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 intellegit
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 PROTECTED
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
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Asterisk-Users mailing list
To UNSUBSCRIBE or upda
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: "Aste
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 abo
[Asterisk-Users] VIDEO ON 1.0.7 stable
- Original Message -
From: "Nardis Dome" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, May 26, 2005 1:59 PM
Subject: Re: [Asterisk-Users] VIDEO ON 1.0.7 stable
--- li
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 about...
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
Asterisk
Did you checked the outbound proxy parameter?
- Original Message -
From: "David Sampson" <[EMAIL PROTECTED]>
To:
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 running and I am able
to make out
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 Discuss
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"
Sent: Thursday, December 23, 2004 4:46 PM
Subject: Re: [Asterisk-Users] Linksys PAP2-NA Config
O
Polycon SoundPoint IP3000, but it´s h.323
- Original Message -
From: "Jay Milk" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, August 23, 2004 11:55 AM
Subject: [Asterisk-Users] SIP "unphones"
> Does anyone know if there are additional SIP devices out there which
> aren't ph
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 ve
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: "Joz
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
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
context=
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] V
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 b
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 t
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 pee
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 PR
l messages can float the system. And if SIP is only required
> you should probably use SER 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 ha
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 PC97317 in
chipset - 32MB Compact
Flash - 64MB Ram - 10/100Mbps, Autosense
1
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: [Ast
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 ba
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
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
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 m
ecs
;allow=ulaw
;allow=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] WARNIN
Hello all,
I just installed the h.323 drivers, make all the
process etc.. and i get that error message and asterisk dont
load:
[chan_h323.so]Apr 23 17:16:39 WARNING[1074420448]: loader.c:239
ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object
file: No such file ordire
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 lis
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
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 iP
Hi All!
What this message means?
Feb 19 18:33:14 WARNING[1142106560]: chan_sip.c:471
retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Request)
192.168.0.34 is my * box...
thanks for any help
Miklos
Ok!
I hope some *guru can make it soon... :-)
but i´m happy to know that my guess is
correct!
thank´s
Miklos
- Original Message -
From:
Alfred R.
Nurnberger
To: [EMAIL PROTECTED]
Sent: Tuesday, February 10, 2004 12:48
PM
Subject: RE: [Asterisk-Users] Ca
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".
I think that the caller id info comes in DTMF
before the 2 ring of
Hi All!
I´m searching for a 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
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 f
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 !
From: "Karsten 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 initializat
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
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
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 havin
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 s
Hi all!
I get this error when trying to start
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 w
Hi List !
I received an unit of the Symbol
NetVision Phone and i will test it with asterisk using H.323 or Skinny , somebody tested this phone
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
Hi All
I´m trying to use asterisk and ser in the same
box.
When i start ser my phones don´t connect with
asterisk anymore.
i have two nics in this machine
192.168.0.31/37
I need to set asterisk and ser to listen in
diferente adresses or ports?
I can use the two softwares at the s
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 p
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 w
Hi all
I just installed the mysql cdr support and my
database is not registering the calls :(
using show modules i see that the cdr_csv.so and
the cdr_addon_mysql.so are loaded
It is necessary to unload the cdr_csv.so? how
to do it?
in crd_mysql.conf i have:
[global]hostname=local
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
IL PROTECTED] Sent: Monday,
December 08, 2003 3:44
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
Hi
The version 1.260 of chan_sip.c already have that patch?:
http://bugs.digium.com/file_download.php?file_id=430&type=bug
thanks!
Miklos
- Original Message -
From: "Leif Madsen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, November 28, 2003 2:10 AM
Subject: [Asterisk-Us
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 no MOH at all..( with the transfer button,
moh plays using a extension).
Someone with that problem?
I downgrade to 2.01
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
[EMAIL PROTECTED]
> 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 calls
from
> around world to verify if it is worki
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 s
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 PROTECTED] > to asterisk
>> snom200
Thank´s for all
Miklos
iP
Hi All
I signed up for an account with voicepulse connect
service and received the info to set up asterisk.
