no need to show you the
extensions.conf setup. Any ideas what can it be wrong that i get only
trunking on one side? thank you guys
p.s. im getting the timers on dallas from ztdummy im using kernel
2.6.12-10-686-smp and im getting the timer on mexico from the digium
card.
Miguel Cavazos
Hi guys I have a question, im trying asterisk realtime in 2 servers.
Im trying to make calls from one server to another, example I call a
sip registered in sip server 1 with a phone register in sip server2
and both using the same database and family both use canreinvite=yes
but still cant
Hi guys, im using realtime and I want to show registered users or
online users on a webpage and offline users. Im taking regseconds
field to make this happend
If regseconds value is 0 then user appers offline, it regseconds is
something else then its online, but sometimes this works and
well
--
Saludos,
Miguel Cavazos
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Saludos,
Miguel Cavazos
your setup to a working level ?
/Sam
- Original Message -
From: Miguel Cavazos [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, January 13, 2005 1:22 AM
Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test
for possible reservations and perhaps ticket information
for the
theater.
- Original Message -
From: Miguel Cavazos [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Wednesday, January 12, 2005 4:22 PM
Subject: [Asterisk
On 13/01/2005, at 9:35 AM, Miguel Cavazos wrote:
Really weird calls are still getting in and i just called the same
number as you did. I will investigate.
here is the context on extensions.conf
[guest]
exten = _,1,Dial(Unicall/g1/${EXTEN},90,Tt)
On 13/01/2005, at 9:22 AM, Gary Carr wrote
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Saludos,
Miguel Cavazos
for routes into Mexico
once you where done testing, if so, could you please contact me off
list with your rates for Mexico City, or anyplace else in Mexico you
service, thank you.
Nathan Goodwin
Diamondleaf LLC
Miguel Cavazos wrote:
Hi guys, I have one E1 with 30 channels in Mexico City, I guess
Yes, thats why i will do it for a very short time to do testing with
real traffic.
On 13/01/2005, at 4:03 PM, Nathan Goodwin wrote:
Wouldn't that make routing free calls illegal as well, your still
bypassing?
Miguel Cavazos wrote:
Thanx but that is consider in Mexico bypass and its illegal
make FREE LOCAL calls to Mexico City till saturday or
maybe until monday to see how stable this can be with REAL traffic. Add
this to your extensions.conf only gsm as a codec is going to be
permitted.
exten =
_,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
--
Saludos,
Miguel
I do have a WISIP and it doesnt give me any problems im all day long on
the street using it. You cant talk of a phone you havent even touch
Miguel
On Fri, 2004-04-23 at 10:33, Andrew Kohlsmith wrote:
why not wisip? its size its like a regular cellphone and it uses wifi
Because it sucks ass?
why not wisip? its size its like a regular cellphone and it uses wifi
Miguel Cavazos
On Fri, 2004-04-23 at 08:00, Chris Hirsch wrote:
Tim Sailer wrote:
Folks,
I'm looking for a SIP or IAX phone for field techs to take with them
when out on service calls. The regular desktop phones
IAX was removed on newer versions replace it with IAX2 just make sure to
change on your extensions.conf IAX/user:[EMAIL PROTECTED]/201 to
IAX2/user:[EMAIL PROTECTED]/201
Good luck
Miguel
On Wed, 2004-04-21 at 03:44, Jan Madsen wrote:
I have been running af Asterisk server Version 0.7.2 for a
update your crappy hardware :)?? atleast with sip you will be able to
allow both codecs.
Miguel
On Mon, 2004-04-19 at 07:51, Serge wrote:
Hello list,
I have many IP hardphone like Siemens 300 basic ( old ) , cisco ata..
etc
I need: G711 from old phones must be convert to G729 via
Yes it works with normal digium hardware
Miguel
On Thu, 2004-04-01 at 08:51, Otto Krumm wrote:
I was wondering if anyone has setup an * connected to E1
in Mexico?, what card would you recomend and do you have some info,
examples or everythig else... or for instance this setup
si funciona con el A y B
Miguel Cavazos
On Thu, 2004-03-25 at 22:47, Carlos Chavez wrote:
I see that I can purchase G.729 licenses for my Asterisk server, but I
have seen that many phones support a G.729 variant like A or B. Are these
suppoted by the same G.729 codec in Asterisk
I just compile -Stable cvs but when I dial an extension I can't hear the
phone ringing, but the phone is really ringing and if someone picks it
up you can talk with the person but i cant really listen when the phone
is ringing sip to sip didn't try anything else
Miguel
a news group could be less flood
Miguel
On Fri, 2004-03-19 at 19:00, Andrew Kohlsmith wrote:
How about using a web form for posts instead of replying to an e-mail? A
How about not.
