thr3ads.net - asterisk users - [Asterisk-Users] No sound on 10% of incoming calls [Feb 2006]

If this information is useful, please help other people find it:
Share via:

Jerome SOUCANY

2006-Feb-07 03:03 UTC

[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

Joe Tahan

2006-Feb-07 09:41 UTC

head link

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

<html><div style='background-color:'><P>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.<BR><BR></P>
<P>Truely/</P>
<P>Ammar</P>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT:
#a0c6e5 2px solid; MARGIN-RIGHT: 0px"><FONT style="FONT-SIZE:
11px; FONT-FAMILY: tahoma,sans-serif">
<HR color=#a0c6e5 SIZE=1>
From: <I>"Jerome SOUCANY"
<soucany@app-line.com></I><BR>Reply-To: <I>Asterisk
Users Mailing List - Non-Commercial
Discussion<asterisk-users@lists.digium.com></I><BR>To:
<I><asterisk-users@lists.digium.com></I><BR>Subject:
<I>[Asterisk-Users] No sound on 10% of incoming
calls</I><BR>Date: <I>Tue, 7 Feb 2006 11:03:49
+0100</I><BR>>Hello,<BR>><BR>>I have a problem
with Asterisk, on 10% of incoming calls the IP Phone ring<BR>>but I
don't hear the caller and the caller doesn't hear me (all IP
Phones<BR>>have the same problem).<BR>><BR>>This
problem appear also if the call is directly send to the second E1
of<BR>>the digium card who is connected to an
IVR.<BR>><BR>>It does not depand on the charge of the server
(I have the problem with only<BR>>one
call).<BR>><BR>>The configuration
:<BR>><BR>>PRI (France Telecom) 15 channels
<====> Asterisk <=====> IP Phone<BR>><BR>>* Server
:<BR>> - Dell power edge 1800SC<BR>> - 2 Ethernet cards (LAN +
VoIP LAN)<BR>> - Digium card : TE 405P<BR>> - Linux Mandriva
LE 2005 (10.2) :<BR>> Linux ASTERISK 2.6.11-12mdksmp #1 SMP i686
Intel(R) Xeon(TM) CPU<BR>>3.00GHz unknown GNU/Linux<BR>> -
Asterisk 1.2.4<BR>> - Zaptel 1.2.3<BR>> - Libpri
1.2.2<BR>><BR>>* IP Phone :<BR>> SNOM 320 (latest
firmware)<BR>><BR>>============================================<BR>>zaptel.conf<BR>><BR>>span=1,1,0,ccs,hdb3<BR>>span=2,1,0,ccs,hdb3,crc4,yellow<BR>>span=3,1,0,ccs,hdb3,crc4,yellow<BR>>span=4,1,0,ccs,hdb3,crc4,yellow<BR>><BR>>bchan
= 1-15, 17-31<BR>>dchan = 16<BR>>bchan =
32-46,48-62<BR>>dchan = 47<BR>>bchan =
63-77,79-93<BR>>dchan = 78<BR>>bchan =
94-108,110-124<BR>>dchan =
109<BR>><BR>>loadzone = fr<BR>>defaultzone =
fr<BR>><BR>>============================================<BR>><BR>>============================================<BR>>zapata.conf<BR>><BR>>[channels]<BR>>switchtype=euroisdn<BR>>pridialplan=national<BR>>signalling=pri_cpe<BR>>usecallerid=yes<BR>>hidecallerid=yes<BR>>usecallingpres=no<BR>>callwaiting=yes<BR>>callwaitingcallerid=yes<BR>>threewaycalling=yes<BR>>transfer=yes<BR>>cancallforward=yes<BR>>echocancel=yes<BR>>echocancelwhenbridged=yes<BR>>echotraining=yes<BR>>rxgain=0.0<BR>>txgain=-6.0<BR>><BR>>group=1<BR>>callgroup=1<BR>>pickupgroup=1<BR>><BR>>immediate=no<BR>>callprogress=yes<BR>><BR>>callerid=asreceived<BR>>group=1<BR>>context=from-pstn<BR>>signalling=pri_cpe<BR>>channel
=> 1-15 ;,17-31 => only 15 first channels on
PRI<BR>><BR>>group=2<BR>>context=from-ivr<BR>>signalling=pri_net<BR>>channel
=>
32-46,48-62<BR>><BR>>group=3<BR>>context=from-ivr-bis<BR>>signalling=pri_net<BR>>channel
=>
63-77,79-93<BR>><BR>>group=4<BR>>signalling=pri_net<BR>>channel
=>
94-108,110-124<BR>>============================================<BR>><BR>><BR>><BR>><BR>>Any
ideas
?<BR>><BR>><BR>><BR>>Regards<BR>><BR>>Jerome<BR>><BR>><BR>>_______________________________________________<BR>>--Bandwidth
and Colocation provided by Easynews.com
--<BR>><BR>>Asterisk-Users mailing list<BR>>To
UNSUBSCRIBE or update options visit:<BR>>
http://lists.digium.com/mailman/listinfo/asterisk-users<BR></FONT></BLOCKQUOTE></div><br
clear=all><hr>Open your e-mail without having to worry about viruses
with  <a href="http://g.msn.com/8HMAENCA/2737??PS=47575"
target="_top">MSN Premium.</a> Join now and get the first two
months FREE*</html>

