RE: [Asterisk-Users] how to configure E400P card?
Hi! I'm not familiar with this card, but it seems you are not using the correct syntax when using command Dial or your * box isn't recognizing the channel "Zap". Did you compile and install zaptel? _How_? Is Asterisk loading "chan_zap.so"?. Please, more info! BR Eduardo -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: jueves, 28 de julio de 2005 9:12 Para: asterisk-users Asunto: [Asterisk-Users] how to configure E400P card? asterisk-users This card did not connect with E1 line? how to configure this card? When calling ,showing error: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' thanks a lot :) dev2002 [EMAIL PROTECTED] 2005-07-28 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe + CONSOLE
Hi all, Can anyone help me to make my soundcard (CONSOLE) to participate in a meetme room automatically from my dialplan. I want the soundcard to join a meetme room when someone else joins the room. Thanks a lot! == Eduardo J. López Martínez <[EMAIL PROTECTED]> Isabel Operation Center <[EMAIL PROTECTED]> DIT - Dept. Ing. Sist. Telemáticos Tlf: +34 91 3367366 (3036) UPM - Univ. Politecnica de Madrid Fax: +34 91 3367333 ETSI Telecomunicacion 28040 Madrid, Spain == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems installing TDM22B
Hi list, I am trying to install a TDM22B in my * box. I am using a Pentium III with SuSE 9.2 (2.6.8-24-default) and * 1.0.7. I installed the TDM22B (no IRQ conflicts), recompiled zaptel, libpri and asterisk. I modified a file (can’t remember which one) to create my devices entrien in /dev/. I also tryied ‘make config’ but modules don’t load at startup. But this is not my problem; I ‘modprobe zaptel’ and ‘modprobe wcfxs’ without errors. My ‘zaptel.conf’ is the following: ## CONFIGURACION TDM22B loadzone=es defaultzone=es fxoks=1,2 fxsks=3,4 My ‘zapata.conf’ is the following: [channels] context=incoming-pstn signalling=fxs_ks hidecallerid=no callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=15.0 txgain=0.0 immediate=no callerid="BrianHomePhone"<(XXX) XXX-> channel => 1 callerid="NotYetConnected2"<(222) 222-> channel => 2 callerid="NotYetConnected3"<(333) 333-> channel => 3 callerid="NotYetConnected4"<(444) 444-> channel => 4 And when i try to start asterisk i have: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found May 24 12:32:56 ERROR[4000]: chan_zap.c:6215 mkintf: Unable to get parameters May 24 12:32:56 ERROR[4000]: chan_zap.c:9155 setup_zap: Unable to register channel '1' May 24 12:32:56 WARNING[4000]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' May 24 12:32:56 WARNING[4000]: loader.c:440 load_modules: Loading module chan_zap.so failed! abeja:~ # Can you help me? Can you send a working Zapata.conf? Thanks a lot! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gateway service under Asterisk
Hello list! I am new in * but i want to learn about its possibilities. I want somebody to tell me if what I want to do is possible with *. I have a teleconference tool which uses SIP and now I am using Asterisk as POTS gateway. When I dial certain number from a telephone I connect with asterisk which asks me for an extension. When I dial certain extension I connect with my SIP application successfully and I'm able to participate as an "audio-only" participant. What I want to do now is to include more than one teleconference room. When I connect to asterisk from a phone I want * to ask me for the room I want to connect to and for a password which should be read from a database and will be different for each room. Depending on the selected room, * should dial one sip address or another, which are read from a database as well. Please, note that I am not talking about "Meetme rooms" (although I don't know if I can archive my goal using it). I only want to dial a new SIP agent depending the selected room. How can achieve this? What additional tools will be necessary? Thanks a lot! Eduardo. ============ Eduardo López Martínez <[EMAIL PROTECTED]> Isabel Operation Center <[EMAIL PROTECTED]> DIT - Dept. Ing. Sist. Telemáticos Tf: +34 913367366 (446) UPM - Univ. Politecnica de Madrid Fax: +34 913367333 Madrid SPAIN ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi compile problem
