What do you do with the other 15 channels?

your zapata.conf says:
channel => 1-15 ;,17-31 => only 15 first channels on PRI

but your zaptel.conf says:
span=1,1,0,ccs,hdb3
bchan = 1-15, 17-31

You use all 30 channels in Zaptel.conf but only 15 in zapta.conf
I never configured Zap on asterisk and frankly do not have a clue how to and I do not have a clue what the both files do, but the use of 15 channels only, makes me wonder.

Did you make a ISDN trace what do the Setup message etc... say which channel is requested by France Telecom and on which channel is the call setup?

Why I ask.
Dead air (2way) usually means channel mismatch, seen this happen many times, the D channel is on kick 16 and you have 15 channels in one file configured and 30 in another.

Why only 15 channels?

Krystian


Joe Tahan wrote:



AnyOne? any help?

As I'm looking at your zapata.conf I recall a problem in receiving dial-outs from a non-asterisk IVR to an * server1 and server1 routs the call to server2 with IAX2 in order to make a final dial command to a ZAP channel, but in server2 cli console I get the error (UNABLE TO CREAT CHANNEL OF TYPE ZAP) , this is my zapata.conf setup:

[channels]

language=en

context=inbound

switchtype=euroisdn

pridialplan=national

prilocaldialplan=national

signalling=pri_cpe

rxwink=300 ; Atlas seems to use long (250ms) winks

usecallerid=yes

hidecallerid=no

callwaiting=yes

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=no

transfer=no

cancallforward=no

callreturn=no

relaxdtmf=yes

rxgain=0.0

txgain=0.0

group=1

callgroup=1

pickupgroup=1

immediate=no

callerid=asreceived

amaflags=billing

busydetect=yes

busycount=8

channel=>32-46,48-62,63-77,79-93,94-108,110-124

channel=>125-139,141-155,156-170,172-186,187-201,203-217

group=2

context=test

channel=>1-15,17-31

;Arpu trunk

group=3

context=arpu

signalling=pri_net

channel=>218-232,234-248

extensions.conf :

[arpu]

exten=>_N.,1,NoCDR

exten=>_N.,2,Dial(Zap/r2/${EXTEN})

exten=>_N.,3,Hangup()

;here I route the call to server2

exten=>_0XXXXXXXXX,1,NoCDR

exten=>_0XXXXXXXXX,2,Dial(IAX2/arpu:[EMAIL PROTECTED]/${EXTEN})

exten=>_0XXXXXXXXX,3,SoftHangup(${CHANNEL})

and server2 zapata.conf:

[channels]

language=en

context=inbound

switchtype=euroisdn

pridialplan=national

prilocaldialplan=national

signalling=pri_cpe

rxwink=300 ; Atlas seems to use long (250ms) winks

usecallerid=yes

hidecallerid=no

callwaiting=yes

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=no

transfer=no

cancallforward=no

callreturn=no

echocancel=no

relaxdtmf=yes

rxgain=0.0

txgain=0.0

group=1

callgroup=1

pickupgroup=1

immediate=no

callerid=asreceived

amaflags=billing

busydetect=yes

busycount=8

;

channel=>1-15,17-31

channel=>32-46,48-62

channel=>63-77,79-93

;Arpu trunk

group=3

context=arpu

signalling=pri_cpe

channel=>94-108,110-124

where extensions.conf for server2 is:

[arpuvoip]

;here I place a Zap call and the console shows (Unable to create a channel of type ZAP)

exten=>_0XXXXXXXXX,1,Answer()

exten=>_0XXXXXXXXX,2,Dial(Zap/g1/${EXTEN})

exten=>_0XXXXXXXXX,3,Hangup()

Any Ideas?

Truely/

Joe

    ------------------------------------------------------------------------
    From: /"Jerome SOUCANY" <[EMAIL PROTECTED]>/
    Reply-To: /Asterisk Users Mailing List - Non-Commercial
    Discussion<[email protected]>/
    To: /<[email protected]>/
    Subject: /[Asterisk-Users] No sound on 10% of incoming calls/
    Date: /Tue, 7 Feb 2006 11:03:49 +0100/
    >Hello,
    >
    >I have a problem with Asterisk, on 10% of incoming calls the IP
    Phone ring
    >but I don't hear the caller and the caller doesn't hear me (all
    IP Phones
    >have the same problem).
    >
    >This problem appear also if the call is directly send to the
    second E1 of
    >the digium card who is connected to an IVR.
    >
    >It does not depand on the charge of the server (I have the
    problem with only
    >one call).
    >
    >The configuration :
    >
    >PRI (France Telecom) 15 channels <====> Asterisk <=====> IP Phone
    >
    >* Server :
    > - Dell power edge 1800SC
    > - 2 Ethernet cards (LAN + VoIP LAN)
    > - Digium card : TE 405P
    > - Linux Mandriva LE 2005 (10.2) :
    > Linux ASTERISK 2.6.11-12mdksmp #1 SMP i686 Intel(R) Xeon(TM) CPU
    >3.00GHz unknown GNU/Linux
    > - Asterisk 1.2.4
    > - Zaptel 1.2.3
    > - Libpri 1.2.2
    >
    >* IP Phone :
    > SNOM 320 (latest firmware)
    >
    >============================================
    >zaptel.conf
    >
    >span=1,1,0,ccs,hdb3
    >span=2,1,0,ccs,hdb3,crc4,yellow
    >span=3,1,0,ccs,hdb3,crc4,yellow
    >span=4,1,0,ccs,hdb3,crc4,yellow
    >
    >bchan = 1-15, 17-31
    >dchan = 16
    >bchan = 32-46,48-62
    >dchan = 47
    >bchan = 63-77,79-93
    >dchan = 78
    >bchan = 94-108,110-124
    >dchan = 109
    >
    >loadzone = fr
    >defaultzone = fr
    >
    >============================================
    >
    >============================================
    >zapata.conf
    >
    >[channels]
    >switchtype=euroisdn
    >pridialplan=national
    >signalling=pri_cpe
    >usecallerid=yes
    >hidecallerid=yes
    >usecallingpres=no
    >callwaiting=yes
    >callwaitingcallerid=yes
    >threewaycalling=yes
    >transfer=yes
    >cancallforward=yes
    >echocancel=yes
    >echocancelwhenbridged=yes
    >echotraining=yes
    >rxgain=0.0
    >txgain=-6.0
    >
    >group=1
    >callgroup=1
    >pickupgroup=1
    >
    >immediate=no
    >callprogress=yes
    >
    >callerid=asreceived
    >group=1
    >context=from-pstn
    >signalling=pri_cpe
    >channel => 1-15 ;,17-31 => only 15 first channels on PRI
    >
    >group=2
    >context=from-ivr
    >signalling=pri_net
    >channel => 32-46,48-62
    >
    >group=3
    >context=from-ivr-bis
    >signalling=pri_net
    >channel => 63-77,79-93
    >
    >group=4
    >signalling=pri_net
    >channel => 94-108,110-124
    >============================================
    >
    >
    >
    >
    >Any ideas ?
    >
    >
    >
    >Regards
    >
    >Jerome
    >
    >
    >_______________________________________________
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    >
    >Asterisk-Users mailing list
    >To UNSUBSCRIBE or update options visit:
    > http://lists.digium.com/mailman/listinfo/asterisk-users


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