Hi All,
Someone can explain how that miracle free landline calls is made?
I´ve tried this with my server and it works, but...how they do it?
Miklos
IPFONE TELEFONIA IP
Rua Caio Graco 735 São Paulo SP
IPBX - +55 11 3488-3800
http://www.ipfone.com.br
[EMAIL PROTECTED]
Balbus balbum
Does anyone know how to use the Asterisk modules in external applications (C++)?
I must develop an application that encode/decode audio files from iLBC
or Speex to PCM and would like to use the Asterisk modules to do this.
But this application will not run in Asterisk box... I would like just
to
I will implement a SIP application and I'm considering using Asterisk for mixing the media
streams (audio). Does anybody know if Asterisk
supports or contains a RTP mixer? If so, how to use it?
Just to be a little more clearer: I will send to Asterisk more
than one RTP stream and they must be
I will implement a SIP application and I'm considering using Asterisk for mixing the media
streams (audio). Does anybody know if Asterisk
supports or contains a RTP mixer? If so, how to use it?
Just to be a little more clearer: I will send to Asterisk more
than one RTP stream and they must be
Hi All,
I need to use - mms://61.112.173.60:81/ as souce for MOH, i cant find
anything about using that souce format in wiki.
If you have some info please advice.
Miklos
IPFONE TELEFONIA IP
Rua Caio Graco 735 São Paulo SP
IPBX - +55 11 3488-3800
http://www.ipfone.com.br
[EMAIL
Hi All!
I need some feedback about the edge-core sip phones, somebody uses it?
They are reliable?
What the community say about them?
Miklos
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or
Try the new conversion module from redice li ..it is greate!
Miklos
IPFONE TELEFONIA IP
Rua Caio Graco 735 São Paulo SP
IPBX - +55 11 3488-3800
http://www.ipfone.com.br
[EMAIL PROTECTED]
Balbus balbum intellegit
- Original Message -
From: Innocent Evil [EMAIL PROTECTED]
To: Asterisk
Hi Kristian,
I installed 0.2.9 today ..it is grate...the zaptel / ztdummy issues are gone
an the systems are going very well.
Thanks and congratulations for the always good work.
Have you seem that new grafical interface using ruby? maybe it can be
integrated in astlinux...what you think
--- listas iPfone [EMAIL PROTECTED] wrote:
Hi all
I need to know if the video support for h.263 is
active in version stable
1.0.7 to use with eyeBeam in asterisk
it works for me...
[]
type=friend
secret=
auth=md5
callerid=myCallerId
canreinvite=no
host=dynamic
disallow=all
Hi All,
I´m tryingo to install asterisk in an PROLIANT ML 150 G2 SATA and can´t make
it work because linux cant recognize the Hd (HP 160 mb).
No drivers for Centos ...Red Hat... i´t´s drivig me crazy..
Someone have a tip? if i make change it to SCSI i think it will work but not
sure
Hi all
I need to know if the video support for h.263 is active in version stable
1.0.7 to use with eyeBeam in asterisk
In the wiki the info is that this support is from CVS HEAD 02/25/2005
Thanks
Miklos
___
Asterisk-Users mailing list
Did you checked the outbound proxy parameter?
- Original Message -
From: David Sampson [EMAIL PROTECTED]
To: Asterisk-Users@lists.digium.com
Sent: Wednesday, May 04, 2005 4:05 PM
Subject: [Asterisk-Users] IP500 Registration
Hello -
I have an IP500 (my first). The phone is up and
Hi Max!
I am Begining in the ASTERISK IP-PABX world, and here in Brazil, have not
any Help to install and configure,
Sure you have!:
http://www.ipfone.com.br/curso.asp
Miklos
- Original Message -
From: Max [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
I read somewhere that it works but you must
recompile the kernel...
there is some
info in:http://www.voip-info.org/tiki-index.php?page=Asterisk%20Data%20Configuration
look for a thread with this subject (it has
plenty of info)
[digium.com #12961] T100P as bandwidth
hope this
Miguel,
you can try using # as a way of transfering the call, but that's a blind
transfer meaning that you will be prompted an extension number and the call
will be transfered and that's it, on the other hand pressing flash put's the
call on hold and then let's you dial another call and if you
You can Dial() extension SIP/line1SIP/line2
even more you can and that will call both extensions only after a 5 seconds
timeout
exten = xxx,1,Dial(SIP/line1,5)
exten = xxx,2,Dial(SIP/line1SIP/line2,10)
etc...
that's if I understood what ou needed...
bye,
M.
