Thanks for firewall pointers, but all appears in order in such regards. This isn't my site's first sip rodeo. We Have two working PBXs -- asterisk and FreeSWITCH -- up as well as an AS node and soft phones.

What has changed since first posting is that it now "just works" ,,, sometimes. It almost seems like a capacity thing on the allstarlink side. Now, I can dial, connect, no audio ... Hang up, redial, reconnect and it to works. Might be an early vs late negotiation skew.

Not sure why,,, if someone on the asl side has access to their sip logs, it might shed some light.

Thanks,
—————

On 2012-11-10, at 18:44, Scott Weis <kb2ear@kb2ear.net> wrote:

Make sure you have the RTP ports forwarded in on your router.

From: app_rpt-users-bounces@ohnosec.org [mailto:app_rpt-users-bounces@ohnosec.org] On Behalf Of Scott
Sent: Saturday, November 10, 2012 5:45 PM
To: App_rpt-users@ohnosec.org
Subject: [App_rpt-users] sip.allstarlink.org sip session no-audio

When trying to connect from a sip-phone to the portal, sip.allstarlink.org using,

sip://1:allstar@sip.allstarlink.org:5060/0
-or-
sip://1:allstar@sip.allstarlink.org:5060/<node_number>
-or-
sip://0@sip.allstarlink.org

It connects, but there's no audio. No errors. PCMU@8000 correctly negotiated. I've tried two SIP-phone clients and a FreeSWITCH user-agent with the same negative outcomes. I've tried variations on the connect stream with same negative results. The BRIA soft-phone even handshakes GSM but still no audio.

Is there something I'm missing?

Thanks,