If you can get something like TBCT working, you can release the media back upstream somewhere along the line. Other than that, I suppose that you could jack the TTL with iptables. Tough call, because the audio delay after that many IP hops will probably trash the call. David From: VoiceOps [mailto:voiceops-bounces at voiceops.org] On Behalf Of Ujjval Karihaloo Sent: Thursday, May 09, 2013 21:30 To: voiceops at voiceops.org Subject: [VoiceOps] Hairpin Call from certain PBXs - No Audio issue Hi Team: Have you all seen issues with hairpin calls from PSTN--> ITSP (Broadsoft/ACME)--> SIP Trunk - which in turn forwards the call back out the SIP trunk to the PSTN and there is no Audio in either direction (Transfer via Auto-attendant OR Blind - Attended Xfer works as Media is anchored/reinitiated on PBX) Having worked on a few PBXs (Allworx, Zultys) that cause this issue on ACME/Broadsoft Sip trunk setup - we see RTP in both direction on our traces, but due to the pass through RTP nature of this call flow, the TTL get way down to 5-6 in some cases, and that is what I believe is the issue, and our upstream vendors like Level3, Bandwidth etc suggest the same. Other than implementing a RTP repeater of sorts, I don't see an alternative. Ideas appreciated Thx Ujjval This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.