- Original Message -
From:
Julian,
I'm also following this issue, so I guess you're not alone in the universe,
even more I'm not sure why nobody's following this issue usefull as it
seems.
Anyway we'll probably start working on it soon if this happens I'll let you
know.
What I'm not sure is why this didn't make it to the
You should use Firefly Third party edition
I'm using it in IAX2 and it works fine
bye,
M.
- Original Message -
From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, December 25, 2004 6:29
Hi,
I have 3 PAP2 connected to*, they work fine
but there are some things which I would like to improve, some of them
are:
- double ring tone when placing
a call (I hear two tones it seems like the PAP2 is generating it's own
tone)
- some kind of noise (like
glitches or something) when I
I tried to configure a TE110XP using HDLC (mainly to see if it worked) I
called the guys at Digium because HDLC support is not natively compiled,
they told me that HDLC was not covered in the free support provided with the
card, I had to rebuildl the kernel but as I had not that much time
Ok I forgot to ask if any of you out there have
fought against any of this issues and have any information that can (and has the
will) to share... or if any of you has any kind of documentation about
this.
thanks again,
Matias
- Original Message -
From:
Listas
Hi!
Use the spa2000 configuration info, the software is the same.
Miklos
- Original Message -
From: Listas [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, December 23, 2004 4:46 PM
Subject: Re: [Asterisk-Users
ok thanks
- Original Message -
From: listas iPfone [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, December 23, 2004 4:03 PM
Subject: Re: [Asterisk-Users] Linksys PAP2-NA Config
Hi!
Use the spa2000
I recently bought 3 of them and I'been using them with no problems at all,
but in the testing phase I had some trouble with one of them getting
disconected (just like if it was resetting itself, with the blue light
turning red and all that stuff) but then I changed the switch and everything
went
Hi I would like to know which is the last stable version of asterisk and how
to get it from the CVS, I mean rather than doing
cvs checkout -r -v1-0_stable asterisk
is there something for a newer stable version?
thanks,
Mat.
___
Asterisk-Users
Hi all
I have question regarding to my nom 200 and asterisk.
I have an * server with two x101p and two lines conected.
When i am in a call in line 1 and a call in line two is received the first
call goes imediatly to hold and the line button blinks indicating that
another call arrived.
It is
Hi!
I´m trying to use firefly 3 party with * and iax2.
I cant figure out why it reapeats every call many times until it is closed.
It is a bug ?
I want it because of the skin changing thing..
Someone have a clue on how to use it with *
Thanks
Miklos
- Original Message -
From:
Hi!
You can do this in the web interface sip conf local settings Digitmap
You can map the number of digits to be dialed before sending..etc...
miklos
- Original Message -
From: Tor Setane [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Thursday, July 29, 2004 9:26
Hi all,
after a good time trying i made the optipoint work with asterisk...
this is very strange but.. maybe someone can do it and tell me what happens:
I have two peers in sip.conf :
[19]
accountcode=19
amaflags=billing
type=friend
username=19
secret=
host=dynamic
nat=yes
qualify=1000
This is very interesting...
Regulations..USA...
But... what can i do faking a caller id? stolen what? what is the point?
miklos
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 07, 2004 12:56 PM
Subject: Re: [Asterisk-Users] VoIP
Hi!
Yes we have many kinds of phones hwere in the show room, snom, polycom,
cisco, grandstream, ipdialog, intracom, d-link, symbol all of them works
with asterisk with some testing and with some issues ...but works.
The optipoint is the only one that i´m really can´t make work till now.
In
Hi!
callerid=br exists?
miklos
- Original Message -
From: Jason Williams [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 22, 2004 9:06 AM
Subject: Re: [Asterisk-Users] No Caller ID from FXO Problem
At 14:39 22/06/2004 +0300, you wrote:
I've compiled and run it but no
Hi!
