Hi all
I hope that someone of you can solve this. Right now I'm stuck!
I'm using asterisk with some SIP extensions. Basically I want to
establish a call between desktop voip phone (ext 181) and embedded sip
system (ext 182)
All I can see in CLI is:
== Using SIP RTP CoS mark 5
--
My question is so complex and I try to explain well.
We have a customer that he wants limits incoming calls to his extensions
to only one. That's not complicated with GROUPCOUNT, DEVICE_STATE or
SIPPEER with curcalls option.But the problem is when you want implement
CCBS service.
If we have
A good solution is use a program that use sipsack for SIP, something like
sipsack for IAX and Linux-HA for asterisk. In this way you check if SIP or
IAX is OK, and if these technologies are bad, you can kill asterisk and
linux-HA will do the rest. In PSTN Field, you can check rxhooksig in struct
http://www.junghanns.net/en/ISDNguard_produkt.html
srsergio
-Mensaje original-
De: John Daragon [mailto:[EMAIL PROTECTED]
Enviado el: jueves, 20 de octubre de 2005 17:24
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] E1/T1 failover hardware
Your card must be a TDM with 4 FXS ports. FXO port is to connect and analog
line, and FXS port is for connect analog phone. Are you sure that in 3rd and
4th ports you have immediate=no?
regards,
srsergio
-Mensaje original-
De: Alex Ongena [mailto:[EMAIL PROTECTED]
Enviado el: jueves,
Discussion
Asunto: RE: [Asterisk-Users] TDM400P off-hook detection problem
On Thu, 2005-10-13 at 12:44 +0200, Sergio Serrano wrote:
Your card must be a TDM with 4 FXS ports. FXO port is to connect and
analog line, and FXS port is for connect analog phone. Are you sure
that in 3rd and 4th ports
You can't put four span in timing, because only one must be like nmaster
sincronization. If one of your telco provide time for your card. Put second
value in all span to 0.
regards,
srsergio
-Mensaje original-
De: Ronald Hartmann [mailto:[EMAIL PROTECTED]
Enviado el: martes, 04 de
Hi
Srs.,
Do you know if it's possible make a videocall from asterisk to UMTS
mobile phone?. Both technologies use H.263 like videocodec.
Any
idea?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
www.inconcertCC.com has a solution based on Asterisk.
regards,
srsergio
-Mensaje original-
De: Bartosz Jozwiak [mailto:[EMAIL PROTECTED]
Enviado el: jueves, 29 de septiembre de 2005 17:17
Para: Asterisk Users Mailing List - Non-Commercial Discussion
CC: Commercial and
You must install libncurses5-dev
regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Enviado el: martes, 27 de septiembre de 2005 9:20
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: [Asterisk-Users] Termcap missing (compile
Hi all,
we have same problem with a x346. Mainly, TE410P shares IRQ with
network card and if you change IRQ for this slot, automatically change IRQ
in network card.
Any idea?
srsergio
-Mensaje original-
De: George Pajari [mailto:[EMAIL PROTECTED]
Enviado el: lunes, 26 de
show
application Queue is your friend.
De: Sander [mailto:[EMAIL PROTECTED]
Enviado el: viernes, 23 de septiembre de 2005 13:11Para:
'Asterisk Users Mailing List - Non-Commercial Discussion'Asunto:
[Asterisk-Users] Queues
Hi there i need to
know if there is a wayto play a ringing sound
Try in www.asterisk-es.org
-Mensaje original-
De: Sebastian Milioto [mailto:[EMAIL PROTECTED]
Enviado el: lunes, 19 de septiembre de 2005 15:08
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] Asterisk in Spanish
Hi all,
I've been installing
Please,
send us zaptel.conf and zapata.conf and say us what card you
have(TE110P, TE410P...). And what is your country.
Regards,
srsergio
-Mensaje original-
De: manish kumar [mailto:[EMAIL PROTECTED]
Enviado el: lunes, 19 de septiembre de 2005 6:32
Para:
Hi,
Try SetCIDNum application before VoiceMail application
regards,
srsergio
-Mensaje original-
De: Chad Brown [mailto:[EMAIL PROTECTED]
Enviado el: miércoles, 31 de agosto de 2005 8:48
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] Manipulate
This option is under Library routines in your kernel configuration.
