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<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> </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(${CHANNEL})</P></FONT>
<P> </P>
<P>and server2 zapata.conf:</P><FONT size=1>
<P>[channels]</P>
<P>language=en</P>
<P>context=inbound</P>
<P>switchtype=euroisdn</P>
<P>pridialplan=national</P>
<P>prilocaldialplan=national</P>
<P>signalling=pri_cpe</P>
<P>rxwink=300 ; Atlas seems to use long (250ms) winks</P>
<P>usecallerid=yes</P>
<P>hidecallerid=no</P>
<P>callwaiting=yes</P>
<P>usecallingpres=yes</P>
<P>callwaitingcallerid=yes</P>
<P>threewaycalling=no</P>
<P>transfer=no</P>
<P>cancallforward=no</P>
<P>callreturn=no</P>
<P>echocancel=no</P>
<P>relaxdtmf=yes</P>
<P>rxgain=0.0</P>
<P>txgain=0.0</P>
<P>group=1</P>
<P>callgroup=1</P>
<P>pickupgroup=1</P>
<P>immediate=no</P>
<P>callerid=asreceived</P>
<P>amaflags=billing</P>
<P>busydetect=yes</P>
<P>busycount=8</P>
<P>;</P>
<P>channel=>1-15,17-31</P>
<P>channel=>32-46,48-62</P>
<P>channel=>63-77,79-93</P>
<P>;Arpu trunk</P>
<P>group=3</P>
<P>context=arpu</P>
<P>signalling=pri_cpe</P>
<P>channel=>94-108,110-124</P></FONT>
<P>where extensions.conf for server2 is:</P><FONT size=1>
<P>[arpuvoip]</P>
<P>;here I place a Zap call and the console shows (Unable to create a channel of type ZAP)</P>
<P>exten=>_0XXXXXXXXX,1,Answer()</P>
<P>exten=>_0XXXXXXXXX,2,Dial(Zap/g1/${EXTEN})</P>
<P>exten=>_0XXXXXXXXX,3,Hangup()</P></FONT>
<P> </P>
<P>Any Ideas?</P>
<P> </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>