I have been tryinf for the last 5 hours to make the b410p with dahdi
Use a german isdn anlagenanschluss.
Could someone post a working
/etc/dahdi/system.conf
And
/etc/asterisk/chan_dahdi.conf?
Thank you so much
Wonder
I have been tryinf for the last 5 hours to make the b410p with dahdi
Use a german isdn anlagenanschluss.
Could someone post a working
/etc/dahdi/system.conf
And
/etc/asterisk/chan_dahdi.conf?
Thank you so much
Wonder
As I only had my cell-phone to post the thread-starter, now the whole thing in more detail…
Hi,
I can’t make the B410P work with a german ISDN Line.
We have got 4 ISDN-Lines. 1 “Mehrgeräteanschluss” 3 “Anlagenanschluss”.
In case you don’t know what that is,
MEHRGERAETEanschluss gives you multiple Numbers for one ISDN Line
like 123456-1 123456-2 123456-3 123456-4 and so on… You can attach ISDN-Phones directly to that kind of line.
ANLAGENANSCHLUSS
gives you one large number-block that can span over multiple ISDN Lines.
like 123456-1 … 123456-99. You can’t atatch an ISDN-phone to that kind of line, it only works with PBXes
As I found out so far the kind of line is chosen by the signaling option in /etc/asterisk/chan-dahdi.conf.
Valid settings might be bri_cpe, bri_cpe_ptmp and bri_net
The “Mehrgeräteanschluss” actually works fine, but i can’t make the 3 “Anlagenanschluss” work.
What i did:
added a few options and assigned groups. As I am not at the customer any more at the moment, I can’t provide you with the exact config, but
this is my config for 1 of the ANLAGENANSCHLUSS spans:
group=2
context=in
switchtype=euroisdn
signalling=bri_cpe
echocancel=yes
overlapdial=yes
channel => 4-5
When i try to dial out on that span i just get
"unable to create channel of type dahdi cause 34 circuit channel congestion" in the asterisk CLI
when i try to dial the number of the ANLAGENANSCHLUSS from my cell, i just hear the congestion sound,
nothing happens on the asterisk CLI with Verbosity set to 100.
Can you provide me with a correct exampe /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf
for my setup?
Can you explain when i would use signalling method bri_cpe, bri_cpe_ptmp or bri_net?
I wasn’t able to find any official documentation on that.
I read that DAHDI now officialy supports the B410P so i didn’t install mISDN oder CAPI or other middleware.
Thanks a lot…
pflege:~# cat /etc/dahdi/system.conf
span=1,1,0,ccs,ami
bchan=1-2
hardhdlc=3
echocanceller=mg2,1-2
span=2,2,0,ccs,ami
bchan=4-5
hardhdlc=6
echocanceller=mg2,4-5
span=3,3,0,ccs,ami
bchan=7-8
hardhdlc=9
echocanceller=mg2,7-8
span=4,4,0,ccs,ami
bchan=10-11
hardhdlc=12
echocanceller=mg2,10-11
loadzone = de
defaultzone = de
#########################################
pflege:~# cat /etc/asterisk/chan_dahdi.conf
[channels]
; ANLAGENANSCHLUSS
language=de
group=2
switchtype = euroisdn
signalling = bri_cpe
pridialplan = local
prilocaldialplan = dynamic
nationalprefix = 0
internationalprefix = 00
localprefix = 208
privateprefix = 20822222
overlapdial=yes
immediate=yes
priindication = inband
echocancel = yes
context=in
channel => 4,5,7,8,10,11
; MEHRGERAETEANSCHLUSS
group=1
switchtype = euroisdn
pridialplan = local
prilocaldialplan = dynamic
nationalprefix = 0
internationalprefix = 00
localprefix = 208
privateprefix = 441111
overlapdial=yes
immediate=yes
priindication = inband
echocancel = yes
signalling = bri_cpe_ptmp
channel => 1,2
############################
/etc/asterisk/extensions.conf
exten => 301,1,Answer()
exten => 301,n,Dial(DAHDI/g1/01781111111,60)
exten => 301,n,Hangup()
exten => 302,1,Answer()
exten => 302,n,Dial(DAHDI/g2/017811111111,60)
exten => 302,n,Hangup()
#############################
Error when Dialing 302:
== Using SIP RTP CoS mark 5
-- Executing [302@sip:1] Answer("SIP/207-f00db3f8", "") in new stack
-- Executing [302@sip:2] Dial("SIP/207-f00db3f8", "DAHDI/g2/0178111111,60") in new stack
[Sep 5 12:43:06] WARNING[2953]: app_dial.c:1528 dial_exec_full: Unable to create channel of type ‘DAHDI’ (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [302@sip:3] Hangup("SIP/207-f00db3f8", "") in new stack
== Spawn extension (sip, 302, 3) exited non-zero on ‘SIP/207-f00db3f8’
After all it was just problems with the telco.
Well. i learned a lot.