<div class="gmail_quote"><span style="border-collapse:collapse;font-family:arial,sans-serif;font-size:13px">Hi <br><br></span><div><span style="border-collapse:collapse;font-family:arial,sans-serif;font-size:13px">I've spent two days trying to solve this issue but to no prevail and I'm hoping to get some help.</span></div>
<div><span style="border-collapse:collapse;font-family:arial,sans-serif;font-size:13px"><br>I've configured Asterisk as a SIP client, running on</span><span style="border-collapse:collapse;font-family:arial,sans-serif;font-size:13px"> OpenWRT on an embedded device with onboard FXS and ATA. Asterisk is connecting to</span><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;"> an external SIP provider on the Internet who in turn provides a PSTN gateway. I'm able to make calls to other SIP accounts registered on the same server who are outside my LAN. However, I can not make calls to any PSTN numbers. When trying to make PSTN calls it sounds like the person at the other end is immediately rejecting the call although I know this is not the case.</span></font><br>
<br><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;">Firstly, I'm absolutely sure that the PSTN gateway is working because I can make outbound PSTN calls with the same SIP account using other SIP clients (Empathy-SIP, SIPDroid) from the same LAN. However, when registering the same SIP account using Asterisk from OpenWRT all PSTN calls fail. Inbound calls from PSTN numbers also fail while calls from other SIP clients on the same server work fine. </span></font><span class="Apple-style-span" style="border-collapse: collapse; font-family: arial, sans-serif; ">Thus, I'm fairly confident the problem is with my Asterisk configuration.</span></div>
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<div><span style="border-collapse:collapse;font-family:arial,sans-serif;font-size:13px"><br></span><div><meta http-equiv="content-type" content="text/html; charset=utf-8"><div><font face="arial, sans-serif"><span style="border-collapse:collapse">The SIP accounts shows as registered in Asterisk. I've attached detailed error logs. The log files 'messages-pstn.log' shows the failed (PSTN) call and 'messages-voip.log' shows the successful (VOIP) call. Note that </span></font><font face="arial, sans-serif"><span style="border-collapse:collapse">I have replaced actual phone numbers and domain names with *** for anonymity.</span></font><br>
<font face="arial, sans-serif"><span style="border-collapse:collapse"><br>I suspect perhaps a codec issue, but I haven't been able to identify the actual problem. Any ideas that will help me towards solving this problem is greatly appreciated.<br>
</span></font><br><font face="arial, sans-serif"><span style="border-collapse:collapse">Regards,</span></font><br><font color="#888888"><font style="border-collapse:collapse;font-family:arial,sans-serif;font-size:13px" color="#888888">Helge</font></font></div>
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