Regards,
Andrew
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no it wont happend with zap cards or other sipphones such as grandstream
and wisip.
Miguel
On Tue, 2004-03-16 at 09:10, Senad Jordanovic wrote:
Miguel Cavazos wrote:
hello guys heres my setup i have 2 asterisk servers in 2 different
houses sipura ulaw --- asterisk --- iax2 (ilbc
if it was related to the dsl line i would notice my other phones such as
grandstream and the ones on zap cards with the same problem im only
having this issue with sipura.
Miguel
On Tue, 2004-03-16 at 12:00, Senad Jordanovic wrote:
Miguel Cavazos wrote:
no it wont happend with zap cards
is this fixed on cvs -stable branch??
Miguel
On Tue, 2004-03-16 at 23:02, Andres wrote:
Miguel Cavazos wrote:
hello guys heres my setup i have 2 asterisk servers in 2 different
houses sipura ulaw --- asterisk --- iax2 (ilbc) --- asterisk -- sipura
ulaw, this is my setup but when i call
thanx for the review michael, could you send some pictures of the phone?
can you tell how long does the battery lives? signaling what do the
menus have how do you configure it etc? maybe after you do a full
testing we can do a Wisip vs. IPC5000 working futures.
Miguel Cavazos
On Mon, 2004-03-15
asterisk but i just get this problem with
sipuras.
Miguel Cavazos
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you buy the unit thats what its call a test unit ipc5000 looks great and
its 28 USD more than wisip i think the lcd is worth
Miguel
On Thu, 2004-03-11 at 19:58, Craig Waddington wrote:
Thanks for the info. Sounds good.
Does that mean I can contact them for a test unit also, to try before
had several call this week from access points on the street and I
think that the voip future is wifi. However the battery on wifi phones
wont be so good for now since many wifi devices eat batterys in a couple
of hours.
Buy IPC5000 the big lcd is worth the extra 28USD
Miguel Cavazos
On Thu, 2004
the phone works for the wlan600 its a great phone poor battery but even
palms with wifi use ALOT more battery when wifi is on and considering
this phone has the wifi ON all the time the 23 stand by hours and 3 hr
talk is ok
it registers with asterisk just fine, try get it from pulver
Miguel
On
ive been looking for a palm os5 client found gphone there webpage claims
to be sip but i just cant make it register against asterisk
Miguel
On Thu, 2004-03-04 at 07:05, Dean Collins wrote:
Does anyone know of a Palm OS5 client that can connect to asterisk?
Hopefully I can use gprs to connect
hi guys finally i got my wisip this week and im very happy with it. It
works but i was wondering anyone know where can i find new firmware,
updates or a wish list? I cross emails with jeff pulver about having a
small http browser for auth on starbucks hotspots mcdonalds or prodigy
movil(mexico).
support G711 64kbps)
Be carefull with what you buy.
Hector.
- Original Message -
From: Miguel Cavazos [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Sent: Sunday, February 15, 2004 9:38 AM
Subject: [Asterisk-Users] Wifi Phones
Hello list, I was going to buy
this model or
seen it working, sorry about the unnesesary traffic to the list, my
question is simple would this work against asterisk if anyone knows
any other Wifi phones besides Wisip and Ciscos expensive toy please tell
me.
Miguel Cavazos
how many simultanius calls does voiceglo permit???
Miguel
On Fri, 2004-02-06 at 01:28, Cameron Palmer wrote:
IAX is what they use with glophone. http://webphone.voiceglo.com. It is a
seperate server from the myphone.voiceglo.com SIP gateway. The IAX gateway
is msps01-nyc.voiceglo.com on port
would be a good idea to put it on the changelog, i see its there but it
doesnt really inform nothing.
Miguel Cavazos
On Wed, 2004-02-04 at 15:21, Mark Spencer wrote:
Asterisk 0.7.2 is now released and contains lots and lots of bug fixes
from the bug tracker. Highly recommended for people
same here, when i recive an incoming call from x100p to line 1 on
sipura, i can hear them but people can't hear me im using 1.0.24 on my
firmware
Miguel
On Sun, 2004-01-25 at 20:54, Chris Higgins wrote:
Frankie Gravato wrote:
I've been beating my head for 5 hours to figure out why my
check
http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html
this email has help me alot with BW and codecs
Miguel
On Mon, 2004-01-12 at 08:13, WipeOut wrote:
Hans-Henrik Andresen wrote:
Hi,
How much bandwidth do I need for 1 conversation ?