I have updated the optipoint to the last software version
I can Call the optipoint from other phones and talk.
The optipoint register with asterisk but in the phone display i have
only no server. and no dial tone.
The only way to register was with no password to the optipoint
for the project. I want to try out the VOCAL
footprint too but didn't had the time to do that yet.
Stefan
listas iPfone wrote:
Hi All,
I have a thin cliente here that i want to run asterisk:
- National Semicondudor Geode GX1 266MHz Geode 266MHz single chip
- NS Cx5530a Southbridge
Hi
That rescue disk sugestion seems to be very good...
Let´s see if i undestood:
1. burn the rescue iso
1. copy the rescue disk to a hard drive
2. compile asterisk
3. copy all to the flash disk
It is that simple?
Miklos
- Original Message -
From: Klaus-Peter Junghanns [EMAIL
Hi All,
I have a thin cliente here that i want to run
asterisk:
- National Semicondudor Geode GX1 266MHz Geode
266MHz single chip- NS Cx5530a
Southbridge National Semiconductors
SC2200- NS PC97317in
chipset- 32MB Compact
Flash - 64MB Ram- 10/100Mbps, Autosense
10/100Mbps, Autosense
Hi !
it was designed for our receptionist
Please post a picture of that recepcionist .. maybe she can be the
asterisk girl of 2004!
Claudio
- Original Message -
From: Kyle Hagan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 03, 2004 6:21 PM
Subject: Re:
Hi all,
I just upgrade my ix66 ...
the new firmware 2.07 have this:
(SIP) Tolerance against Asterisk PBX registration
deviation.
regards
Miklos
Audiocodes MP124
- Original Message -
From: Michael Welter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, May 18, 2004 12:45 PM
Subject: [Asterisk-Users] ATA devices
Does anyone know of a 24 port ATA device that could be installed in a
phone closet? Like a channel bank,
Hi!
I know that is a very posted matter but i have a
question:
Some one can translate that messages for me? what
is the mean of that messages? can i do something to correct this and get
the caller id to work?
May 7 11:26:19 ERROR[1288925632]:
callerid.c:192 callerid_feed: fsk_serie made
Symbol have the netvision line of h.323 wireless phones used in hospitals
with multiple logins etc... , i have one here in my office and it works very
well with a simple 3com officeconnect gateway, makes direct calls, have
integration with various pbx.. a good product.
www.symbol.com
Miklos
SIP Scenario Generator
http://www.ipc.com/
runs under windows
Miklos
- Original Message -
From: Brancaleoni Matteo [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, May 07, 2004 2:51 PM
Subject: Re: [Asterisk-Users] SIP Wokflow diagram
I use callflow (callflow.sourceforge.net)
=alaw
;allow=gsm
;allow=g729
;allow=ilbc
I´m doing something wrong here??
miklos
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, April 23, 2004 7:18 PM
Subject: Re: [Asterisk-Users] WARNING[1074420448]
listas iPfone wrote:
plase
I downloaded the cvs development of zaptel...
It seems to compile ok, but when ismod I get:
lp:/usr/local/src/zaptel# /sbin/insmod wcfxo.o
wcfxo.o: unresolved symbol zt_ec_chunk
wcfxo.o: unresolved symbol zt_unregister
wcfxo.o: unresolved symbol zt_alarm_notify
wcfxo.o: unresolved symbol
You where SOO right.
My idea is to connect the line to my pbx and call from internet
(h323?) to my linux box, and then dial an extension.. Is there any doc
there? Cause Ive read a few and I did not get much really. I should
configure asterisk now and the job is done?
-Mensaje original-
Hi list
I have configured some siemens optipoint 400 sip to work with asterisk.
I works very well with messages, moh etc... a good choice in my opinion...
Someone else have good/ bad experiences with that phones?
Miklos
___
Asterisk-Users mailing
Olá Ana,
Estou aguardando as informações sobre nosso acordo de revenda
Atenciosamente
Cláudio
- Original Message -
From: Adam Goryachev [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 31, 2004 5:25 AM
Subject: Re: [Asterisk-Users] Noises and echo effects
Doesn't
Hi!