Hi all, I’m using * with a Suse 8.1 (kernel 2.4.19-4GB). My hardware are a Eicon Diva 2.01 PCI and a Eicon Diva Server 4BRI; I have installed the drivers wich came with the distribution using the distro installer (yast2). I’m experiencing problems with outgoing calls so I’ll try with chan_capi instead of chan_modem. The point is that I’m having problems with the compilation: taiwan:/usr/src/chan_capi-0.3.5 # make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c cc1: warning: changing search order for system directory "/usr/include" cc1: warning: as it has already been specified as a non-system directory chan_capi.c: In function `capi_new': chan_capi.c:1073: structure has no member named `callerid' chan_capi.c:1074: structure has no member named `dnid' chan_capi.c: In function `pipe_msg': chan_capi.c:1724: structure has no member named `dnid' chan_capi.c:1724: structure has no member named `dnid' chan_capi.c:1724: structure has no member named `dnid' chan_capi.c:1724: structure has no member named `dnid' chan_capi.c:1724: structure has no member named `dnid' chan_capi.c:1724: structure has no member named `dnid' chan_capi.c:1724: structure has no member named `dnid' chan_capi.c:1724: structure has no member named `dnid' chan_capi.c: In function `load_module': chan_capi.c:2793: warning: passing arg 4 of `ast_channel_register' from incompatible pointer type make: *** [chan_capi.o] Error 1 taiwan:/usr/src/chan_capi-0.3.5 # Can anybody help me? Thanks in advance! Eduardo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy message on ISDN cards?
Hi all,
I'm experiencing some problems with i4l and i can't find a solution. I'm
using
Eicon Diva 1 BRI
Eicon Diva Server 4 Bri
A ISDN PBX where I connect the first ISDN card (exten 204 in the
ISDN PBX) and a ISDN phone (exten 210 in the ISDN PBX)
Suse 8.1 with ISDN4Linux
Asterisk 1.0
I'm trying to make phone calls from a softphone (Windows Messenger 4.6) to
the ISDN phone but I get this:
Asterisk Ready.
*CLI> -- Executing Dial("SIP/edu-ee6e", "Modem/g1/204:210") in new stack
Urgent handler
Jan 1 18:51:19 WARNING[29364]: chan_modem_i4l.c:601 i4l_dial: Outgoing MSN
edu not allowed (see outgoingmsn=,*, in modem.conf)
-- Called g1/204:210
Urgent handler
-- Modem[i4l]/ttyI0 is busy
Urgent handler
-- Hungup 'Modem[i4l]/ttyI0'
Urgent handler
Urgent handler
== Everyone is busy/congested at this time
== Auto fallthrough, channel 'SIP/edu-ee6e' status is 'CHANUNAVAIL'
Urgent handler
I see how the modem activates using imon (ISDN monitor) and tries to
connect.
When I add "outgoingmsn=edu" in modem.conf the modem does not respond, and I
get the following:
Asterisk Ready.
*CLI> -- Executing Dial("SIP/edu-c490", "Modem/g1/204:210") in new stack
Urgent handler
Urgent handler
-- Called g1/204:210
(...long time passed...)
Urgent handler
-- Hungup 'Modem[i4l]/ttyI0'
Urgent handler
Urgent handler
Jan 1 18:58:55 WARNING[29394]: chan_sip.c:694 retrans_pkt: Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno 1
(Critical Response)
Calls from the ISDN phone to a softphone WORK.
Can anyone help me??? Thanks in advance ;)
Eduardo.
My configuration is the following:
**
modem.conf
**
[interfaces]
context=INCOMING_ISDN
driver=i4l
language=en
type=autodetect
dtmfmode=i4l
dialtype=tone
mode=immediate
group=1
msn=204
incomingmsn=*,0,210
outgoingmsn=*
device=>/dev/ttyI0
;device=>/dev/ttyI1
;device=>/dev/ttyI2
;device=>/dev/ttyI3
;device=>/dev/ttyI4
;device=>/dev/ttyI5
;device=>/dev/ttyI6
;device=>/dev/ttyI7
***
extensions.conf
***
[INCOMING-SIP]
exten=>6000,1,Dial,SIP/edu
exten=>7000,1,Dial,SIP/edu2
exten=>4000,1,Dial,Modem/g1/204:210
[INCOMING_ISDN]
exten=>s,1,Wait(1)
exten=>s,2,Answer
exten=>s,3,Dial,SIP/edu
*
sip.conf
*
[edu]
type=friend
username=edu
host=dynamic
diallow=all
allow=gsm
;allow=ulaw
allow=alaw
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[Asterisk-Users] ISDN outgoing calls problem
Hello all,
I'm trying to make phone calls from a softphone through an ISDN line. The
problem I have is that when I try to make a call (outgoing) my ISDN card
does not respond.