I know it depends on the
sad, yes but who needs speakfreely when you have asterisk and soft/hard
phones. The author seems really unmotivated so let him find the path
maybe he could join asterisk development team :)
Miguel
On Sun, 2004-01-11 at 19:33, Lists wrote:
If you did not see slashdot today, check out this
The following errors occurred during your registration:
* The username you entered as your referrer could not be located.
cant create a username
Miguel
On Sun, 2004-01-11 at 22:53, Brancaleoni Matteo wrote:
Hi
http://www.asterisk.bz Alternative to the asterisk-users list
sip phones have alot of nice features and they really work, you can try
some phones under $200 yes, but about the analog phones, people like to
have there cordless phones, or there micky mouse phone or garfield phone
so consider that.
You loss some features but your customers get the phones they
you need a sound card on the box that works
Miguel
On Thu, 2004-01-08 at 20:54, SamW wrote:
I have 2 installations of asterisk. On CLI one server has Dial
command. Other installation do not have Dial command on the CLI. What
I am missing. How to enable dial command from the CLI.
Thanks,
2 sipuras SPA2000, sold at 100USD each they have 2 FXS ports its like
cisco ata
Miguel
On Mon, 2004-01-05 at 07:38, Asterisk Newbie wrote:
Does anyone know of any inexpensive alternatives to the four port
analog module offered by Digium ($305) what work seamlessly with
asterisk?
Thanks
on the 5 to 8 secs of echo if your using on your zapata.conf
echotraining=yes you should get rid of this echo problem
Miguel
On Tue, 2004-01-06 at 19:15, Ryan R. Fligg wrote:
Hi,
Have posted to this list a couple of times and have always received
great responses and help. I have a
hi,
guys I have a weird, really weird problem with FWD (free world dialup),
My serveris a P2 400mHz 64 on Ram. This server is setup only to answer
to its FWD # for a friend to do calls to its local PBX. When i boot up
the server running debian Woody, with kernel 2.4.23, running asterisk
CVS from
On Wed, 2003-12-24 at 08:18, Brian Capouch wrote:
I have about a dozen Budgetone 101s and I'm pretty much satisfied with them.
Sorry, Brian; they've got some rough edges, but they're $65, for God's sake.
They are $65 yes, but you can get the best analog phones on the market
for that price and
merry xmas olle and you all in the list happy holidays!!!
Miguel
On Tue, 2003-12-23 at 23:04, Olle E. Johansson wrote:
It's the day before Christmas here in Sweden, actually the night before at this
time...
We celebrate Xmas on the 24th, so I'm about to log off and switch my Asterisk into
they can give out information for
this IAX firmware
Miguel Cavazos
On Wed, 2003-12-24 at 05:43, Brian West wrote:
Today class we are going to be talking about the wonderful line of
GrandStream products. Or should I say BarbieTone phones?
Who else is having MAJOR issues with the grandstream
On Thu, 2003-12-18 at 04:03, Brian West wrote:
Stop using beta firmware... I honestly think that GrandStream needs to
either fix the phones or stop making them.. THEY SUCKS! I think I would
rather eat glass than work with a grandstream phone.
bkw
Brian, GS has people that works very hard
well i never had a asterisk server with a monitor or keyword all my
servers i do remote login with ssh its better more private.
Miguel
On Thu, 2003-12-18 at 22:02, Michael Welter wrote:
Because of space limitations and because of the location of the
punch-down blocks, my * server is located on
good to hear theres going to be support for this phones, but why not put
it on the wiki??? so we can have all the faq in one place.
Miguel
On Mon, 2003-12-15 at 22:23, mattf wrote:
I just got off of the phone with Scott Willard at Polycom and things seem
promising. He's going to send me the
thats an old review jeff pulver said firmware was better now, is it
worth 250USD? im thinking of one
does it autodetect ssid? works with low signal? sound quality?
