RE: [Asterisk-Users] No sound on 10% of incoming calls
Hello, I changed these parameters in zapata.conf : callprogress=no busydetect=no And now it's working fine. Jerome -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jerome SOUCANY Envoyé : mardi 7 février 2006 11:04 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] No sound on 10% of incoming calls 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No sound on 10% of incoming calls
Not really sure, but once I had a problem when I changed the txgain and rxgain, so set them again to 0.0 and see how it will work. Truely/ Ammar From: "Jerome SOUCANY" [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] No sound on 10% of incoming callsDate: Tue, 7 Feb 2006 11:03:49 +0100Hello,I have a problem with Asterisk, on 10% of incoming calls the IP Phone ringbut I don't hear the caller and the caller doesn't hear me (all IP Phoneshave the same problem).This problem appear also if the call is directly send to the second E1 ofthe digium card who is connected to an IVR.It does not depand on the charge of the server (I have the problem with onlyone 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) CPU3.00GHz unknown GNU/Linux - Asterisk 1.2.4 - Zaptel 1.2.3 - Libpri 1.2.2* IP Phone : SNOM 320 (latest firmware)zaptel.confspan=1,1,0,ccs,hdb3span=2,1,0,ccs,hdb3,crc4,yellowspan=3,1,0,ccs,hdb3,crc4,yellowspan=4,1,0,ccs,hdb3,crc4,yellowbchan = 1-15, 17-31dchan = 16bchan = 32-46,48-62dchan = 47bchan = 63-77,79-93dchan = 78bchan = 94-108,110-124dchan = 109loadzone = frdefaultzone = frzapata.conf[channels]switchtype=euroisdnpridialplan=nationalsignalling=pri_cpeusecallerid=yeshidecallerid=yesusecallingpres=nocallwaiting=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yesechocancel=yesechocancelwhenbridged=yesechotraining=yesrxgain=0.0txgain=-6.0group=1callgroup=1pickupgroup=1immediate=nocallprogress=yescallerid=asreceivedgroup=1context=from-pstnsignalling=pri_cpechannel = 1-15 ;,17-31 = only 15 first channels on PRIgroup=2context=from-ivrsignalling=pri_netchannel = 32-46,48-62group=3context=from-ivr-bissignalling=pri_netchannel = 63-77,79-93group=4signalling=pri_netchannel = 94-108,110-124Any ideas ?RegardsJerome___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersOpen your e-mail without having to worry about viruses with MSN Premium. Join now and get the first two months FREE*< /html> ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No sound on 10% of incoming calls
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/${EXTEN})
exten=_N.,3,Hangup()
;here I route the call to server2
exten=_0X,1,NoCDR
exten=_0X,2,Dial(IAX2/arpu:[EMAIL PROTECTED]/${EXTEN})
exten=_0X,3,SoftHangup(${CHANNEL})
and server2 zapata.conf:
[channels]
language=en
context=inbound
switchtype=euroisdn
pridialplan=national
prilocaldialplan=national
signalling=pri_cpe
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=no
transfer=no
cancallforward=no
callreturn=no
echocancel=no
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
callerid=asreceived
amaflags=billing
busydetect=yes
busycount=8
;
channel=1-15,17-31
channel=32-46,48-62
channel=63-77,79-93
;Arpu trunk
group=3
context=arpu
signalling=pri_cpe
channel=94-108,110-124
where extensions.conf for server2 is:
[arpuvoip]
;here I place a Zap call and the console shows (Unable to create a channel of type ZAP)
exten=_0X,1,Answer()
exten=_0X,2,Dial(Zap/g1/${EXTEN})
exten=_0X,3,Hangup()
Any Ideas?
Truely/
Joe
From: "Jerome SOUCANY" [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] No sound on 10% of incoming callsDate: Tue, 7 Feb 2006 11:03:49 +0100Hello,I have a problem with Asterisk, on 10% of incoming calls the IP Phone ringbut I don't hear the caller and the caller doesn't hear me (all IP Phoneshave the same problem).This problem appear also if the call is directly send to the second E1 ofthe digium card who is connected to an IVR.It does not depand on the charge of the server (I have the problem with onlyone 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) CPU3.00GHz unknown GNU/Linux - Asterisk 1.2.4 - Zaptel 1.2.3 - Libpri 1.2.2* IP Phone : SNOM 320 (latest firmware)zaptel.confspan=1,1,0,ccs,hdb3span=2,1,0,ccs,hdb3,crc4,yellowspan=3,1,0,ccs,hdb3,crc4,yellowspan=4,1,0,ccs,hdb3,crc4,yellowbchan = 1-15, 17-31dchan = 16bchan = 32-46,48-62dchan = 47bchan = 63-77,79-93dchan = 78bchan = 94-108,110-124dchan =
109loadzone = frdefaultzone = frzapata.conf[channels]switchtype=euroisdnpridialplan=nationalsignalling=pri_cpeusecallerid=yeshidecallerid=yesusecallingpres=nocallwaiting=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yesechocancel=yesechocancelwhenbridged=yesechotraining=yesrxgain=0.0txgain=-6.0group=1callgroup=1pickupgroup=1immediate=nocallprogress=yescallerid=asreceivedgroup=1context=from-pstnsignalling=pri_cpechannel = 1-15 ;,17-31 = only 15 first channels on
PRIgroup=2context=from-ivrsignalling=pri_netchannel = 32-46,48-62group=3context=from-ivr-bissignalling=pri_netchannel = 63-77,79-93group=4signalling=pri_netchannel = 94-108,110-124Any ideas ?RegardsJerome___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersFree yourself from those irritating pop-up ads with MSN Premium. Join now and get the first two months FREE*
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Re: [Asterisk-Users] No sound on 10% of incoming calls
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/${EXTEN})
exten=_N.,3,Hangup()
;here I route the call to server2
exten=_0X,1,NoCDR
exten=_0X,2,Dial(IAX2/arpu:[EMAIL PROTECTED]/${EXTEN})
exten=_0X,3,SoftHangup(${CHANNEL})
and server2 zapata.conf:
[channels]
language=en
context=inbound
switchtype=euroisdn
pridialplan=national
prilocaldialplan=national
signalling=pri_cpe
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=no
transfer=no
cancallforward=no
callreturn=no
echocancel=no
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
callerid=asreceived
amaflags=billing
busydetect=yes
busycount=8
;
channel=1-15,17-31
channel=32-46,48-62
channel=63-77,79-93
;Arpu trunk
group=3
context=arpu
signalling=pri_cpe
channel=94-108,110-124
where extensions.conf for server2 is:
[arpuvoip]
;here I place a Zap call and the console shows (Unable to create a
channel of type ZAP)
exten=_0X,1,Answer()
exten=_0X,2,Dial(Zap/g1/${EXTEN})
exten=_0X,3,Hangup()
Any Ideas?
Truely/
Joe
From: /Jerome SOUCANY [EMAIL PROTECTED]/
Reply-To: /Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com/
To: /asterisk-users@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