Regards,
srsergio
-Mensaje original-
De: Christoph Eicke [mailto:[EMAIL PROTECTED]
Enviado el: miércoles, 31 de agosto de 2005 10:59
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re:
Hi Sr.
rrmemory i same like roundrobin, but this policy store which is the next
when a call get into your system.
For example with next queue:
SIP/1
SIP/2
SIP/3
and roundrobin, all calls stars with SIP/1 and with rrmemory first call
starts with SIP/1, second call with SIP/2 and so on.
How do you do monitoritng? How Server B knows that Servar A is down? I just
do a rsync and MySQL Replication, but I try to do a C program that monitor
Server. If you know how can I do this monitoring I will be pleasant with
you.
regards,
srsergio
-Mensaje original-
De: Senad J
If I use hearbeat I need a failover system for ISDN Lines, not? I waould
like that if Server A crashes, Server B Control SIP Registration and ISDN
Lines. Do you know about this?
regards,
srsergio
-Mensaje original-
De: Senad J [mailto:[EMAIL PROTECTED]
Enviado el: martes, 23 de agosto
Title: Mensaje
I'm sorry for my
holidays message, but I think it's too hard span me from list, don't you think?
Could admin return to list, please?
Regards,
srsergio
___
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Hasta el día 31 de Julio permaneceré de vacaciones, por lo que cualquier tipo de consulta, técnica o comercial debe redirigirla a [EMAIL PROTECTED] o a [EMAIL PROTECTED]
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Asterisk-Users@lists.digium.com
Hasta el día 31 de Julio permaneceré de vacaciones, por lo que cualquier tipo de consulta, técnica o comercial debe redirigirla a [EMAIL PROTECTED] o a [EMAIL PROTECTED]
___
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Asterisk-Users@lists.digium.com
Hasta el día 31 de Julio permaneceré de vacaciones, por lo que cualquier tipo de consulta, técnica o comercial debe redirigirla a [EMAIL PROTECTED] o a [EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Title: Mensaje
I'm sorry for the
several messages with holidays message.
Regards,
srsergio
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
Please,
send us zapata.conf. It's possible that you don't have well
configure zapata.conf, because in your trace you try to dial through g0
group and your Zap/4(I understand is your Zap connected to PSTN) must be
into the 0 group.
Regards,
srsergio
-Mensaje original-
De:
: this is a trunk. Create a ZAP trunk in AMP for Channel 4
context=from-pstn channel = 4
At 10:10 AM 6/20/2005, Sergio Serrano wrote:
Please,
send us zapata.conf. It's possible that you don't have well
configure zapata.conf, because in your trace you try to dial through g0
group and your Zap/4(I
Before change OS try to do next steps:
first, stop asterisk. Second, you must do ztcfg -s to shutdown
span. Unload modules, load modules if you need and do ztcfg -vv again.
Start asterisk
Regards
Srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En
Title: Mensaje
Hi
all,
I have to
interconnect Asterisk with a Siemens HIPATH 3750. In siemens we can configure
ECMA-QSIG Master, ISO-QSIG Master,Point to Point link withCRC4 and
Point to Point link withouthCRC4): Siemens has BNC connector.
I use a balun with BNC and RH45
connectro. I
Fantastic!! Thanks to your works
regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Areski
Enviado el: miércoles, 26 de enero de 2005 18:05
Para: Asterisk-Users Mailing-list
Asunto: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard
You can try to set one context for each extension.
regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Alen Salamun
Enviado el: martes, 25 de enero de 2005 16:04
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto:
Title: Mensaje
Have you solve your
Problem?, I have same problem after with
recompile kernel.
Regards,
srsergio
Monday, December
20, 2004, 12:44:36 PM, Matt wrote:MR Have you tried doing a modprobe
-r first?Before reboot I did
rmmod wcte11xp. If you mean that.now modprobe -r wcte11xp
Title: Mensaje
Hi all,
I have
installed a TE110P in a BOX but when I load zaptel module I can't see any device
in /proc/zaptel. And led of the card is green.