Every time i make or receive a call with my x100p i
receive that notice:
NOTICE[1234379840]: chan_zap.c:3640 zt_read: Fax
detected, but no fax extension
Maybe that is problem with brazilian
lines?
How can i stop it?
Miklos
iPFONE Telefonia IPRua
Caio Graco 735 São Paulo SP iPBX
Hi All!
I have this problem with callerid detection with my
x100p here in brazil., my line have this function and it works with a very cheap
aplliance that i have here in the office, here in brazil it is called
"detecta".
Ithink that the caller id info comes in DTMF
before the 2 ring of
needs to spend the time and programm it into the zaptel
driver.
Alfred.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of listas
iPfoneSent: Tuesday, February 10, 2004 8:20 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Callerid
Hi All!
I´msearching fora compact external fxo
device , a little box like sipura adaptor,with one or maybe
two fxo.
Searching google the only device that shows is the
x100p,
Anyone knows about a device like that?
miklos
Snom Does gives the souce and more:
http://www.snom.com/sources_en.php
- Original Message -
From: Chris Albertson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, February 03, 2004 4:01 PM
Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
I read a report of
Hi!
If the number of calls are really greate maybe you are listed in the fwd
welcome (5) line by mistake...
Miklos
- Original Message -
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 29, 2004 9:53 AM
Subject: Re: [Asterisk-Users] Junk calls from
Hi Jeroen1
I think that´s maybe a bug
I really don´t found the problem in my logs, i´m starting it by hand :-(
I update you if i can figure it out.
regards
Miklos
- Original Message -
From: Jeroen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 26, 2004 11:23 AM
Ok!
Thanks
miklos
- Original Message -
From: Karsten Wemheuer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 24, 2004 9:42 AM
Subject: Re: [Asterisk-Users] rc.local dont works
Hi Miklos,
listas iPfone wrote:
Hi ! thanks for the answer..
I use rh9...
Sorry
I use it in that way, it works very well:
exten = s,4,AbsoluteTimeout,600
miklos
- Original Message -
From: Wes Marderness [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 12:33 PM
Subject: [Asterisk-Users] SIP Absolute Timeout
Hi All,
I've been having a hard
Hi
I sugest you to make a reset and switch off the phone before upgrade.
It solved many problems for me.
Miklos
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 11:32 AM
Subject: Re: [Asterisk-Users] Snom 200 phones not
Hi
All
I have a problem with initialization of asterisk
using my rc.local file. when i call asterisk from the prompt it works well but
don´t in the initialization...
I have in my file that comands:
touch /var/lock/subsys/localmodprobe
zaptelmodprobe wcfxosafe_asterisk
I read in
Wemheuer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 5:07 PM
Subject: Re: [Asterisk-Users] rc.local dont works
Hi
listas iPfone wrote:
Hi All
I have a problem with initialization of asterisk using my rc.local
file. when i call asterisk from the prompt
Hi All!
I installed * in RH9 with yesterday cvs and i
have a x100p in that system.
My problem is that when rh9 loads, it loads the
zaptel modules ( wcfxo and the usb driver) automagically, and when it
calls my rc.local with:
modprobe zaptelmodprobe
wcfxosafe_asterisk
asterisk dont
Hi all!
I get this error when trying tostart
asterisk:
ERROR[8192]: File asterisk.c, Line 1349 (main):
Unable to connect to remote asterisk
What can be the problem?
Thank you!
Miklos
iPFONE Telefonia IPRua
Caio Graco 735 São Paulo SP iPBX +55 11 3801-3702UK +44 870 -
3403539FWD
I think that it will be greate to include * inside of a router like ix66
from intertex... 1 GB usb removable flash to record voice mail.and prompts
in the computer..2 fxo...real internal sip server ...internal dns
server..good user interface.. all nat / firewall nightmare ended, no
computers to
HiList !
I received an unit of the Symbol
NetVision Phone and i will test it with asteriskusing H.323 or Skinny , somebody tested thisphone
with asterisk and can share experience?
Miklos
Hi!
Last week i talk to a person in senegal (i´m in brazil) with a 64 Kbs
sattelite link and the latency was about 10 seconds!
Like you are talking to the moon.
miklos
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 17,
Thanks for all!
