[Asterisk-Users] CISCO 7960G FIRMWARE
By the way: To be legal you also need to buy a SIP license (150 US$ list price). Cisco spare components are sold without any license or software... It would had been better you had bought a phone with a software license. Regards Sascha -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von xfastjackx Gesendet: Mittwoch, 14. Juli 2004 20:26 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] CISCO 7960G FIRMWARE Hi everybody, I will receive my CISCO 7960G tomorrow. I've ordered it as a global spare without any callmanager licence. Now I don't know if I can get firmware-updates so could please someone send me the SIP-firmware? Is the default firmware the skinny one? Wich would be better to use with asterisk? Thank you very much ___ 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 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
AW: [Asterisk-Users] CAPI problems when loading chan_capi.so
Hi Martin I use SuSE 9.0 Pro. I don't see any capi.conf - the only similar thing is /etc/capisuite/capisuite.conf but I don't know if we're talking about the same file... The module is loaded at system boot: --- pbx:~ # dmesg | grep -i capi capifs: Rev 1.1.4.1 CAPI-driver Rev 1.1.4.1: loaded capi20: started up with major 68 kcapi: capi20 attached capi20: Rev 1.1.4.2: started up with major 68 (middleware+capifs) --- I hope it's the right one... Fine. But this is only the general CAPI driver. You also need the hardware specific driver (which know how to talk to your isdn card). The system has an Eicon Diva Server BRI 2M... and by now I can't find an specific module... Unfortunately I don't have a Diva Card so I can´t tell you the module name. There are two drivers out there. The original one by Eicon is close source and nailed to a specific kernel (as I think). You are looking for the driver by www.melware.de (I think it somehow sponsored by eicon). They may not be provided by SuSE (check the package list maybe for eicon or so). So it´s possible that you have to get your hands dirty :-) Grüße Sascha --- Sascha Knific K Systems Design Tel. +49-8151-773260Wittelsbacherstr. 6a Fax. +49-8151-77326282319 Starnberg, Germany [EMAIL PROTECTED] http://www.k-sysdes.net ___ 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
AW: [Asterisk-Users] CAPI problems when loading chan_capi.so
Hi Martin [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif: ast_capi_pvt(91xx,*,pstn,0x2,2) (1,2,64) (0)(0.80/0.80) 0 Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif: ast_capi_pvt(91xx,*,pstn,0x2,2) (1,2,64) (0)(0.80/0.80) 0 Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2675 load_module: CAPI not installed! What is the output of capiinfo? Sascha --- Sascha Knific K Systems Design Tel. +49-8151-773260Wittelsbacherstr. 6a Fax. +49-8151-77326282319 Starnberg, Germany [EMAIL PROTECTED] http://www.k-sysdes.net ___ 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
AW: [Asterisk-Users] CAPI problems when loading chan_capi.so
Hi capiinfo gives: --- capi not installed - No such device or address (6) --- It´s not just about installing the apropriate package but you have to load the capi kernel module for your isdn card. The module to load on boot time is set in /etc/isdn/capi.conf (on Debian). You have to check how it´s done on your distro (I presume RedHat or SuSE). You can load the module manually. For a AVM Fritz!Card PCI you would do: modprobe fcpci Sascha --- Sascha Knific K Systems Design Tel. +49-8151-773260Wittelsbacherstr. 6a Fax. +49-8151-77326282319 Starnberg, Germany [EMAIL PROTECTED] http://www.k-sysdes.net ___ 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
AW: [Asterisk-Users] XML Phone book software.
Check: http://www.jaredsmith.net/misc/cisco7960/Directory-0.1.tgz Regards Sascha --- Sascha Knific K Systems Design Tel. +49-8151-773260Wittelsbacherstr. 6a Fax. +49-8151-77326282319 Starnberg, Germany [EMAIL PROTECTED] http://www.k-sysdes.net -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Brian R. Swan Gesendet: Donnerstag, 11. März 2004 23:07 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] XML Phone book software. Hi gang, I'm looking into writing a some phone book XML/PHP software for my Cisco phones. Specifically, I'd like to be able to use a web interface (on the computer) to maintain a contact list, and then dial from it on the phone. Maybe using MySql on the back end or something (to be determined). Before I start, and duplicate something else that exists, I wanted to see if anyone has heard of software like that? Searches of Sourceforge, Freshmeat, and Google didn't turn up much or anything. Thanks! Brian ___ 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 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] AW: [Asterisk-Users] Dial out on Capi with more MSN´s
Hi Jan
I am trying to make a outgoing call that displayes the correct Calling
number.