My zaptel.conf is the
next:
span=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16loadzone=esdefaultzone=es
and cat /proc/pci throguh
]
Enviado el: miércoles, 29 de diciembre de 2004 18:45
Para: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel
On Wed, Dec 29, 2004 at 11:59:58AM +0100, Sergio Serrano wrote:
Hi all,
I have installed
: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel
On Wed, Dec 29, 2004 at 09:17:45PM +0100, Sergio Serrano wrote:
Yes I'm sure that I load wcte11xp. When I do wcte11xp I obtain next
error:
/lib/modules
Try
exten= _X.,1, Dial(Zap/g2) If your 2nd TE110 has defined a group.
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Jeremy SALMON
Enviado el: miércoles, 22 de diciembre de 2004 15:01
Para: asterisk-users@lists.digium.com
Title: Mensaje
Hi all,
again I try
configure Hylafax with asterisk. I would like configure Asterisk in the next
way:
1)An incoming fax go into through X100P
2)Asterisk detects Fax and redirect fax to
Hylafax
Is it
possible?
Any idea woluld be great
idea?
regards,
srsergio
--
Please, could you send us cat /proc/pci?. Could you compile libpri?
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Ning Zhou
Enviado el: martes, 23 de noviembre de 2004 16:10
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Error when
Hi,
this call is from? Zap channel, Capi channel or other channel? It is
possible that you don't detect well hangup from incoming channel.
Regards.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Altus Snyman
Enviado el: lunes, 22 de noviembre
Hi all,
I have a PBX working for a year with an Eicon Diva Server 4BRI. One
day it was a storm and nothing occurs, but after a a few days I can't send
and receive any calls. I have connected TEIs to Asterisk and other PBX and
when I try to dial, I hear correct tone two times, but then line
Hi all,
I try to compile app_icd to test it but I can't compile it. I have
installed asterisk 1.0.2 and I download ICD and put files into
/usr/src/asterisk/apps/icd directory. I think that make.conf in icd
directory is ok but when I try to compile icd I obtain next error:
=== Compile:
Hi,
Voicemoil capabilities are in Asterisk. You can use Asterisk
voicemail from any SIP Software.
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Ashling
O'Driscoll
Enviado el: miércoles, 17 de noviembre de 2004 18:28
Para:
Hi all,
I'm sorry, but I'm stupid because I haven't load res_parking.so.
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sergio
Serrano
Enviado el: viernes, 10 de septiembre de 2004 9:35
Para: 'Asterisk Users Mailing List - Non
Hi all,
I'm trying to configure a swissvoice IP10S but after a minutes
this phones appears like UKNOWN in sip show peers and it is unaccesible.
This phone can make call but it can't receive calls.
Any idea?
Regards,
srsergio
___
Asterisk-Users
SIP version
IP10 SP v0.0.1 (Build 5)
Regards,
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Florian
Overkamp
Enviado el: lunes, 06 de septiembre de 2004 13:42
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: RE: [Asterisk-Users]
had same problem? Could anyone help me with this
problem?
Best regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sergio
Serrano
Enviado el: miércoles, 01 de septiembre de 2004 0:46
Para: 'Asterisk Users Mailing List - Non-Commercial
,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sergio
Serrano
Enviado el: miércoles, 01 de septiembre de 2004 12:51
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: RE: [Asterisk-Users] Asterisk SIP between two networks
19:16
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: RE: [Asterisk-Users] Asterisk SIP between two networks
Sergio Serrano [EMAIL PROTECTED] wrote:
SIP Provider---ADSL router---localnet
192.168.20.0---ASTERISK---localnet 172.24.240.0---softphones
first
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))
Content-Length: 0
It is a bug? Why if I put bindaddr=0.0.0.0 packet received by asterisk
is broken?
Could anyone help me?
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sergio
Serrano
Hi all,
I have next configuration:
SIP Provider---ADSL router---localnet
192.168.20.0---ASTERISK---localnet 172.24.240.0---softphones
first localnet 192.168.20.0
second localnet 172.28.240.0
in second localnet we have
Title: Mensaje
Push
send after you number,
srsergio
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de James
DuttonEnviado el: jueves, 05 de agosto de 2004 12:28Para:
[EMAIL PROTECTED]Asunto: [Asterisk-Users] Call Transfer
Problems with Grandstream
Hi,
in Spain that process is correct. If you setup a communication between
a caller and a called, if called phone hangs, in caller side hear a
silence, but is a correct process. It's is due to in the called side you
can hangup a phone and pickup other phone without lost communication.
