Steve B Posted December 17, 2013 Report Share Posted December 17, 2013 Hi guys, I have been having a bit of an issue when it comes to one way audio through an incoming vitelity trunk on a TLS connected phone. So I was playing around with the ports and pow, no audio at all in TLS, I can get two way audio in UDP. And when I say no audio, no audio on the trunk or when dialing voicemail. If the call makes it to my cellphone, no problem. I have factory reset the PBX and no luck with audio in TLS, am I going to have to wipe the system and start again? Also, no audio from webRTC now either. I have changed routers to make sure this wasn't the issue. Still could be but UDP calls to and from the Vodia PBX and another PBX work flawlessly. The router's nat profile is port-restricted cone on both routers. I put the phone in the DMZ and no help there. Thanks, Steve Quote Link to comment Share on other sites More sharing options...
Vodia support Posted December 18, 2013 Report Share Posted December 18, 2013 Hi, Steve can you check this article and run the PCAP setting on the extension level when you using TLS http://wiki.snomone.com/index.php?title=Extension_PCAP You can also activate the PCAP setting on the trunk level for inbound and outbound call. http://wiki.snomone.com/index.php?title=Trunk_PCAP_Generation Best regards Quote Link to comment Share on other sites More sharing options...
Steve B Posted December 18, 2013 Author Report Share Posted December 18, 2013 Hi, Steve can you check this article and run the PCAP setting on the extension level when you using TLS http://wiki.snomone.com/index.php?title=Extension_PCAP You can also activate the PCAP setting on the trunk level for inbound and outbound call. http://wiki.snomone.com/index.php?title=Trunk_PCAP_Generation Best regards I sent you a PM with the PCAP. Quote Link to comment Share on other sites More sharing options...
Vodia support Posted December 20, 2013 Report Share Posted December 20, 2013 Looks like the phone is remote? The phone send it's SDP with 192.168.1.182 and the PBX is sending it's SDP with 24.119.220.155. The PBX cannot send RTP traffic to a local IP. Check if you have a firewall set at the server level and on the router you can turn them off for testing services and try your the VM test. i this case if the phone was remote it would be sending SDP based on its Dynamic IP. INVITE sip:*97@voip.barrettsys.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-543qhk05dj5z;rport From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz To: <sip:*97@voip.barrettsys.com;user=phone> Call-ID: 52b0f3c5d925-dsalfsaxbvip CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:112@192.168.1.182:2066;transport=tls;line=ihfiyuz5>;reg-id=1 X-Serialnumber: 000413244C69 P-Key-Flags: keys="3" User-Agent: snom320/8.7.3.25 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons-snom320 Content-Type: application/sdp Content-Length: 502 v=0 o=root 584979493 584979493 IN IP4 192.168.1.182 s=call c=IN IP4 192.168.1.182 t=0 0 m=audio 50780 RTP/AVP 0 8 3 9 99 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:6Dwq4Jkn2h+aiF4YkKmZnmu9wwiBVgl01psqpScw a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-543qhk05dj5z;rport=2066;received=24.119.220.154 From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a Call-ID: 52b0f3c5d925-dsalfsaxbvip CSeq: 1 INVITE Content-Length: 0 SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-543qhk05dj5z;rport=2066;received=24.119.220.154 From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a Call-ID: 52b0f3c5d925-dsalfsaxbvip CSeq: 1 INVITE Contact: <sip:112@24.119.220.155:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Vodia-PBX/5.1.3 Content-Type: application/sdp Content-Length: 348 v=0 o=- 677813057 677813057 IN IP4 24.119.220.155 s=- c=IN IP4 24.119.220.155 t=0 0 m=audio 19170 RTP/AVP 9 0 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:KIovh3hlKqgGlVvULirklht4JS79+jXzdtOGGeMr a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ACK sip:112@24.119.220.155:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-xbj4q151mgkb;rport From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a Call-ID: 52b0f3c5d925-dsalfsaxbvip CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:112@192.168.1.182:2066;transport=tls;line=ihfiyuz5>;reg-id=1 Proxy-Require: buttons-snom320 Content-Length: 0 BYE sip:112@24.119.220.155:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-x0tq5z5vo8lp;rport From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a Call-ID: 52b0f3c5d925-dsalfsaxbvip CSeq: 2 BYE Max-Forwards: 70 Contact: <sip:112@192.168.1.182:2066;transport=tls;line=ihfiyuz5>;reg-id=1 User-Agent: snom320/8.7.3.25 RTP-RxStat: Total_Rx_Pkts=0,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=502,Tx_Pkts=502,Remote_Tx_Pkts=0 Proxy-Require: buttons-snom320 Content-Length: 0 SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-x0tq5z5vo8lp;rport=2066;received=24.119.220.154 From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a Call-ID: 52b0f3c5d925-dsalfsaxbvip CSeq: 2 BYE Contact: <sip:112@24.119.220.155:5061;transport=tls> User-Agent: Vodia-PBX/5.1.3 Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
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