Joe Tahan

2006-Feb-07 10:19 UTC

head link

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

<html><div
style='background-color:'><P><BR><BR></P>
<DIV>
<P>AnyOne? any help?</P>
<P>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:</P><FONT size=1>
<P>[channels]</P>
<P>language=en</P>
<P>context=inbound</P>
<P>switchtype=euroisdn</P>
<P>pridialplan=national</P>
<P>prilocaldialplan=national</P>
<P>signalling=pri_cpe</P>
<P>rxwink=300 ; Atlas seems to use long (250ms) winks</P>
<P>usecallerid=yes</P>
<P>hidecallerid=no</P>
<P>callwaiting=yes</P>
<P>usecallingpres=yes</P>
<P>callwaitingcallerid=yes</P>
<P>threewaycalling=no</P>
<P>transfer=no</P>
<P>cancallforward=no</P>
<P>callreturn=no</P>
<P>relaxdtmf=yes</P>
<P>rxgain=0.0</P>
<P>txgain=0.0</P>
<P>group=1</P>
<P>callgroup=1</P>
<P>pickupgroup=1</P>
<P>immediate=no</P>
<P>callerid=asreceived</P>
<P>amaflags=billing</P>
<P>busydetect=yes</P>
<P>busycount=8</P>
<P>channel=>32-46,48-62,63-77,79-93,94-108,110-124</P>
<P>channel=>125-139,141-155,156-170,172-186,187-201,203-217</P>
<P>group=2</P>
<P>context=test</P>
<P>channel=>1-15,17-31</P>
<P>;Arpu trunk</P>
<P>group=3</P>
<P>context=arpu</P>
<P>signalling=pri_net</P>
<P>channel=>218-232,234-248</P></FONT>
<P>&nbsp;</P>
<P>extensions.conf :</P><FONT size=1>
<P>[arpu]</P>
<P>exten=>_N.,1,NoCDR</P>
<P>exten=>_N.,2,Dial(Zap/r2/${EXTEN})</P>
<P>exten=>_N.,3,Hangup()</P>
<P>;here I route the call to server2</P>
<P>exten=>_0XXXXXXXXX,1,NoCDR</P>
<P>exten=>_0XXXXXXXXX,2,Dial(IAX2/arpu:arpu@192.168.1.3/${EXTEN})</P>
<P>exten=>_0XXXXXXXXX,3,SoftHangup(${CHANNEL})</P></FONT>
<P>&nbsp;</P>
<P>and server2 zapata.conf:</P><FONT size=1>
<P>[channels]</P>
<P>language=en</P>
<P>context=inbound</P>
<P>switchtype=euroisdn</P>
<P>pridialplan=national</P>
<P>prilocaldialplan=national</P>
<P>signalling=pri_cpe</P>
<P>rxwink=300 ; Atlas seems to use long (250ms) winks</P>
<P>usecallerid=yes</P>
<P>hidecallerid=no</P>
<P>callwaiting=yes</P>
<P>usecallingpres=yes</P>
<P>callwaitingcallerid=yes</P>
<P>threewaycalling=no</P>
<P>transfer=no</P>
<P>cancallforward=no</P>
<P>callreturn=no</P>
<P>echocancel=no</P>
<P>relaxdtmf=yes</P>
<P>rxgain=0.0</P>
<P>txgain=0.0</P>
<P>group=1</P>
<P>callgroup=1</P>
<P>pickupgroup=1</P>
<P>immediate=no</P>
<P>callerid=asreceived</P>
<P>amaflags=billing</P>
<P>busydetect=yes</P>
<P>busycount=8</P>
<P>;</P>
<P>channel=>1-15,17-31</P>
<P>channel=>32-46,48-62</P>
<P>channel=>63-77,79-93</P>
<P>;Arpu trunk</P>
<P>group=3</P>
<P>context=arpu</P>
<P>signalling=pri_cpe</P>
<P>channel=>94-108,110-124</P></FONT>
<P>where extensions.conf for server2 is:</P><FONT size=1>
<P>[arpuvoip]</P>
<P>;here I place a Zap call and the console shows (Unable to create a
channel of type ZAP)</P>
<P>exten=>_0XXXXXXXXX,1,Answer()</P>
<P>exten=>_0XXXXXXXXX,2,Dial(Zap/g1/${EXTEN})</P>
<P>exten=>_0XXXXXXXXX,3,Hangup()</P></FONT>
<P>&nbsp;</P>
<P>Any Ideas?</P>
<P>&nbsp;</P>
<P>Truely/</P>
<P>Joe</P>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT:
#a0c6e5 2px solid; MARGIN-RIGHT: 0px"><FONT style="FONT-SIZE:
11px; FONT-FAMILY: tahoma,sans-serif">
<HR color=#a0c6e5 SIZE=1>
From: <I>"Jerome SOUCANY"
<soucany@app-line.com></I><BR>Reply-To: <I>Asterisk
Users Mailing List - Non-Commercial
Discussion<asterisk-users@lists.digium.com></I><BR>To:
<I><asterisk-users@lists.digium.com></I><BR>Subject:
<I>[Asterisk-Users] No sound on 10% of incoming
calls</I><BR>Date: <I>Tue, 7 Feb 2006 11:03:49
+0100</I><BR>>Hello,<BR>><BR>>I have a problem
with Asterisk, on 10% of incoming calls the IP Phone ring<BR>>but I
