Alcatel Unleashed

krzysioD wrote:anyone have sucessuly used qsig w/ asterisk (via pri of course)

For me works fine, sangoma + PRA2.

But I have other problem - sometimes link hungs up. I cannot make call 4400->Asterisk, after one call Asterisk->4400, link go back to normal state and randomly after some time hung up again. I have the same situation with qsig and isdn Any ideas ?

QSIG should work with the latest libpri, if you switch the OXE to ISO mode.
In the last half year there has been added some more features like name presentation in libpri.

I made this week an patch for someone with an hipath, connected to asterisk, to let the called name display (i.e. some alcatel user dials in to asterisk, so asterisk returns, while in ringing state, the name of the called extension back to the switch). But this isn't tested yet - i have no hardware here, for tests with libpri.

Mario

I followed the instructions and have this setup:

Asterisk -> OXE R6.2 -> PSTN

I'm able to call extensions on Alcatel and vice versa. My problem is calling a PSTN line.

When i dial a number like 2814059 from my sip extension 220

Asterisk CLI shows:

Called g1/2814059
zap/1-1 is proceeding passing it to SIP/220-b7907578
Channel 0/1 span 1 got hangup, cause 1
Hungup 'Zap/1-1'
Everyone is busy/congested at this time (1:0/0/1)

without an trace from your OXE (t3) or Asterisk, it's hard to say, where the problem is...

Mario

Oxe gives:
mtracer started ...
______________________________________________________________________________
| (713865:000028) Physical-Event :
| long: 45 desti: 0 source: 0 cryst: 0 cpl: 9 us: 0 term: 0 type a5
| tei: 0 >>>> message received : SETUP [05] Call ref : 00 0a
|______________________________________________________________________________
|
| IE:[04] BEARER_CAPABILITY (l=3) 80 90 a3
| IE:[18] CHANNEL (l=3) a9 83 81 -> T2 : B channel 1 exclusive
| IE:[6c] CALLING_NUMBER (l=5) -> 21 80 Num : 220
| IE:[70] CALLED_NUMBER (l=8) -> c1 Num : 2814059
|______________________________________________________________________________

______________________________________________________________________________
| (713865:000029) 979: Send_IO1 (link-nbr=9, sapi=0, tei=0) :
| long: 23 desti: 0 source: 15 cryst: 0 cpl: 9 us: 8 term: 0 type a5
| tei: 0 <<<< message sent : CALL PROC (02) Call ref : 80 0a
|______________________________________________________________________________
|
| IE:[18] CHANNEL (l=3) a1 83 81 -> T2 : B channel 1 preferred
|______________________________________________________________________________

______________________________________________________________________________
| (713865:000030) 979: Send_IO1 (link-nbr=9, sapi=0, tei=0) :
| long: 23 desti: 0 source: 15 cryst: 0 cpl: 9 us: 8 term: 0 type a5
| tei: 0 <<<< message sent : DISCONNECT [45] Call ref : 80 0a
|______________________________________________________________________________
|
| IE:[08] CAUSE (l=3) 80 81 80 -> [81] UNALLOCATED NUMBER
|______________________________________________________________________________

______________________________________________________________________________
| (713865:000031) Physical-Event :
| long: 22 desti: 0 source: 0 cryst: 0 cpl: 9 us: 0 term: 0 type a5
| tei: 0 >>>> message received : RELEASE [4d] Call ref : 00 0a
|______________________________________________________________________________
|
| IE:[08] CAUSE (l=2) 81 81 -> [81] UNALLOCATED NUMBER
|______________________________________________________________________________

______________________________________________________________________________
| (713865:000032) 979: Send_IO1 (link-nbr=9, sapi=0, tei=0) :
| long: 23 desti: 0 source: 15 cryst: 0 cpl: 9 us: 8 term: 0 type a5
| tei: 0 <<<< message sent : REL COMP [5a] Call ref : 80 0a
|______________________________________________________________________________
|
| IE:[08] CAUSE (l=3) 80 90 80 -> [90] NORMAL CALL CLEARING
|______________________________________________________________________________


(1)xa000000>


Asterisk gives:

-- Got SIP response 400 "Bad Request" back from 10.0.0.80
-- Executing Answer("SIP/220-0920c9b8", "") in new stack
-- Executing Dial("SIP/220-0920c9b8", "zap/g1/2814059") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/2814059
-- Zap/1-1 is proceeding passing it to SIP/220-0920c9b8
-- Channel 0/1, span 1 got hangup request, cause 1
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)

Ok, the OXE refuses the call, because of a invalid number. Check your numbering plan in your OXE (and your trunk settings - set did transcoding to false).

Mario

does your zapata.conf contain:

overlapdialing=yes ?

btw: strange, dialed numbers should be avalible in t3 isdn trace

krzysioD wrote:does your zapata.conf contain:

overlapdialing=yes ?

btw: strange, dialed numbers should be avalible in t3 isdn trace

where is the problem - the dialled number is visible in this trace...
The OXE simply complains about a wrong number...

Mario

You are dialing very long number.

Should you use a ARS prefix like 0 or 9 or a Trunk Group prefix like *#*# 19 ?

Asterisk is dialing a very long (7 digit) number. Is you oxe having users with that kind of directory numbers assigned?

or, maybe you should set "type of number" aka TON in asterisk, so asterisk will send a "0" or "00" to PRA/PRI line.

see:
| IE:[6c] CALLING_NUMBER (l=5) -> 21 80 Num : 220
| IE:[70] CALLED_NUMBER (l=8) -> c1 Num : 2814059

220 is presentation of asterisk. It looks good.

the 281 40 59 looks like number without area code, you really need to pass some ARS prefix (like "0") and some area code.

Try with something simple: dial some directory number on you OXE like 100 or 1000

Thanks guys, my ARS wasn't working well. I created a prefix 503 for the trunks i want to access. Now am able to dial 503 then the 281... number. It works fine.