I have a BRI connection with 12 MSN´s.
The numbers are like this:
33445566
33445567
33445568
33445569
were the 4 last digit is the extension number.
I have the following in extensions.conf;
;ISDN2 access for outgoing call
exten = _0XXX.,1,StripMSD,1
exten = _XXX.,2,Dial,CAPI/*:bBYEXTENSION
Change the last 2 lines to:
Exten = _0XXX.,1,Dial(CAPI/33445566:b${EXTEN:1})
- ${EXTEN:1} passes the dialled extension minus the first digit
- CAPI/33445566 send the outgoing call with your MSN 33445566
Regards
Sascha
---
Sascha Knific K Systems Design
Tel. +49-8151-773260Wittelsbacherstr. 6a
Fax. +49-8151-77326282319 Starnberg, Germany
Leo +49-8151-773261WGS84: N57°59,875' E011°20,568'
[EMAIL PROTECTED] http://www.k-sysdes.net
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] Im Auftrag von Jan Larsen
Gesendet: Montag, 1. März 2004 12:28
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] Dial out on Capi with more MSN´s
Hi
I am trying to make a outgoing call that displayes the correct Calling
number.
I have a BRI connection with 12 MSN´s.
The numbers are like this:
33445566
33445567
33445568
33445569
were the 4 last digit is the extension number.
I have the following in extensions.conf;
;ISDN2 access for outgoing call
exten = _0XXX.,1,StripMSD,1
exten = _XXX.,2,Dial,CAPI/*:bBYEXTENSION
in CAPI.conf I have:
[general]
;nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=* ; Dial out via this
incomingmsn=* ; Incomming
controller=1
softdtmf=1
;accountcode=
context=incomming-ISDN2
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2
This causes all extension to dial out with the same number CLI, how do
make
it possible to call out with
individual numbers, CLI
Regards
Jan Larsen
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AW: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
Hi Birk
I´m messing arround for the last 2 day with sipgate.de. My latest
configuration seems to work only when X-lite is running on a PC on my
lan (!!!) and tried to play a call. So I think that there must be some
authentification problem or so...
When x-lite in not running I also get: 403 Forbidden ...
sip.conf
...
register = ACCOUNT-NO:SIP-PASSWORD@sipgate.de
[peer-sipgate]
type=peer
username=ACCOUNT-NO
secret=SIP-PASSWORD
fromuser=ACCOUNT-NO
fromdomain=sipgate.de
host=sipgate.de
context=from-sipgate
...
extension.conf:
---
...
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
[from-sipgate]
calls from sipgate arrive here
exten = s,1,...
...
---
Sascha
---
Sascha Knific K Systems Design
Tel. +49-8151-773260Wittelsbacherstr. 6a
Fax. +49-8151-77326282319 Starnberg, Germany
Leo +49-8151-773261WGS84: N57°59,875' E011°20,568'
[EMAIL PROTECTED] http://www.k-sysdes.net
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] Im Auftrag von Birk Bremer
Gesendet: Freitag, 27. Februar 2004 17:47
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello everybody,
has anybody managed to call a (old fashioned) phone using Sipgate.de
and
asterisk? (yes I have money on my account :-) )
The configuration I got from the sipgate.de people is at the botton of
the mail
Here is mine:
sip.conf:
register = 800:[EMAIL PROTECTED]/02115800
[sipgate]
type=friend
username=800
secret=SECRET
host=sipgate.de
fromuser=800
fromdomain=sipgate.net
nat=no
;dtmfband=3Dinband
context=sipin
canreinvite=no
extension.conf:
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
To be called on my sipgate number - no problem
If I want to call somebody I get the following error:
When I call a number directly out of the softphone:
Executing Dial([EMAIL PROTECTED]/2, SIP/[EMAIL PROTECTED]|30|tr)
in new stack
~-- Called [EMAIL PROTECTED]
~-- Got SIP response 403 Forbidden back from 217.10.79.9
~ == No one is available to answer at this time
~-- Hungup '[EMAIL PROTECTED]/2
when I use the webinterface at sipgate.de I get a ring at my
softphone,
when I pick the call I get the message (in the appearing box)
Teilnehmer nicht gefunden - User/Number not found
sometimes (while tried different config. I also got (at * console) to
many hops...
