[Asterisk-Users] No sound on 10% of incoming calls

Krystian Filiks krystian.filiks at kfiliks.com
Tue Feb 7 11:07:58 MST 2006


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/${EXTEN})
>
> exten=>_N.,3,Hangup()
>
> ;here I route the call to server2
>
> exten=>_0XXXXXXXXX,1,NoCDR
>
> exten=>_0XXXXXXXXX,2,Dial(IAX2/arpu:arpu at 192.168.1.3/${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" <soucany at app-line.com>/
>     Reply-To: /Asterisk Users Mailing List - Non-Commercial
>     Discussion<asterisk-users at lists.digium.com>/
>     To: /<asterisk-users at lists.digium.com>/
>     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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