The point is that i am being able to make phone calls from an ISDN phone
connected to a ISDN-PBX (the same ISDN-PBX where I connect my
Asterisk-computer).
My ISDN Hardware
DIVA Server 4Bri
My OS
---
Suse 8.1
My config
--
**MODEM.CONF**
[interfaces]
context=INCOMING_ISDN
driver=i4l
language=es
type=autodetect
dialtype=tone
mode=immediate
group=1
msn=204
incomingmsn=*,0
outgoingmsn=0,edu,*,f10
device=>/dev/ttyI0
;device=>/dev/ttyI1
;device=>/dev/ttyI2
;device=>/dev/ttyI3
;device=>/dev/ttyI4
;device=>/dev/ttyI5
;device=>/dev/ttyI6
;device=>/dev/ttyI7
**SIP.CONF**
..
context=INCOMING-SIP; Default context for incoming calls
..
[edu]
type=friend
username=edu
host=dynamic
diallow=all
allow=gsm
;allow=ulaw
allow=alaw
[edu2]
type=friend
username=edu2
host=dynamic
diallow=all
allow=gsm
allow=ulaw
allow=alaw
**EXTENSION.CONF**
.
[INCOMING-SIP]
exten=>6000,1,Dial,SIP/edu
exten=>7000,1,Dial,SIP/edu2
exten=>4000,1,Dial,Modem/ttyI0:210
[INCOMING_ISDN]
exten=>s,1,Wait(1)
exten=>s,2,Answer
exten=>s,3,Dial,SIP/edu
My ISDN phone extension is 210 and my ISDN card extension is 204 (in my
ISDN-PBX).
I'm using Windows Messenger 4.6 as SIP softphone.
What I get when I try to make a phone call from my messenger to the ISDN
phone is:
Asterisk Ready.
*CLI> -- Registered SIP 'edu' at 138.4.24.61 port 14165 expires 120
-- Saved useragent "RTC/1.2.4949" for peer edu
-- Executing Ringing("SIP/edu-7d39", "Modem/ttyI0:210") in new stack
-- Executing Dial("SIP/edu-7d39", "Modem/ttyI0:210") in new stack
-- Called ttyI0:210
But the phone does not ring.
Any help, please?
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RE: [Asterisk-Users] Busy message on ISDN cards?
Hello,
You can fix it adding in your "modem.conf":
outgoingmsn=*
I'm not sure is you have to write a "*" or a "0" or simply "andrew".
Then, if you make it work, tell me if your outgoing ISDN calls are working.
I'm having problems whit them.
B.R.
Eduardo.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Hatzis,
Michael
Enviado el: martes, 14 de diciembre de 2004 23:21
Para: Andrew Furey; Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: RE: [Asterisk-Users] Busy message on ISDN cards?
I had the same problem even though it was with capi, this may help. Have
you set your msn as Andrew or your line number??
Try this
exten => 2468,1,Dial(${TRUNK}/91234567:0412345678:1)
Regards
Michael Hatzis
0421 476 211
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Furey
Sent: Tuesday, 14 December 2004 4:43 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Busy message on ISDN cards?
Hi all,
I'm new to asterisk and not too knowledgeable on ISDN, so please be
gentle :)
I have a dual-channel Eicon Diehl Diva card in a Debian Woody box with
kernel 2.4.27, connecting to a Telstra (Australia) Onramp Home Highway
ISDN line. I'm pretty certain the card and line both work since
they've been used in this machine for PPP before this (but with an
older kernel with DoV patches, which are no longer to be used).