Anyone have 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 all!
Every time i receive a sip call MOH begin to
play and i can´t talk to the caller.
My setup is the default.
Someone knows what is the problem?
thanks
Miklos
iPFONE Telefonia IPRua
Caio Graco 735 São Paulo SP iPBX +55 11 3801-3702FWD 64662ICH
31451543www.ipfone.com.br[EMAIL
Hi!
where i can find info about using gastman and astman?
Thanks!
Miklos
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
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 PROTE
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
,CANCEL,OPTIONS,SUBSCRIBE,INFO,NOTIFY
Content-Length: 0
11 headers, 0 lines
localhost*CLI>
- Original Message -
From: "Martin Pycko" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, October 14, 2003 12:39 PM
Subject: Re: [Asterisk-Users] WARNING[49159]
Hi All
I receive that warning 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
ober 10, 2003 10:10 AM
Subject: 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 ast
03 8:08 AM
Subject: Re: [Asterisk-Users] my phone shows "asterisk"
> listas iPfone wrote:
>
> >Hi!
> >
> >My setup is:
> >
> >pstn > X100P>ASTERISK>SNOM 200
> >
> >thanks
> >
> >miklos
> >- Original Message -
&g
"
> What hardware are you using to connect 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
> >"a
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
___
Asterisk-Users mailing list
[EMA
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!
Anybody have experience using asterisk and 3com voip systems?
Miklos
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi!
I´m thinking in an 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: [Aste
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.
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 messa
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 PROT
t; > -Original Message-
> > From: [EMAIL 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 wit
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
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
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 !
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. 75511367523
That really help me:
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+files
miklos
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Hi!
My question is about callerid i hope somebody can help.
In my snom 200 the call log show:
Received Calls Date Number
9/26/2003 11:15AM yanapoll loland
9/26/2003 10:50AM prcheyne
9/26/2003 10:49AM Administrator
9/26/2003 10:45AM asterisk
9/26/2003 10:32AM Ale
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 t
Hi
I have that error messages, what does it mean?
*CLI> WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 102 (Request)
WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] f
I´m doing the same, ix66 > asterisk.
Did you registered asterisk in the ix66?
Please share your set up, i´m with some truble using ICH .
Miklos
- Original Message -
From: "Dave Cotton" <[EMAIL PROTECTED]>
To: "Asterisk List" <[EMAIL PROTECTED]>
Sent: Thursday, September 25, 2003 2:03 PM
Hi!
There is my sip.conf:
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
maxexpirey=180 ; Max length of incoming registration we allow
defaultexpirey=160 ; Default length of i
Hi All
Asterisk is registered with ICH with no problems, but i can´t make a call,
somebody can tell me if that messages from cli are correct or there is any
problem?
Executing Dial("SIP/33-4a71", "SIP/[EMAIL PROTECTED]") in
new stack
-- Called [EMAIL PROTECTED]
*CLI>
== Spawn extension (fr
Plese somebody knows what is this message :
*CLI> WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno 102 (Request)
It is happening all time
miklos
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[EMAI
Hi
I have an ix66 from intertex and use it with asterisk..it have a dyndns
custom domain registered and resolving.
My question is about setting up a domain for asterisk, how can i do it, i
can´t find info about. I have to install a dns server in my machine runing
redhat 8?
If someone have an ix6
Thanks Gavin!
It works now.
Miklos
- Original Message -
From: "Adams, Gavin" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 23, 2003 9:51 AM
Subject: RE: [Asterisk-Users] error message playing .mp3
> -Original Message-----
>
Hi All
Somebody knows why asterisk gives me that error wile playing .mp3 files?
The files play well but the message aperas any way:
*CLI> -- Starting simple switch on 'Zap/1-1'
NOTICE[131089]: File chan_zap.c, Line 4277 (ss_thread): Got event 2
(Ring/Answered)...
-- Executing Wait("Zap/1
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