Regards,
Title: Mensaje
It's
more easy download tarball and compile it.
srsergio
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de
YanEnviado el: jueves, 22 de julio de 2004 13:31Para:
[EMAIL PROTECTED]Asunto: [Asterisk-Users] debian
install zaptel
Hi:
Title: Mensaje
I have the same problem.I'm usinr
asterisk-1.0-RC1. Anyone could help us?
regards,
srsergio
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Dan
FernandezEnviado el: viernes, 16 de julio de 2004
20:42Para: [EMAIL PROTECTED]Asunto:
Try to compile with lastest CVS
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Martin
List-Petersen
Enviado el: jueves, 15 de julio de 2004 1:12
Para: [EMAIL PROTECTED]
CC: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] Chan_Capi 0.3.4a error
, S.L.
Sergio Serrano RevueltoRD
Manager
Avda. Juan López de Peñalver 17Edificio
Hi all, I try to do next transfer:
A person contact with me, I would like transfer to other person
in next manner. I call to other person and when I say who wants talk
with him I hangup phones an call is redirect automatically to other
person:
1. call to me
2. Hold the
: Transfer with Budgetone
Sergio Serrano wrote:
Hi all, I try to do next transfer:
A person contact with me, I would like transfer to other person
in
next manner. I call to other person and when I say who wants talk with
him I hangup phones an call is redirect automatically to other
person
I have just to talk with Grandstream and they say to me that they ar
working in 3-way conferencing for BT-100 series. I hope they have FW
soon. One question more? How can I do parking call with Budgetone.
Before # works fine, but Now it doesn't work.
-Mensaje original-
De: [EMAIL
Hi all,
Are there any way to clean codec_g729b license ffrom Asterisk. I
would like to clean a license to install other more big, but when I do
../codec_g729b/Registration --XX I obtain a
segmentation fault.
Any idea?
srsergio
Hi all,
I have just put a message from a few days with a problem with
CAPI hangup. I have noticed that line with 97% of hangs, is a line
connected with a ATA286 with a modem-fax. Could it be the problem?
Regards,
srsergio
___
Asterisk-Users
First thing you must is read next url
http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO
and if you hav done this, please attach your capi.conf.
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de massimo
Enviado el: martes, 04 de mayo de 2004
Hi all,
I have a PC working with a DIVA Eicon Server 4BRI during a lot
of time. Now I can't make call but I can receive calls.
I load diva with command: divactrl load -c 1 -f ETSI -u -t 0
Country: Spain
Isdnmode: point to point
My capi.conf is the next:
[global]
mode=immediate
?
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sergio
Serrano Revuelto
Enviado el: miércoles, 19 de mayo de 2004 12:00
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] CAPI Eicon Diva Server 4BRI
Hi all,
I have a PC working
Hi all,
I would like to know if chan_capi is prepared to receive faxes. I have a
eicon deiva server 4bri with chan_capi and Grandstream HandyTone connected
to a Fax, but this fax can't receive faxes.
Any idea?
Thanks,
srsergio
___
Asterisk-Users
Try to add a qualify= to sip.conf, and try to exec a sip show peers.
In spite of phones appears like register, if you use NAT, your firewall
can cut communication. Try the next:
Just after phone register call to it, and then wait for a minutes and
try to call again. Could you call first time
Hi all,
I try to install a G.729 license in SCSI system with a IDE CDROM
but I can't do it. Any one has experience to do this?
Regards,
srsergio
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
]
Asunto: RE: [Asterisk-Users] G.729 and SCSI
Sergio Serrano wrote:
Hi all,
I try to install a G.729 license in SCSI system with a IDE CDROM
but
I can't do it. Any one has experience to do this?
Regards,
srsergio
Here is the wiki page for g729:
http://www.voip-info.org/wiki
Hi,
I have a problem with a Eicon Diva Server 4 BRI. I have 4 BRI ISDN and
11 number for these 4 ISDN. At first I have connected one of these 4 ISDN.
