the PRI Line is working, but without ring tone between asterisk and alcatel!
"i do not think the chan_dahdi will effect the sip to sip calls in asterisk. it is totally different things."
But it is the truth. I also did't understand this. This problem is reason why i am posting in this board.
So, did you know about a soulution or did you know about this Problem ??
hello:
you must test that in this way:
dial a extension from Alcatel to asterisk though PRI, and let the asterisk play IVR or voice, see what the incoming is?
can you post the debug info for the problem?
regards!
James.zhu
at the moment, i don't know how to implement the ivr system.
I tested the System once again.
Resolution:
If i am disconeting the alcatel pbx and the dahdi is up and the asterisk pbx is connected with the telco,
the call between sip phones are with ring tones.
But i don't know why. The effect with the disableing from the dahdi channel, is a becouse of disconnecting the Alcatel PBX.
Now i thinking this problem could be a result of the clocksource and master constelation of the dahdi channels.
Here is the currently connstelation:
hello:
i guess the setting in your both of sides from alcatel and D210P are wrong. actually the pri between asterisk and alcatel does not work properly due to the unsynchronized clock source! in you case, you can try set the first port of card connecting to PRi with salve clock and CPE and second port as NET connecting CPE with alctel.
regards!
James.zhu
so we found a reason why the ringing tone are disabled.
in our dialplan is the first try of dialing to connecting the alcatel PBX. In asterisk we use the application dail with this parameters.
<exten>,1,Dial(Dahdi/g2/<number>)
Asterisk output if no number accessible:
== Using UDPTL CoS mark 5
-- Executing [211@sip:1] Dial("SIP/223-082c5480", "DAHDI/g2/<number>") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/<number>
-- Channel 0/1, span 1 got hangup, cause 1
-- Hungup 'DAHDI/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
if the secound application is to dial a sip phone or a other phone, the ringtone are disabled.
> Now if the first application, or the first try of dialing a sip phone or a channnel over misdn the ringtone is hearing over the line.
Resulution: The Call from the asterisk pbx to alcatel disabling the ringtone.
But i don't why.
Could anybody verify this problem, and could anybody post a solution for this problem of dialing.
If someone need a log for the debug of the pri channels, please let me know what kind of debug of the pri channels are you need.
hello,
please make sure you have a right dialplan for that. what you mean by "Called g2/<number>"?
do you have a group 2 in your zapata.conf, does the system fit a number as your number in your dialplan?
if the log is a real log, of course, it can not dial out. at least it should be:
Called g2/339933 // call s testing number 339933. you call out from group 2 with the destination number 339933.
regards!
James.zhu