Has anybody managed this - can you please send me your configuration
(sip, extensions) or can anybody help
Thanks in advance
Birk Bremer
The configuration the sipgate people suggest:
~ register = 800:[EMAIL PROTECTED]/800
^ can't be correct
|
|
|
| [sipgate]
|
| type=friend
|
| username=800
|
| secret=sipgatepasswort
|
| host=sipgate.de
|
| fromuser=800
|
| fromdomain=sipgate.net
|
| nat=yes
|
| ;dtmfband=inband
|
| context=incomingsipgate
|
| canreinvite=no
|
|
|
| Aus der extensions.conf :
|
|
|
| [incomingsipgate]
|
| exten = h,1,Hangup
|
| exten = 800,1,Dial(SIP/internestelefon,20,tr)
|
|
|
| [sipgate]
|
| exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
|
| exten = _9.,2,Playback(invalid)
|
| exten = _9.,3,Hangup
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
iD8DBQFAP3R87QhrwFQeHVsRAjx+AJ9SvPdV4YY5iSZflo9XX/Xi97YM3wCghniD
5HUMSd5i2HUik75eajuJtpU=
=01sy
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AW: [Asterisk-Users] Loading module chan_capi.so failed!
Hi Bodo You have to load res_parking.so before chan_capi.so in you modules.conf - this is new for version 0.3.1. Sascha --- Sascha Knific K Systems Design Tel. +49-8151-773260Wittelsbacherstr. 6a Fax. +49-8151-77326282319 Starnberg, Germany Leo +49-8151-773261WGS84: N57°59,875' E011°20,568' [EMAIL PROTECTED] http://www.k-sysdes.net -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Bodo Hahnke Gesendet: Mittwoch, 11. Februar 2004 05:08 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] Loading module chan_capi.so failed! Hi Everyone, I just having my first expierence with Asterisk and after solving the first little problem now I am stuck a little. Perhaps anyone can help. Running Debian/Woody w/ 2.4.18 kernel ... think I have installed all necessary packages for running Asterisk. ... == Parsing '/etc/asterisk/modules.conf': Found [chan_capi.so] Feb 11 03:35:57 WARNING[1024]: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group Feb 11 03:35:57 WARNING[1024]: loader.c:358 load_modules: Loading module chan_capi.so failed! The chan_capi.so failed to load :( really tried to find the problem, what does the ast_get_group undefined symbol mean`? I would be very happy if anyone could help or give me a hint ... after reading some documentation about asterisk and installing the FritzCard driver I think that the problem really has something to do with the chan_capi.so ... but there is not very much documentation about it around, so please help ;)) thanks, Bodo ___ 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 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
AW: [Asterisk-Users] Eicon Diva Server
Hi Sergio, I don´t have any setup like you but looking over you config I saw this: My capi.conf is the next: [global] mode=immediate isdnmode=multipoint txgain=0.5 rxgain=0.5 [interfaces] msn=951014943 incomingmsn=951014943 controller=1 context=default echocancel=1 echotail=64 devices=2 msn=951014944 incoming=951014944 controller=1 ^^^ Maybe you should try controller=2 here. context=default echocancel=1 echotail=64 devices=2 Tell me if it helps. Sascha --- Sascha Knific K Systems Design Tel. +49-8151-773260Wittelsbacherstr. 6a Fax. +49-8151-77326282319 Starnberg, Germany Leo +49-8151-773261WGS84: N57°59,875' E011°20,568' [EMAIL PROTECTED] http://www.k-sysdes.net ___ 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
AW: [Asterisk-Users] Bluetooth discussions
