<div dir="ltr"><div><div><div>Dear Amr,<br><br></div>In ss7.conf you have configured two ss7 links on link-l1 and link-l2.<br></div>for l3 and l4 if there is no signaling you need to use all channels like 1-31.<br></div>firstcic for l2 , l3 and l4 should 33, 65 and 97 respectively.<br>
<div><br><br>[link-l2]<br>linkset => siuc<br>channels => 1-15,17-31<br>schannel => 16<br>firstcic => 33<br>sltm => no<br>sls => 0<br><br>[link-l3]<br>linkset => siuc<br>channels => 1-31<br>schannel =><br>
firstcic => 65<br>sltm => no<br><br><div class="gmail_extra"><br>[link-l4]<br>linkset => siuc<br>channels => 1-31<br>schannel =><br>firstcic => 97<br>sltm => no<br><br><br></div><div class="gmail_extra">
Regards,<br></div><div class="gmail_extra">-Aashiqbhatti<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Jun 18, 2013 at 12:46 PM, Amr Adel <span dir="ltr"><<a href="mailto:amr132@gmail.com" target="_blank">amr132@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><p>Hello</p>
<p>
Kindly, I need your assist as I have installed chan_ss7 with asterisk and got
no voice on calls although asterisk is working fine With sip I am using</p>
<p>-sangoma A104de</p>
<p>-dahdi 2.6.6</p>
<p>-Asterisk 1.6.2</p>
<p>-chan_ss7 2.2.0</p>
<p>Also I get <b><u>NOTICE[22933]: mtp.c:1993 mtp_thread_main:
Full dahdi input buffer detected, incoming packets may have been lost on link
'l1' (count=65.)</u></b></p>
<p>At the beginning of an incoming call is it related to my
case?</p>
<p>You can find my config files attached </p></div>
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