don't hear the caller and the caller doesn't hear me (all IP
Phones<BR>>have the same problem).<BR>><BR>>This
problem appear also if the call is directly send to the second E1
of<BR>>the digium card who is connected to an
IVR.<BR>><BR>>It does not depand on the charge of the server
(I have the problem with only<BR>>one
call).<BR>><BR>>The configuration
:<BR>><BR>>PRI (France Telecom) 15 channels
<====> Asterisk <=====> IP Phone<BR>><BR>>* Server
:<BR>> - Dell power edge 1800SC<BR>> - 2 Ethernet cards (LAN +
VoIP LAN)<BR>> - Digium card : TE 405P<BR>> - Linux Mandriva
LE 2005 (10.2) :<BR>> Linux ASTERISK 2.6.11-12mdksmp #1 SMP i686
Intel(R) Xeon(TM) CPU<BR>>3.00GHz unknown GNU/Linux<BR>> -
Asterisk 1.2.4<BR>> - Zaptel 1.2.3<BR>> - Libpri
1.2.2<BR>><BR>>* IP Phone :<BR>> SNOM 320 (latest
firmware)<BR>><BR>>============================================<BR>>zaptel.conf<BR>><BR>>span=1,1,0,ccs,hdb3<BR>>span=2,1,0,ccs,hdb3,crc4,yellow<BR>>span=3,1,0,ccs,hdb3,crc4,yellow<BR>>span=4,1,0,ccs,hdb3,crc4,yellow<BR>><BR>>bchan
= 1-15, 17-31<BR>>dchan = 16<BR>>bchan =
32-46,48-62<BR>>dchan = 47<BR>>bchan =
63-77,79-93<BR>>dchan = 78<BR>>bchan =
94-108,110-124<BR>>dchan =
109<BR>><BR>>loadzone = fr<BR>>defaultzone =
fr<BR>><BR>>============================================<BR>><BR>>============================================<BR>>zapata.conf<BR>><BR>>[channels]<BR>>switchtype=euroisdn<BR>>pridialplan=national<BR>>signalling=pri_cpe<BR>>usecallerid=yes<BR>>hidecallerid=yes<BR>>usecallingpres=no<BR>>callwaiting=yes<BR>>callwaitingcallerid=yes<BR>>threewaycalling=yes<BR>>transfer=yes<BR>>cancallforward=yes<BR>>echocancel=yes<BR>>echocancelwhenbridged=yes<BR>>echotraining=yes<BR>>rxgain=0.0<BR>>txgain=-6.0<BR>><BR>>group=1<BR>>callgroup=1<BR>>pickupgroup=1<BR>><BR>>immediate=no<BR>>callprogress=yes<BR>><BR>>callerid=asreceived<BR>>group=1<BR>>context=from-pstn<BR>>signalling=pri_cpe<BR>>channel
=> 1-15 ;,17-31 => only 15 first channels on
PRI<BR>><BR>>group=2<BR>>context=from-ivr<BR>>signalling=pri_net<BR>>channel
=>
32-46,48-62<BR>><BR>>group=3<BR>>context=from-ivr-bis<BR>>signalling=pri_net<BR>>channel
=>
63-77,79-93<BR>><BR>>group=4<BR>>signalling=pri_net<BR>>channel
=>
94-108,110-124<BR>>============================================<BR>><BR>><BR>><BR>><BR>>Any
ideas
?<BR>><BR>><BR>><BR>>Regards<BR>><BR>>Jerome<BR>><BR>><BR>>_______________________________________________<BR>>--Bandwidth
and Colocation provided by Easynews.com
--<BR>><BR>>Asterisk-Users mailing list<BR>>To
UNSUBSCRIBE or update options visit:<BR>>
http://lists.digium.com/mailman/listinfo/asterisk-users<BR></FONT></BLOCKQUOTE></DIV></div><br
clear=all><hr>Free yourself from those irritating pop-up ads with 
<a href="http://g.msn.com/8HMAENCA/2752??PS=47575"
target="_top">MSN Premium.</a> Join now and get the first two
months FREE*</html>

Jerome SOUCANY

2006-Feb-09 04:09 UTC

head link

[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 : asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] De la part de Jerome
SOUCANY
Envoy? : mardi 7 f?vrier 2006 11:04
? : 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

Maybe Matching Threads

Search for more reasonably related threads

asterisk users - Feb 2006 - No sound on 10% of incoming calls

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

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

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

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

Maybe Matching Threads