Hi I don´t know if this is CTP compatible but it uses Bluetooth: http://www.olympia-it.de/cdp.htm (Sorry german only) Sascha -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] Im Auftrag von Senad Jordanovic Gesendet: Samstag, 24. Januar 2004 18:21 An: [EMAIL PROTECTED] Betreff: RE: [Asterisk-Users] Bluetooth discussions Linus Surguy wrote: IRC channel chatter says that there are some new developments with a cool presence trick that Mark has come up with for bluetooth devices. I know a bit about it, but I think the general population here would like to see some details if they're available. I don't know if this is what you are talking about, but I know of other experiments where a bluetooth enabled server allows bluetooth enabled mobile (cellular) phones to register and then carry two way calls over bluetooth rather than GSM. This would be a cool trick if Asterisk could do this too... Linus I have heard of this as well... But apparently a mobile using This service needs to be CTP (Cordless Telephony Profile) compatible. At the moment (at last to my knowledge) there is no such mobile phone available, although Apparently Samsung is to release one soon. Ta SJ ___ 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
AW: AW: [Asterisk-Users] chan_capi: suppress calling number on outbound dialing?
Sascha Knific wrote: I never had the time to try out CLIR. Now I did and it doesn´t work for me as well. Make sure you have CLIR enabled by your telekom provider (Fallweise Unterdrückung der Rufnummer). It was not enabled on my MSNs, so @ didn't work. Now my provider has enabled CLIR and everything is working as kapejod has said it would. :) I called the telecom provider (T-Com). They told me that my number is set to be always suppressed as I refused to be listed in the telephone directory. The funny thing is that nevertheless my number got always passed by default to the called party no matter what phone or pbx I used... I asked them to change it. Let´s see what happens... Sascha --- Sascha Knific K Systems Design Tel. +49-8151-773260Wittelsbacherstr. 6a Fax. +49-8151-77326282319 Starnberg, Germany Leo +49-8151-773261WGS84: N57°59,875' E011°20,568' [EMAIL PROTECTED] http://www.k-sysdes.net ___ 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
AW: [Asterisk-Users] Phone Unprovisioned Message in IP 7940 ?
Hi Tony The configuration looks fine to me. Did you check the log of your tftp server? Do the phone config files get loaded correctly? Do check also the Settings/Status/Status Messages of your phone for any errors. Sascha --- Sascha Knific K Systems Design Tel. +49-8151-773260 Wittelsbacherstr. 6a Fax. +49-8151-773262 82319 Starnberg, Germany Leo +49-8151-773261 WGS84: N57°59,875' E011°20,568' [EMAIL PROTECTED] http://www.k-sysdes.net -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von tony banks Gesendet: Montag, 8. Dezember 2003 03:21 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] Phone Unprovisioned Message in IP 7940 ? Hello all, I am newbie to Telephony world (IP and PSTN). Please excuse me if you find my questions very dumb. I am trying to configure my IP 7940 with the Asterisk, when phone boots up it only shows the message Phone Unprovisioned on the LCD panel. Under Settings--SIP Configuration--Line 1 Settings I noticed that Proxy Address is set the UNPROVISIONED, I am not sure why it is showing that though I did set proxy1_address: `129.82.44.223 in SIPDefault.conf, which is my Astersik server. Following SIP image is installed on the IP 7940. Application Load ID POS30203 My sip.conf has following lines added for the the Phone [810] type=friend secret=pass host=dynamic callerid=JOSE 810 defaultip=129.82.44.205 In my SIPmac.conf file I have made following entries # Line 1 appearance line1_name: 810 # Line 1 Registration Authentication line1_authname: 810 # Line 1 Registration Password line1_password: pass Do you see any problem here, Please let me know if I should give any more information. Regards Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