When I try to call I receive the next trace:
-- Executing ChanIsAvail(SIP/716-b0cd,
We are going to do this test next week. I will say the result
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Jim
Rosenberg
Enviado el: domingo, 29 de febrero de 2004 1:15
Para: Asterisk
Asunto: [Asterisk-Users] PCphoneline FXO to FXS
If your BG 101 is in intranet, try to adjust your qualify parameter to
60.
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Matthew B
Marlowe
Enviado el: viernes, 27 de febrero de 2004 2:08
Para: [EMAIL PROTECTED]
Asunto: RE:
Title: Mensaje
Hi
all,
For a few days we have a veryextrange
problem. We have an intranet with Budgetone and others SIP Phones.
In the
extranet We HaveBudgetone Phones. The whole system was working well
between the extranet and the intranet until a few days ago.
When
we try to speak
www.avanzada7.com
Sergio Serrano RevueltoRD
Manager
Avanzada 7
Hi all,
I will ptobe your answers tomorrow. I'll say the results.
Thanks for all.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sascha
Knific
Enviado el: martes, 10 de febrero de 2004 22:08
Para: [EMAIL PROTECTED]
Asunto: AW: [Asterisk-Users]
Hi all,
anyone could help me with capi.conf?. I have installed an Eicon
Diva Server 4BRI. I have 2 EuroISDN BRI lines,
First line number: 951014943
Second line number: 951014944
I try to do 4 calls but, I can't do more than two call.
My capi.conf is the next:
You must use Monitor Application
Happy New Year,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Edoardo
Borghesi [fabbricadigitale]
Enviado el: viernes, 02 de enero de 2004 12:33
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Call recording
PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Masakazu
Nakano
Enviado el: domingo, 21 de diciembre de 2003 5:37
Para: [EMAIL PROTECTED]
CC: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] RxFAX application
Hi sergio
On Fri, 19 Dec 2003 14:49:15 +0100
Sergio Serrano Revuelto [EMAIL PROTECTED
Hi all,
I have tested RxFAX application through X100P card. When Fax
arrive i obtain the next trace:
-- Starting simple switch on 'Zap/1-1'
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing SetMusicOnHold(Zap/1-1, random) in new stack
-- Executing
Hey Srs.
I have a little problem with the next scenario:
Internal Phone(801)--Asterisk(public IP) --INTERNET--ADSL
Router--Budgetone(716)
|--ADSL Router--Budgetone(717)
My sip.conf is the next:
[general]
port = 5060 ; Port to bind to
bindaddr =
Next configuration must work:
zaptel.conf
fxoks=1-4
loadzone=fi
defaultzone=fi
Zapata.conf
[channels]
group=1
context=internt
signalling=fxo_ks
channel=1-4
Srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de JanM
Enviado el: jueves, 20 de
Title: Mensaje
try to
cvs
srsergio
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Quan Le
TrungEnviado el: jueves, 13 de noviembre de 2003
10:43Para: [EMAIL PROTECTED]CC:
[EMAIL PROTECTED];
[EMAIL PROTECTED]Asunto:
Try to load module manually: modprobe wcfxo; ztcfg -
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Cameron
Palmer
Enviado el: viernes, 31 de octubre de 2003 6:27
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] 2 X100Ps give error
I
or address (6)
cameron.
On Fri, 31 Oct 2003, Sergio Serrano Revuelto wrote:
Try to load module manually: modprobe wcfxo; ztcfg -
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Cameron
Palmer Enviado el: viernes, 31 de
I need connect up to 100 analog phone to a H.323 network through *. I
think use TE410P, But I need to know what channel bank is better. I use
E1 lines
Any idea?
Thanks in advance,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de DUSTIN
WILDES
Title: Mensaje
AVM
Fritz it good for Asterisk. A little difficult to configure but not
impossible.
srsergio
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Tomica
CrnekEnviado el: jueves, 16 de octubre de 2003 14:36Para:
[EMAIL PROTECTED]Asunto:
Hi,
I have a problem with sip.conf. After some hours my sip
phone(netergy) hangs. In clonse appears the next logs repeatly:
10 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.168.0.155 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK5ffceb80
From: asterisk sip:[EMAIL
Title: Mensaje
Yes yo
can do it.
srsergio
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de
e-smithEnviado el: miércoles, 24 de septiembre de 2003
15:02Para: [EMAIL PROTECTED]Asunto:
[Asterisk-Users] Using Asterisk in an netted
scenario
Hi,
Just
exten=XXX,1,Dial(h323/3|17|tTm)
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger
Enviado el: martes, 23 de septiembre de 2003 11:07
Para: [EMAIL PROTECTED]
Asunto: RE: [Asterisk-Users] how to dial a h323 destination ?
