Tomislav Parčina [EMAIL PROTECTED] said:
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5
over Grandstream HT488 ATA.
snip
Personally I found the FXO port on the HT-488 to unworkable except as a
backup for power outages.
I
On 2006-08-24 08:43:01 -0700, Ronald Wiplinger [EMAIL PROTECTED] said:
I want that each call from PSTN goes to Asterisk to the context for
this line. Within this context can be a menu or a dial command, ...
As more I read, as more I get confused, ... and each try is not working!
I don't know
On 2006-08-24 08:43:01 -0700, Ronald Wiplinger [EMAIL PROTECTED] said:
I want that each call from PSTN goes to Asterisk to the context for
this line. Within this context can be a menu or a dial command, ...
As more I read, as more I get confused, ... and each try is not working!
My sip.conf:
On 2006-08-26 18:35:27 -0700, Tzafrir Cohen [EMAIL PROTECTED] said:
On Sat, Aug 26, 2006 at 02:02:51PM -0700, Martin Joseph wrote:
On 2006-08-22 01:59:09 -0700, Tomislav ParÄina [EMAIL PROTECTED]
said:
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.
5
On 2006-08-28 00:30:22 -0700, Martin Joseph [EMAIL PROTECTED] said:
On 2006-08-24 08:43:01 -0700, Ronald Wiplinger [EMAIL PROTECTED] said:
I want that each call from PSTN goes to Asterisk to the context for
this line. Within this context can be a menu or a dial command, ...
As more I read
On 2006-08-29 01:06:39 -0700, Tomislav Parčina [EMAIL PROTECTED] said:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
2) If the phone is answered on the first ring the call goes off to la
la land. Explaining to users (or myself) that you need to wait for
the
second audible ring on
On 2006-08-31 19:12:03 -0700, Xue Liangliang [EMAIL PROTECTED] said:
Hi, all.
I have a Adit 3104, and I configured it to work with Asterisk, the
voice quality is quite good, however it just randomly restart, I don't
know whether you guys have the same experience, is it due the firmware
On 2006-09-07 06:07:09 -0700, Nick Ellson [EMAIL PROTECTED] said:
Bruce,
I *just* tested the XtremePhone, IAX2 softphone. Other than trying to
figure out how to get it to send proper CallerID to the other phones,
it worked right off, in both directions. Excellent!
Perhaps working the IAX2
On 2006-09-06 20:10:11 -0700, Ma Zhiyong [EMAIL PROTECTED] said:
I use both IAX2 channels and SIP channels. IAX2 channels reduce
bandwidth effectively.
But sometime my cli show
NOTICE[1281]: chan_iax2.c:1628 iax2_destroy: Avoiding IAX destroy deadlock
WARNING[1281]: chan_iax2.c:708
On 2006-09-13 06:51:50 -0700, Zeeshan Zakaria [EMAIL PROTECTED] said:
I've followd the instructions as in tutorials, created folder stream,
created file stream.mp3, in musiconhold.conf added 'stream =
mp3:/var/lib/asterisk/mohmp3/stream,http://216.126.84.50:8000', and in
extensions.conf added
For all of us using these devices, I have some good news. There is a
self installable firmware update available from Nokia here (requires
windows box to install):
http://www.nokia.co.uk/nokia/0,1522,,00.html?orig=/softwareupdate
This seems to radically improve the behavior of the SIP
On 2006-09-15 13:42:21 -0700, Lists [EMAIL PROTECTED] said:
Not sure what to tell you. But for the price, I might have to try one
of these instead:
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