Anyone have any ideas on this?<br><br>-stephan<br><br><div class="gmail_quote">On Thu, Sep 30, 2010 at 10:30 AM, Stephan Ellis <span dir="ltr"><<a href="mailto:stephan.ellis@gmail.com">stephan.ellis@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Yes that's correct. Sorry, I should be more clear about my setup. I work for a rural telephone company. We have our asterisk box connected to a Siemens EWSD. I have my softphone connected directly to the asterisk box. The box I am calling is an asterisk box connected to a PRI from bell. I get no audio there. I also tested against a call center that has PRIs from bell and I get the same issue. Your guess is as good as mine as to what they are using.<br>
<br>To complicate matters, I also have my main phone system (asterisk) connected to a PRI on my EWSD. This is a completely different box, but connected to the same switch. When I call it from my ss7 box I get audio just fine. <br>
<br>We contacted our siemens guys about this and they say that when I call a remote PRI from our ss7 box, our switch is sending the asterisk box a pass along message, which we seem to be ignoring.<br><br>Hope that clears up my situation a little better. Thanks!<br>
<font color="#888888">
<br>-stephan</font><div><div></div><div class="h5"><br><br><div class="gmail_quote">On Thu, Sep 30, 2010 at 10:21 AM, Jean Cérien <span dir="ltr"><<a href="mailto:cerien.jean@gmail.com" target="_blank">cerien.jean@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
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<div>just to clarify... you have the following setup: ss7 -> asterisk -> sip -> softphone </div>
<div> </div>
<div>where is the PRI ?</div><div><div></div><div>
<div><br><br> </div>
<div class="gmail_quote">On Thu, Sep 30, 2010 at 11:04 AM, Stephan Ellis <span dir="ltr"><<a href="mailto:stephan.ellis@gmail.com" target="_blank">stephan.ellis@gmail.com</a>></span> wrote:<br>
<blockquote style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;" class="gmail_quote">I do see audio being received, but I don't hear it on my softphone. I see no TX at all. Interestingly, the guy on the pri I was calling said he could hear me. The remote pri is an asterisk box, so i set a DID on it to go straight to the echo test. While that system is playing demo-echo I see RX on my end, but when the actual echo test starts i see nothing.<br>
<font color="#888888"><br>-stephan</font>
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<div><br><br>
<div class="gmail_quote">On Thu, Sep 30, 2010 at 9:45 AM, Jean Cérien <span dir="ltr"><<a href="mailto:cerien.jean@gmail.com" target="_blank">cerien.jean@gmail.com</a>></span> wrote:<br>
<blockquote style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;" class="gmail_quote">
<div> </div>
<div>Hi</div>
<div> </div>
<div>Have you tried using dahdi_monitor to see if any sound is received ? </div>
<div> </div>
<div>Rgds,</div>
<div>J.<br><br></div>
<div class="gmail_quote">
<div>
<div></div>
<div>On Thu, Sep 30, 2010 at 10:15 AM, Stephan Ellis <span dir="ltr"><<a href="mailto:stephan.ellis@gmail.com" target="_blank">stephan.ellis@gmail.com</a>></span> wrote:<br></div></div>
<blockquote style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;" class="gmail_quote">
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<div>All,<br><br> I've got a problem on my SS7 implementation. When I originate calls across my SS7 link and the call lands on a PRI, I get no audio in either direction. The stack I am using is:<br><br>Asterisk 1.6.2.13<br>
DAHDI 2.4.0<br>libss7 1.0.2<br>libpri 1.4.11 (not sure if i need that, but thought it might be needed for ISUP stuff)<br>WANPIPE 3.5.15.4<br>Linux Kernel 2.6.18-194.11.4.el5 on Centos 5.5<br><br>The whole stack was hand compiled on the server (not from repos).<br>
<br>My dialplan is pretty simple, possibly too simple:<br><br>exten => _XXXXXXX,1,Dial(DAHDI/g0/${EXTEN})<br>exten => _XXXXXXX,n,Hangup()<br><br>My chan_dahdi.conf looks like this:<br><br>;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit<br>
;autogenrated on 2010-09-24<br>;Dahdi Channels Configurations<br>;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak<br><br>[trunkgroups]<br><br>[channels]<br>context=default<br>usecallerid=yes<br>hidecallerid=no<br>
callwaiting=yes<br>usecallingpres=yes<br>callwaitingcallerid=yes<br>threewaycalling=yes<br>transfer=yes<br>canpark=yes<br>cancallforward=yes<br>callreturn=yes<br>echocancel=no<br>echocancelwhenbridged=no<br>relaxdtmf=yes<br>
rxgain=0.0<br>txgain=0.0<br>group=1<br>callgroup=1<br>pickupgroup=1<br>immediate=no<br><br>ss7type=ansi<br>signalling=ss7<br>ss7_called_nai=dynamic<br>ss7_calling_nai=dynamic<br>ss7_internationalprefix=00<br>ss7_nationalprefix=0<br>
ss7_subscriberprefix=<br>ss7_unknownprefix=<br>networkindicator=national<br>explicitacm=yes<br>linkset=1<br>pointcode=1-1-1<br>defaultdpc=5-9-192<br>adjpointcode=5-9-192<br>group=0<br>cicbeginswith=1<br>channel=2-24<br>sigchan=1<br>
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