Help in English for * on FBF, please

gcf · 11 Nov 2007

Hi all,
this is a great forum, but unfortunately, I have to read it through Google translate

I own an FBF 7140 and I am trying to setup * on it.
I have installed * from spblinux.de/fbox, and it starts, but I don't know how to configure it, so I ask for your help, or if you know a guide in english, that would be also useful.
First I'd like to ask how to avoid this error when asterisk starts:
asterisk: can't load library 'libresolv.so.0'
and if it's serious.

Then I'd like to know the basic configuration settings I have to make in order to:
a) access asterisk from FBF's analog extensions FON1 and FON2
b) register a spa3102 ata to the * as an extension.
c) add trunks to use my voip providers.
d) route my analog fixed line to the PBX.

I would be grateful for any help,
greetings from Athens, Greece.
George

BlueAndi · 12 Nov 2007

Use .../fbox.new/cfg_asterisk14 instead of .../fbox/... and follow the install instructions again.

gcf · 12 Nov 2007

Thanks, I try that and during installation I get:
asterisk14.sqf -> /var/ram0: Start extracting files ... please wait!
cp: /var/asterisk14.tmp/modules/*: No such file or directory
asterisk14 has been successfully installed to usb device MassStorageDevice-Partition-0-1
total used free shared buffers
Mem: 30328 22980 7348 0 632
Swap: 0 0 0
Total: 30328 22980 7348
to use asterisk14 type ./cfg_asterisk14 start
(or /var/media/ftp/MassStorageDevice-Partition-0-1/addons/cfg_asterisk14 start)

when I start it:
zebedee(2273/1024): Listening on local port 4570
# asterisk: can't load library 'libresolv.so.0'

Please notice I found a libresolv.so.0 file in this thread
http://www.ip-phone-forum.de/showpost.php?p=727319&postcount=78
but it seems to be 0 sized, and I don't know how to add it anyway.

gcf · 12 Nov 2007

analog line und asterisk - FBF

Dear all,
I have an ISP issued FBF. After I installed the official AVM f/w, asterisk starts fine.
I only need some help with getting the analog fixed line through asterisk.
How can I do that?

Thanks in advance

BlueAndi · 13 Nov 2007

Open the capi.conf file and read the instructions there.
Hint: You have to use the 4 for the controller.

gcf · 14 Nov 2007

7140 FON1 and FON2 CAPI - asterisk on FBF

Thanks for your help, analog in works fine now.
One last question, how can I manage the two analog internal extensions (FON1 & FON2) of my 7140 with asterisk through capi?

BlueAndi · 15 Nov 2007

You need a SIP-connection between the FritzBox and Asterisk.
Create 2 SIP connections in the Asterisk (sip.conf) and assign them in the FritzBox webinterface to the internet dial numbers.

Now you've been able to assign the FON1 and FON2 the SIPs for outgoing.

Take a look here at point 6 for the screenshots, maybe its easier to understand.

http://www.juerging.net/index.html?http://www.juerging.net/projekte/Fritzbox-Asterisk/

ramik · 13 Jan 2008

gcf schrieb:
Thanks for your help, analog in works fine now.
One last question, how can I manage the two analog internal extensions (FON1 & FON2) of my 7140 with asterisk through capi?

Can you please post the confif files you make to allow the analog in to be forwarded to an extension on asterisk?
i am able only to do outgoing calls from asterisk through the analog pstn line.

ramik · 13 Jan 2008

I managed to make my 7140's pstn ring on an extension in sip.

capi.conf

Code:

