SIP Trunk (No audio)

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SIP Trunk (No audio)

Postby sombragris » Fri Jan 05, 2007 7:37 am


I've configured SIP for voipbuster inside providers.conf, than Ive configured a macro for trunkdial, and the corresponding extensions. When I dial from my softphone it keeps giving me the dial tones (as if it was dialing over and over again), but it finally gets through voipbuster and my home phone rings. When I pick up the line the dtmf tone of the softphone stop (connected) but there's no audio at all (both ways are muted).

I've allowed all traffic from my asterisk server to the internet on the firewall.

¿Any ideas?
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Postby leemason » Fri Jan 05, 2007 10:55 am

Have you set the SIP trunk up with nat=yes. If you haven't already done so you will need to setup the externip and localnet values in SIP.CONF. There should be some notes in SIP.CONF for this.
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Postby bkruse » Fri Jan 05, 2007 2:10 pm

You right, your rtp traffic is being blocked (ports 10000-20000), but your sip session on port 5060 is being let through......thus establishing the session but no audio.

nat=yes rocks.

configure it in sip.conf as the person above has said.

Just until we add that option in the GUI at least ;]
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