Miguel
On Sun, 2003-12-14 at 18:34, John Todd wrote:
http://www.loligo.com/asterisk/misc/WiSIP/
Works decently enough. Still
it doesnt work here, same firmware 4.26 tryed it with 4.18 also and it
doesnt work, i press any number and it gets screw up
i will try it with the handytone ata286 and see if it works, anyway its
the same firmware but its worth to try out
Miguel
On Sun, 2003-12-14 at 21:13, John Breeden wrote:
it's a firmware problem on GS, they are working on that but it seems its
not that simple to make volume higher on the speaker and echo go away,
anyway 4.26 seems stable for now and with many new features!
Miguel
On Fri, 2003-12-12 at 17:55, Bob Knight wrote:
John, did you ever get any feedback
biggest is DSL can now be connected directly to the GS no need of dhcp
or static ip no more, mute/del button works now the date tap on the
screen and many more cant remember download it at
http://www.grandstream.com/TEMP/FIRMWARE/
or update with 4.3.153.50 tftp
Miguel
On Fri, 2003-12-12 at
im using 1.0.4.24 and i havent seen any problems yet!
http://www.grandstream.com/TEMP/FIRMWARE/
Miguel
On Tue, 2003-12-09 at 04:11, TeleSIP wrote:
We are a service provider and are constantly being handed new firmware by
Grandstream to test on our network. Every time we ask when it will be
i have this very same problem but i have a different context
; Answering incoming calls
[incoming]
include = asterisk
exten = s,1,Wait,15
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Background(${SOUNDS}/casa)
exten = 1,1,Dial(SIP/101)
exten = 2,1,Dial(SIP/152)
here my problem is they can
On Fri, 2003-12-05 at 07:13, Steven Critchfield wrote:
During Phreaknic, Mark was showing off a Xbox running asterisk with 4
S100U interfaces connected to the game ports on the front. It was
interesting. In the end, I don't think it is cost effective as a real PC
since you can also build a PC
On Fri, 2003-12-05 at 11:17, Senad Jordanovic wrote:
I like you idea. Very Cool :)
Is RAM upgradable on xbox?
Thanks
no its not, BUT its very optimize the xbox hardware should work REALLY
REALLY GOOD i dont know how good it will be on long uptimes
Miguel Cavazos
about the hardware the xbox has HD 8 GB, ethernet port and 4 usbs a
nvidia video card and all you need is the usb adapters and some hours
reading on how to break in without a mod chip
using an xbox its a cool idea because its nice and small and cheap!
using the S100U was a great idea for the
Hello guys, i have been on this mailing list for some weeks now, and i
was wondering if someone here has installed linux on the XBOX and use it
as a dedicated server. Its a 200 USD computer and could make it perfect
to asterisk, its little and doesnt really take much space. My question
is could
if you have 80calls going, its time to think on getting a good dedicated
server, switches, for the work and UPS with big batterys also some good
power supplie:)
Miguel
On Fri, 2003-11-28 at 12:51, Matthew Asham wrote:
Hello,
I'm trying to figure out what portions of Asterisk need a lot of CPU
try base instalation of debian delete the documentation and asterisk
sources should be less than 150megas all
Miguel
On Wed, 2003-11-19 at 16:12, John Todd wrote:
At 3:41 PM + 11/19/03, WipeOut wrote:
Hi,
If anyone is looking for a small Asterisk installation I have
managed to get it
http://www.pulverinnovations.com/
On Tue, 2003-11-18 at 21:57, [EMAIL PROTECTED] wrote:
Where can I buy the Wifi600 phones ??
Or does anyone know of other Wifi SIP phones ??
Any help would be appreciated
Regards Mick
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anyone knows if this phones now support auth with sid ??? my school
wireless lan needs the auth
Miguel
On Tue, 2003-11-18 at 21:57, [EMAIL PROTECTED] wrote:
Where can I buy the Wifi600 phones ??
Or does anyone know of other Wifi SIP phones ??
Any help would be appreciated
Regards Mick
i followed what you said didint work heres what console says i cant do
the 1800 call anyway
-- Executing Macro(SIP/101-8376, callerid-pstn) in new stack
-- Executing SetVar(SIP/101-8376, SIP_CODEC=g729) in new stack
-- Executing Dial(SIP/101-8376, SIP/[EMAIL PROTECTED]) in new
stack
-- Executing SetVar(SIP/101-a9e5, SIP_CODEC=g729) in new stack
-- Executing Dial(SIP/101-a9e5, SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
== No one is available to answer at this time
On Tue, 2003-11-18 at 22:50, Barton Hodges wrote:
[EMAIL PROTECTED] wrote:
i
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