[ISDN3]          ; fritzbox 7050 internal S0
ntmode=no       ;if isdn card operates in nt mode, set this to yes
isdnmode=msn     ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
                 ;when using NT-mode, 'DID' should be set in any case
incomingmsn=*    ;allow incoming calls to this list of MSNs/DIDs, * = any
;defaultcid=123  ;set a default caller id to that interface for dial-out,
                 ;this caller id will be used when dial option 'd' is set.
controller=4     ;capi controller number to use
group=1          ;dialout group
softdtmf=off     ;enable/disable software dtmf detection, recommended for AVM cards
relaxdtmf=off    ;in addition to softdtmf, you can use relaxed dtmf detection
accountcode=     ;PBX accountcode to use in CDRs
context=mycapi ;context for incoming calls
;holdtype=hold   ;when the PBX puts the call on hold, ISDN HOLD will be used. If
                 ;set to 'local' (default value), no hold is done and the PBX may
                 ;play MOH.
immediate=yes   ;DID: immediate start of pbx with extension 's' if no digits were
                 ;     received on incoming call (no destination number yet)
                 ;MSN: start pbx on CONNECT_IND and don't wait for SETUP/SENDING-COMPLETE.
                 ;     info like REDIRECTINGNUMBER may be lost, but this is necessary for
                 ;     drivers/pbx/telco which does not send SETUP or SENDING-COMPLETE.
bridge=yes        ;native bridging (CAPI line interconnect) if available
devices=1        ;number of concurrent calls on this controller
                 ;(2 makes sense for single BRI, 30 for PRI)

extionsions.conf

Code:

[mycapi]
exten => s,1,Dial,SIP/771

also outbound trafic goes through the pstn when dialing from the softphone, by simply dialing the number with 5 as prefix

but now i have a new problem, how can i connect to this asterisk inside the fritz from remote computer?
i tried to open forward with virtualip to 192.168.178.253:5061 tcp / udp, but registration from remote doesn't work

dynamic · 14 Jan 2008

@ramik

Did you make your FBF reachable from the internet ( i.e. are you using DynDNS ) ?
If your box is generally reachable then you need to make further mandatory adjustments in your "sip.conf" as described in the default-example:
- update externhost to refer to your dynDNS hostname
- update localnet to match your loca IP Range

Then it should work

Regards
dynamic

ramik · 14 Jan 2008

I managed to access my asterfriz but through OpenVPN, i activated the OpenVPN in server mode and from remote machine first open a vpn connection then access asterisk with local 192.168.178.1 ip

for now it is ok later i will try to make it with sip.conf.

ramik · 16 Jan 2008

dynamic schrieb:
If your box is generally reachable then you need to make further mandatory adjustments in your "sip.conf" as described in the default-example:
- update externhost to refer to your dynDNS hostname
- update localnet to match your loca IP Range

Then it should work

I have been trying to make it work, but it seems that my client at work (behind nat) is not seeing the home asterisk, it keeps asking for registering.
on FritzBox forward rules i added rules to forward udp adn tcp on port 5061 to the ip 192.168.178.253 which is a virtual ip of the same fritz box... any other things i should make?

dynamic · 19 Jan 2008

@ramik
Is the Domain-Name of your FBF pingable from the internet ?

gcf · 19 Feb 2008

A couple of questions in English...

A couple of new questions:
-Is it possible to record calls to a storage device attached to the fritz!box?
-Is it possible to implement a simple answering machine?

Thanks in advance

dynamic · 19 Feb 2008

-Is it possible to record calls to a storage device attached to the fritz!box?
-Is it possible to implement a simple answering machine?

Yes, the default configuration files of the * installation already come with an example to set-up a voicemail without the app_voicemail.so module. In there you can also change the location of the files recorded to another location ( i.e. external storage ) if you want.

Regards
dynamic

Languages

Suche

Languages

Suche

Help in English for * on FBF, please

gcf

Neuer User

BlueAndi

Neuer User

gcf

Neuer User

gcf

Neuer User

BlueAndi

Neuer User

gcf

Neuer User

BlueAndi

Neuer User

ramik

Neuer User

ramik

Neuer User

dynamic

Aktives Mitglied

ramik

Neuer User

ramik

Neuer User

dynamic

Aktives Mitglied

gcf

Neuer User

dynamic

Aktives Mitglied

Zurzeit aktive Besucher

Neueste Beiträge

Statistik des Forums