<html><head></head><body bgcolor="#FFFFFF"><div>Is there any sip alg feature on the router? Try disabling it if there is.</div><div><br></div><div>Also does the router support upnp? If it does then FS can setup the port mapping that way.</div><div><br></div><div>Is it FS or the client that's behind nat?<br><br>Steve on iPhone</div><div><br>On 10 Jul 2011, at 21:22, Henk Oegema <<a href="mailto:henk@oegema.com">henk@oegema.com</a>> wrote:<br><br></div><div></div><blockquote type="cite"><div>My problem is even more serious than what I was facing earlier (see previous message)<br><br>Due to a defect router (too much rain inside :-( I changed to a new one. but now there is no voice on incoming calls. I can't hear them. but they can hear me.<br>
The other way around, when I call to a mobile (or landline) eveything is ok.<br>I was not facing this problem with the old modem.<br><br>My port forwarding is:<br>SIP 5060-5091 TCP&UDP<br>RTP 16384-32768<br>
STUN 3478-3479<br><br>What should I do ??<br><br><div class="gmail_quote">---------- Forwarded message ----------<br>From: <b class="gmail_sendername">Henk Oegema</b> <span dir="ltr"><<a href="mailto:henk@oegema.com"><a href="mailto:henk@oegema.com">henk@oegema.com</a></a>></span><br>
Date: Fri, Jul 1, 2011 at 6:29 PM<br>Subject: No voice channels to mobile.<br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org"><a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a></a><br><br><br><u></u>
<div>
I'm facing the following problem:<br>
<br>
External calls come in via an ITSP (Localphone) and then routed to internal extension 1000.<br>
That works OK.<br>
<br>
If there's no answer, then the call is routed outbound via ITSP (Powervoip) to my mobile +316xxxxxxxxx)<br>
In this case there is NO voice communication at all. <br>
(a direct call from 1000 to a mobile is OK)<br>
<br>
<br>
<extension name="1000"><br>
<condition field="destination_number" expression="^1000$"><br>
..........................................................................................<br>
..........................................................................................<br>
<action application="set" data="continue_on_fail=true"/><br>
<action application="set" data="hangup_after_bridge=true"/><br>
<action application="bridge" data="{originate_timeout=20}sofia/internal/1000%$${domain}"/> <----------audio OK<br>
<action application="bridge" data="{originate_timeout=20}sofia/gateway/powervoip/003163086xxxx"/> <------------NO audio both ways<br>
<action application="voicemail" data="default $${domain} 1000"/><br>
</condition><br>
</extension><br>
<br>
How do I approach this problem?<br>
<br>
Rgds,<br><font color="#888888">
Henk
</font></div>
</div><br>
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