It is working now :-)
Regards
Miklos
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 13, 2003 3:16 PM
Subject: Re: [Asterisk-Users] Mysql CDR
On Saturday 13 December 2003 11:02, Mireia Munoz de jesus wrote:
Hi!
i just tryied the 2.03b firmware.
Now i have that message when the phone boots:
Challenge User: 6466212364662
64662
pressing ok the display shows PW: iputmypassword
When i put my password i get a loop returning for Challenge User:
6466212364662 again
64662 is my FWD number
Now the
Hi
The version 1.260 of chan_sip.c already have that patch?:
http://bugs.digium.com/file_download.php?file_id=430type=bug
thanks!
Miklos
- Original Message -
From: Leif Madsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 28, 2003 2:10 AM
Subject: [Asterisk-Users]
4
PMTo:
[EMAIL PROTECTED]Subject: Re: [Asterisk-Users] snom X
MOH
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 fi
Hi!
I have one ipdialog working well with cvs 10/09 but with latest cvs i have
the same problem.
regards
miklos
- Original Message -
From: Ariel Batista [EMAIL PROTECTED]
To: Asterisk User List [EMAIL PROTECTED]
Sent: Tuesday, December 09, 2003 5:34 PM
Subject: [Asterisk-Users]
Hi all!
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 noMOH at all..( with the transfer button,
moh playsusing a extension).
Someone with that problem?
I downgrade to 2.01s but
Hi!
I need help to undestand the options:
externip= static/ dynamic ip? can be a domain?
localnet= internal ip of * machine?
localmask= 255.255.255.0 ?
Thanks
- Original Message -
From: Leif Madsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003 7:25 AM
Hi all!
We set up a sipserver using asteriskX ix66
and need some test calls from around world toverify if it is working
ok.
If you can :-)please call us:
sip:[EMAIL PROTECTED] direct to
snom200
or
sip:[EMAIL PROTECTED] to asterisk
snom200
Thank´s for all
Miklos
iPFONE Telefonia
Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 4:02 PM
Subject: Re: [Asterisk-Users] test call request
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote:
Hi all!
We set up a sipserver using asterisk X ix66 and need some test
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote:
Hi all!
We set up a sipserver using asterisk X ix66 and need some test calls
from
around world to verify if it is working ok.
If you can :-) please call us:
sip:[EMAIL PROTECTED] direct to snom200
or
sip:[EMAIL
Hi All
I signed up for an account with voicepulse connect
service and received the info to set up asterisk.
Anyonehave that confs to send as an example?
Thanks
Miklos
Title: Mensaje
Try this guide:
http://www.automated.it/guidetoasterisk.htm
Miklos
- Original Message -
From:
Sergio Serrano Revuelto
To: [EMAIL PROTECTED]
; [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Thursday, November 13, 2003 8:02
AM
Subject: RE:
Hi all!
I´m testing an intracom sw netphone with asterisk,
someone have one netphone or have any experience to share about?
miklos
Hi!
How to use that externip new parameter?
Where in sip.conf and what is the format?
thanks
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 03, 2003 3:34 PM
Subject: Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
I have the same problem and it was solved setting:
# Uncomment for aggressive residual echo supression under
# MARK2 echo canceller
#
KFLAGS+=-DAGGRESSIVE_SUPPRESSOR
in the makefile of zaptel and recompiling.
miklos
- Original Message -
From: Ernest W. Lessenger [EMAIL PROTECTED]
Hi!
try to use in sip.conf :
register =x:[EMAIL PROTECTED]/xx
[iconnect]
type=friend
secret=
username=xxx
host=sipauth.deltathree.com
dtmfmode=inband
context=yourcontext
and in extensions.conf:
exten = _7X.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
This works for me
regards
Miklos
Hi!
I don´t have an inbound number to, this registration is for an outbound
account
sorry if i don´t explain better in he first time
register=username:[EMAIL PROTECTED]/extension
hope this helps
miklos
- Original Message -
From: rnc Info Lists [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Hi!
where i can find info about using gastman and astman?
Thanks!
Miklos
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi All
I receive thatwarning message:
WARNING[49159]: File chan_sip.c, Line 2220
(__transmit_response): Unable to determine sequence number from
''
What is it?