Please, can
Could you send me your h323.conf and you gnugk.ini?
Sergio Serrano Revuelto
Responsable de Consultoría
Avanzada 7, S.L.
Teléfono / Fax: +34 951 01 49 47 / +34 951 01 09 22
www.avanzada7.com
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger
Try to add gwprefix in oh323.conf after your alias. You must know that
you can configure * gw in gnugk.ini or in oh323.conf. I recommend you
put in your oh323.conf.
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger
Enviado el:
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Sergio
Serrano Revuelto
Gesendet: Dienstag, 23. September 2003 12:27
An: [EMAIL PROTECTED]
Betreff: RE: [Asterisk-Users] how to dial a h323 destination ?
Try to add gwprefix in oh323.conf after your alias
You can try AVM FRITZ with chan_capi from kapejod.
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de YO Internet
Information
Enviado el: lunes, 22 de septiembre de 2003 0:03
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] ISDN BRI hardware
Hi all,
when I try register my netergy SIP Phone with *, I can't do it
due to the next message:
1 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK3438300a
From: asterisk sip:[EMAIL PROTECTED];tag=as34fa433f
To:
Title: Mensaje
Hi,
I
would like to configure a stage for SIP phones. This stage would be the
next:
two
netergy SIP phones connected to Asterisk through chan_sip.
one
X100P or AVM FRITZ to outside lines.
I
think that sip.conf would be the next:
;;
SIP Configuration for
I have the same problem,
Asterisk debug is the next:
REGISTER sip:AVANZADA7 SIP/2.0
Call-ID: [EMAIL PROTECTED]
From: 704sip:[EMAIL PROTECTED];tag=230b0-e0
To: 704sip:[EMAIL PROTECTED]
CSeq: 101 REGISTER
Via: SIP/2.0/UDP 192.168.0.154:5060
Contact: sip:[EMAIL PROTECTED]:5060
Max-Forwards: 70
];tag=230b0-e0
instead of this:
From: 704sip:[EMAIL PROTECTED];tag=230b0-e0
Jan.
On 19-09 08:38, Sergio Serrano Revuelto wrote:
I have the same problem,
Asterisk debug is the next:
REGISTER sip:AVANZADA7 SIP/2.0
Call-ID: [EMAIL PROTECTED]
From: 704sip:[EMAIL PROTECTED];tag=230b0-e0
];tag=230b0-e0
instead of this:
From: 704sip:[EMAIL PROTECTED];tag=230b0-e0
Jan.
On 19-09 08:38, Sergio Serrano Revuelto wrote:
I have the same problem,
Asterisk debug is the next:
REGISTER sip:AVANZADA7 SIP/2.0
Call-ID: [EMAIL PROTECTED]
From: 704sip:[EMAIL PROTECTED];tag=230b0-e0
HI, I am probing chan_modem_i4l again with AVM FRITZ but I can hear
nothing in phone outside of asterisk, I explain
Phone 1-- AVM_FRITZ--Asterisk-- Phone 2
From Phone1 to Phone 2 I can hear, but
From phone 2 to phone 1 I can't hera nothing.
Any idea?
srsergio
-Mensaje original-
De:
Hi,
I would like to know how suppress number for outside dialling in
CDR table. For example, if I need press 9 key to make an outside call, I
would like that the number in dst field in cdr table was the outside
number without 9 key. It's possible?
Thanks in advance,
srsergio
Hi, I would like to know how do two things.
First, it is possible simulate PBX scenary?, I explain. I would like
that when an user press 9(outgoing key) asterisk will generate a new
dial tone in H.323 EP and then the user could press number for dial.
Second, it's possible modify time interdigit.
Hi, I would like to know how do two things.
First, it is possible simulate PBX scenary?, I explain. I would like
that when an user press 9(outgoing key) asterisk will generate a new
dial tone in H.323 EP and then the user could press number for dial.
Second, it's possible modify time interdigit.
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