<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Hi Group,<div><br></div><div>some advise needed. I am doing International call termination using Asterisk/Sangoma/Zaptel. My setup is as follows: </div><div><br></div><div>voippartners (internet)</div><div>|</div><div>sip</div><div>|</div><div>asteriskbox(sip only, high end server, cdr database, monitoring scripts, call routing based on destination)-------sip-----terminationbox1 (asterisk/libss7/sangoma)----ss7-----telecom1</div><div>|</div><div>sip</div><div>|</div><div>terminationboxN (Asterisk/libss7/sangoma)</div><div>|</div><div>ss7</div><div>|</div><div>TelecomN</div><div><br></div><div><br></div><div>I had many issues to get above setup up and running (from misconfigured zaptel to miss mapped CIC on the telecom switch), but now i have reached a point where I do about 40K call minutes per day on each of the </div><div>termination boxes and this project is finally starting to pay off. However i have noticed that of late I am getting the below warnings a lot:</div><div><br></div><div>WARNING[23136] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)</div><div><br></div><div>Am getting this warnings even when I have enough channel capacity on my E1s. During this time SIP channels still work and i dont see any complaints from the kernel. The call termination box gets the call in via sip but asterisk is not able to push it to the zaptel driver it seems. Yet when this starts happening it doesn't seem to affect all new calls. Any help is appreciated.</div><div><br></div><div>--</div><div>Markus</div><div><br></div><div><br></div><div><font class="Apple-style-span" color="#404040" face="'Lucida Grande'" size="2"><span class="Apple-style-span" style="font-size: 10px;"><br></span></font></div></body></html>