There is some documentation with all error
messages?
thanks
miklos
]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 14, 2003 12:39 PM
Subject: Re: [Asterisk-Users] WARNING[49159]
It means that your SIP device sends some SIP packets and we can't parse
the CSeq numbers. Can you paste the 'sip debug' of that ?
regards
Martin
On Tue, 14 Oct 2003, listas iPfone
to the PSTN?
G
At 07:35 AM 10/9/2003, listas iPfone [EMAIL PROTECTED] wrote:
Hi all,
When i receive a call from pstn ( calls from sip works well) my phone
shows
asterisk and not the number of the phone.
How can i make asterisk show the phone number of the person who caled?
thanks!
Miklos
-Users] my phone shows asterisk
listas iPfone wrote:
Hi!
My setup is:
pstn X100PASTERISKSNOM 200
thanks
miklos
- Original Message -
From: Gerry Boudreaux [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 09, 2003 8:12 PM
Subject: Re: [Asterisk-Users] my
: Re: [Asterisk-Users] my phone shows asterisk
listas iPfone wrote:
Hi!
Thanks for the advice i will do it.
There is a way to know if the CallerID enabled from my telco is
compatible
with asterisk?
regards
Miklos
I guess if it is enabled and it does not work then chances
Hi all,
When i receive a call from pstn ( calls from sip works well) my phone shows
asterisk and not the number of the phone.
How can i make asterisk show the phone number of the person who caled?
thanks!
Miklos
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Hi All,
I´m thinking in install apache in my asterisk machine to host a litle site.
Anybody knows about problems doing that?
thanks
miklos
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi!
I have some question about the use of codecs in sip.conf
I have that lines in sip.conf:
disallow=all
allow=gsm
allow=ulaw
allow=alaw
when i use show codecs:
localhost*CLI show codecs
1 (1 0) G.723.1
2 (1 1) GSM
4 (1 2) G.711 u-law
8 (1 3) G.711
Hi!
I´m thinking inan incoming number from
ICH
please share your sip and extensions.conf files off
list, it will help me a lot.
miklos
- Original Message -
From:
Glenn
Dalgliesh
To: [EMAIL PROTECTED]
Sent: Friday, October 03, 2003 2:17
PM
Subject:
Hi!
Anybody have experience using asterisk and 3com voip systems?
Miklos
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[EMAIL PROTECTED]
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Hi All
I have that error message:
WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno 102
(Request)
What can be the problem?
Thanks!
miklos
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[EMAIL
Hi Martin
Please explain, why did you send the messages?
miklos
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 02, 2003 2:04 PM
Subject: Re: [Asterisk-Users] error message 49159
Martin Pycko wrote:
We send SIP messages to
Hi!
I have that message:
*CLI WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 177 (Request)
I was thinking..why that call is for 127.0.0.1 is it the loopback of the
asterisk machine?
Thanks for any help
Miklos
Hi!
I have a strange problem with ICH calls.
When i try to make a call with asterisk for ICH nothing happens ( register
is ok)
But when i register my snom 200 with ich it works very well with the same
register data.
Someone knows anything about?
miklos
PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of listas iPfone
Sent: Tuesday, September 30, 2003 3:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] I have a strange problem with ICH calls
Hi!
I have a strange problem with ICH calls.
When i try to make a call
Hi !
I´m using * with a snom 200 phone, i can use FWD but cant use ICH.
Someone can tell me if my setup is correct?
sip.conf:
register =user:[EMAIL PROTECTED]/33
extensions.conf:
exten = _7X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
In CLI the registration is ok but when i try ex.
Oi Adriane!
Minha mãe foi internada hoje de madrugada no 9 de julho por causa de um
problema de estomago..
Já viu que não vou conseguir ir hoje tb.
Já estou de pé desde ontem a noite.
arrumei um micro aqui no hospital para te escrever. esqueci meu celular em
casa.
Amanhã ainda dá tempo né? eu
Hi!
Thaanks the problem was the same, now i´m using a static ip and all is
working fine.
regards
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 25, 2003 4:07 PM
Subject: RE: [Asterisk-Users] ERROR MESSAGE
I had this
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