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01.03.2008 at 06:36AM PST, ID: 23055547
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7.0

Conversation from analog telephone to a SIP-Client works but not from SIP to analog telephone

Asked by TREXman in Asterisk Open Source Telephony, Voice Over IP

Tags: , , ,

I have a little problem with asterisk on my Fritz!Box, i hope somebody can help me out.

Hardware&Configuration:
AVM Fritz!Box Fon 7170 with asterisk(1.2.19) on a USB-Stick;.(How-To´s i used http://www.ip-phone-forum.de/showthread.php?t=146132 and http://www.juerging.net/index.html?http://www.juerging.net/projekte/Fritzbox-Asterisk/ )

Fritz!Box Webinterface:
Analog telephone number(28); communication between Fritz Box an telephone system is ok;
SIP User 28; Sip-Server is localhost on Port 5061(5060 the std. Port is used by the Fritz!Box Software)
call-through: all call to 28(analog) are forwarded to SIP: 28

Asterisk: Configfiles see below
SIP-User: 9999
SIP-User: 28

all Calls to 28 are forwarded to 9999
all Call to XX are forwarded to the analoge Port

telephone system with a configured analog port(telephone nr. 28)

I tested this configuration:
when i try to call 28 then my wireles phone(9999) rings and i´m able to make a talk
if i dial some number on the telephone system the call is correctly forwarded to that number but if sombebody dial 28, my wlan telephone is ringing and i can answer, then i can hear what the other person is saying but if i want to say something the other person can´t hear it.

Thanks...
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sip.conf:
 
[general]
context=default			; Default context for incoming calls
 
bindport=5061			; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0		; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes			; Enable DNS SRV lookups on outbound calls
language=de
 
[9999]
context=intern
callerid=9999
host=dynamic
;qualify=no                     ; X-Lite is behind a NAT router
type=friend
user=9999
secret=1234
;disallow=all
;allow=gsm                     ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
 
[28]
context=28
callerid=28
host=dynamic
type=friend
user=28
secret=1234
 
 
extensions.conf:
 
[general]
;static=yes
;writeprotect=no
 
[analog]
exten => _.,1,Dial,SIP/9999
 
[intern]
exten => _.,1,Dial,CAPI/ISDN1/${EXTEN}
 
[28]
exten => _X.,1,Dial(SIP/9999)
 
 
capi.conf:
 
general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=de      ;set default language (en/de...)
 
[ISDN1]          ; fritzbox 7050/7170 external S0 (or external analog line: experimental)
ntmode=no      ;if isdn card operates in nt mode, set this to yes
isdnmode=did     ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
incomingmsn=*    ;allow incoming calls to this list of MSNs/DIDs, * = any,
                       ;a ="analog controller": empty incoming msn gets replaced
		       ;with defaultcid (-> fritzbox 7050/7170 at analog line)
;defaultcid=1234567  ;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 (=4: fritzbox 7050/7150 at analog line)
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=analog   ;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.
bridge=no      ;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)
immediate=yes
;echotail=64
 
 
[+][-]01.03.2008 at 06:45AM PST, ID: 20573144

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[+][-]01.03.2008 at 08:36AM PST, ID: 20574121

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[+][-]01.05.2008 at 05:28AM PST, ID: 20589041

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[+][-]01.05.2008 at 05:33AM PST, ID: 20589045

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[+][-]01.08.2008 at 06:27AM PST, ID: 20608749

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[+][-]01.08.2008 at 06:50AM PST, ID: 20608991

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About this solution

Zones: Asterisk Open Source Telephony, Voice Over IP
Tags: asterisk, 1.2.19, AVM, Fritz!Box Fon 7170
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Solution Provided By: grblades
Participating Experts: 1
Solution Grade: B
 
 
[+][-]01.09.2008 at 05:24AM PST, ID: 20617841

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