Hello,

We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround?

Thanks for any advice.

Hi;

How can I know the duration that the DAHDI channel is still used? I need to know its status and since when it is in this status, how?

Also, is it possible to hangup the channel if it has been openned more than 90 minute? Other than using the timeout in the Dial command (because this I know it).

What is happening with me that from time to time, I find some DAHDI channels are stayed connected (stuck) for long time. I know how to write the extensions.conf in a way to handle the hangup properly, also I send the incoming calls to the voicemail to be sure it is hanged up properly. One more thing, I set the rtptimeout in case there is any problem in the sip phone and the network .. But, still after sometime, I am surprised that some channels are stuck and stayed connected and then I have to reset it manually !! This is happening only in the analoge channels.

What other than the rtptimeout, the hangup in the extensions.conf, the voicemail? Is there anything I have to take care for it that might cause this stuck and keeping the channel openned?

By the way, for such cases, what should I place the value of the rtpkeepalive as currently it is 0?

What other things I have to take care for it?

Regards Bilal

Hello listers,

I’m trying to run DAHDI 1.4 on a 3.0 Debian Kernel in an embedded system, but have faced lots of problems mainly because it has lots of functions looking for the PCI.

Have seen so many problems, I’m in fact thinking it cannot be possibly done
(at least not in a couple of weeks, by one only man). Has anyone out there had any experience on something like this? or can someone shed some light on how to overcome this issues?

Any ideas are very welcome

Hi

I want to know actual DAHDI channel number (pseudo), which received the call or dialed the call. Where as when Asterisk receives a call on DAHDI
channel, it shows channel as “DAHDI/i5/112-15″
Is there any way / configuration to change this behavior and get actual channel number? Earlier we used to get this.

I am using Asterisk 1.8.11 and Dahdi 2.4

Thanks & Regards, Amit Patkar

Hi,


I have 6 Red FXO with TDM2400p in my PC. I have install asterisk and DAHDI driver. Scenario is

jitsi—–> asterisk server—–> analog PBX —-> landline phone

I configured this scenario as follow

in chan_dahdi.conf file

; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
;FXO Modules
group=2
echocancel=yes
signalling=fxs_ks
context=Incoming
channel=1-20



After loading module in astrisk giving o/p below

module load chan_dahdi.so
Loaded chan_dahdi.so
== Parsing ‘/etc/asterisk/chan_dahdi.conf’: == Found
== Parsing ‘/etc/asterisk/users.conf’: == Found
— Registered channel 1, FXS Kewlstart signalling
— Registered channel 2, FXS Kewlstart signalling
— Registered channel 3, FXS Kewlstart signalling
— Registered channel 4, FXS Kewlstart signalling
— Registered channel 5, FXS Kewlstart signalling
— Registered channel 6, FXS Kewlstart signalling
— Registered channel 7, FXS Kewlstart signalling
— Registered channel 8, FXS Kewlstart signalling
— Registered channel 9, FXS Kewlstart signalling
— Registered channel 10, FXS Kewlstart signalling
— Registered channel 11, FXS Kewlstart signalling
— Registered channel 12, FXS Kewlstart signalling
— Registered channel 13, FXS Kewlstart signalling
— Registered channel 14, FXS Kewlstart signalling
— Registered channel 15, FXS Kewlstart signalling
— Registered channel 16, FXS Kewlstart signalling
— Registered channel 17, FXS Kewlstart signalling
— Registered channel 18, FXS Kewlstart signalling
— Registered channel 19, FXS Kewlstart signalling
— Registered channel 20, FXS Kewlstart signalling
— Automatically generated pseudo channel
[Nov 2 14:38:50] WARNING[1886]: chan_dahdi.c:17278 process_dahdi:
Ignoring any changes to ‘userbase’ (on reload) at line 23.
[Nov 2 14:38:50] WARNING[1886]: chan_dahdi.c:17278 process_dahdi:
Ignoring any changes to ‘vmsecret’ (on reload) at line 31.
[Nov 2 14:38:50] WARNING[1886]: chan_dahdi.c:17278 process_dahdi:
Ignoring any changes to ‘hassip’ (on reload) at line 35.
[Nov 2 14:38:50] WARNING[1886]: chan_dahdi.c:17278 process_dahdi:
Ignoring any changes to ‘hasiax’ (on reload) at line 39.
[Nov 2 14:38:50] WARNING[1886]: chan_dahdi.c:17278 process_dahdi:
Ignoring any changes to ‘hasmanager’ (on reload) at line 47.
== Registered channel type ‘DAHDI’ (DAHDI Telephony Driver)
== Manager registered action DAHDITransfer
== Manager registered action DAHDIHangup
== Manager registered action DAHDIDialOffhook
== Manager registered action DAHDIDNDon
== Manager registered action DAHDIDNDoff
== Manager registered action DAHDIShowChannels
== Manager registered action DAHDIRestart
Loaded chan_dahdi.so => (DAHDI Telephony Driver)

In my extension.conf file i wrote dialplan for user so sandeep is jitsi user and 81 and 88 is landline number.


[general]
static=yes
writeprotect=no
clearglobalvars=no

[Incoming]
exten => s,1,Answer
exten => s,2,Dial(DAHDI/g1,20,rt)
exten => s,3,Voicemail(1000,u)
exten => s,103,Voicemail(1000,b)
exten => sandeep,1,Dial(SIP/sandeep)
exten => sandeep,n,Hangup()

exten => 1004,4,Dial(SIP/sandeep)
exten => 1004,n,Hangup()
; Testing extension, prepare to be insulted like a
; Monthy Python knight

exten => 81,1,Dial(DAHDI/1,20,rt)
exten => 81,n,Hangup()

exten => 88,1,Dial(DAHDI/1,20,rt)
exten => 88,n,Hangup()

exten => 8500,1,VoiceMailMain
exten => 8501,1,MusicOnHold
exten => _9.,1,Dial(DAHDI/g2/www${EXTEN:1})
exten => _9.,2,Congestion

exten => 201,1,Answer()
exten => 201,n,Playback(tt-monty-knights)
exten => 201,n,Hangup()

; Echo-test, it is good to test if we have sound in both directions.
; The call is answered
exten => 202,1,Answer()
; Welcome message is played
exten => 202,n,Playback(welcome)
; Play information about the echo test
exten => 202,n,Playback(demo-echotest)
; Do the echo test, end with the # key
exten => 202,n,Echo()
; Plays information that the echo test is done
exten => 202,n,Playback(demo-echodone)
; Goodbye message is played
exten => 202,n,Playback(vm-goodbye)
; Hangup() ends the call, hangs up the line
exten => 202,n,Hangup()


After loading extension and dahdi, i called from jitsi and dialed 81 but asterisk is giving o/p as below and busy tone is coming on jitsi

— Executing [81@myphones:1] Dial(“SIP/sandeep-00000000″,
“DAHDI/1,20,rt”) in new stack
— Called 1
[Nov 2 14:45:31] WARNING[2145]: chan_dahdi.c:7536 handle_alarms:
Detected alarm on channel 1: Red Alarm
— Hanging up on ‘DAHDI/1-1′
— Hungup ‘DAHDI/1-1′
== Everyone is busy/congested at this time (1:0/0/1)
— Executing [81@myphones:2] Hangup(“SIP/sandeep-00000000″, “”) in new stack
== Spawn extension (myphones, 81, 2) exited non-zero on
‘SIP/sandeep-00000000′


Any help to resolve this problem.

THanks