If I do
# modprobe hisax type=11,11 protocol=2,2 id="HiSax"
it responds (in the syslog) with:
kernel: ISDN subsystem Rev:
1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1 loaded
kernel: HiSax: Linux Driver for passive ISDN cards
kernel: HiSax: Version 3.5 (module)
kernel: HiSax: Layer1 Revision 1.1.4.1
kernel: HiSax: Layer2 Revision 1.1.4.1
kernel: HiSax: TeiMgr Revision 1.1.4.1
kernel: HiSax: Layer3 Revision 1.1.4.1
kernel: HiSax: LinkLayer Revision 1.1.4.1
kernel: HiSax: Total 2 cards defined
kernel: HiSax: Card 1 Protocol EDSS1 Id=HiSax (0)
kernel: HiSax: Eicon.Diehl Diva driver Rev. 1.1.4.2
kernel: PCI: Found IRQ 9 for device 00:09.0
kernel: PCI: Sharing IRQ 9 with 00:04.2
kernel: Diva: IPAC PCI card configured at 0xd0862000 IRQ 9
kernel: Diva: IPAC PCI space at 0xd086
kernel: Diva: IPAC version 1
kernel: Eicon.Diehl Diva: IRQ 9 count 1697
kernel: Eicon.Diehl Diva: IRQ 9 count 1705
kernel: HiSax: DSS1 Rev. 1.1.4.1
kernel: HiSax: 2 channels added
kernel: HiSax: MAX_WAITING_CALLS added
so it appears to be detected. I'm using the following modem.conf:
[interfaces]
context=remote
driver=i4l
language=en
type=autodetect
dialtype=tone
mode=immediate
group=1
msn=91234567
incomingmsn=*
device => /dev/ttyI0
Starting asterisk with -c returns:
== Parsing '/etc/asterisk/modules.conf': Found
[chan_modem.so] => (Generic Voice Modem Driver)
== Parsing '/etc/asterisk/modem.conf': Found
== Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated Modem
Driver)
But if I define a test extension such as:
TRUNK=Modem/g1
exten => 2468,1,Dial(${TRUNK}/91234567:0412345678)
and try to dial it, the console says:
Dec 14 13:29:17 WARNING[15375]: chan_modem_i4l.c:608 i4l_dial:
Outgoing MSN andrew not allowed (see outgoingmsn=,, in modem.conf)
-- Called g1/91234567:0412345678
-- Modem[i4l]/ttyI0 is busy
-- Hungup 'Modem[i4l]/ttyI0'
I gather than "busy" is used for pretty much everything except for no
connection, but are there any suggestions of where to look?
Thanks in advance,
Andrew
--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
-- Bill Garrett
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[Asterisk-Users] Analog modem testing
Hello all, I’m new in all this and i need your help. I have some legacy 56k and 33.6K modems and I want to test them to work with asterisk before purchasing any new hardware. Can anyone provide me instructions to test them. My hardware is: - Pentium 500 MHz with Suse 8.1 - ISDN BRI 1 port. (I have no additional hardware or an ISDN line to test it) - Several modems. - A conventional telephon line. - A lan connection. The only thing know how to configure (more or less) is the SIP agent. I’ve been able to connect two “windows messengers” using Asterisk by adding two blocks in “sip.conf” and two lines in “extensions.conf” (I read how to do it somewhere J). The idea is to make calls from a softphone to the PSTN and viceversa. Thanks a lot! Eduardo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can i test a modem with Asterisk?
Hello all, I'm a newbee and i'd like to test some analog modems i have before purchasing any new hardware. I'm using: - Pentium 500 MHz - Several 56K and 33.6K analog modems (internal and external). - ISDN BRI 1 port card. - A LAN i've successfully tested some SIP sofphones. - An ordinary telephon line (and an ordinary phone :) ). I only modified extensions.conf in order to test two softphones but i have no idea of configuring Arterisk to test a modem (special drivers, signaling configuration, complex dialplan configuration). Can anybody provide me a simple set of config files to thest an analog modem? Thanks a lot!!! Eduardo Lopez PD.- I have no aditional hardware to thes the ISDN card. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
