Found Rtp Audio Format 18 | Audio-Digital.net

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Skype connect sip profile to Asterisk - Microsoft Community

https://answers.microsoft.com/en-us/skype/forum/all/skype-connect-sip-profile-to-asterisk/a492986d-209f-45b4-9e8c-5ef00a7e6934

Found peer 'skype' for '3727120253' from 63.209.144.201:5060 //call from peer skype == Using SIP RTP CoS mark 5. Found RTP audio format 8. Found RTP audio format 0. Found RTP audio format 18. Found RTP audio format 13. Found RTP audio format 101. Found audio description format PCMA for ID 8 //trying to use these codecs formats

SDP invite issues - Asterisk SIP - Asterisk Community

https://community.asterisk.org/t/sdp-invite-issues/77884

Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101

Inbound calls not connecting - Asterisk SIP - Asterisk ...

https://community.asterisk.org/t/inbound-calls-not-connecting/87704

== Using SIP RTP CoS mark 5 Got SDP version 1076374171 and unique parts [HuaweiSoftx3000 1076374170 IN IP4 provider_SIP_SignalIP] Found RTP audio format 108 Found RTP audio format 102 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 116 Found RTP audio format 3 …

SIP read from UDP:172.16.18.3:5060 --->

https://issues.asterisk.org/jira/secure/attachment/45155/mohdebug.pdf

c=IN IP4 172.16.18.130 t=0 0 a=sendrecv m=audio 10658 RTP/AVP 9 0 8 18 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 <-----> --- (15 headers 13 lines) --- Sending to 172.16.18.3:5060 (NAT)

Calls dropping after approx 6-8 seconds - General Help ...

https://community.freepbx.org/t/calls-dropping-after-approx-6-8-seconds/18618

m=audio 10020 RTP/AVP 8 2 18 9 110 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:2 G726-32/8000 a=ptime:30 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:30 a=rtpmap:110 telephone-event/8000 a=fmtp:110 0-15 <-----> e[KE2-SIP-01*CLI> e[0K— (16 headers 16 lines) — Sending to 5.5.5.5:5060 (NAT)

[SOLVED] SIP and OpenVPN - General Help - FreePBX ...

https://community.freepbx.org/t/solved-sip-and-openvpn/46178

Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 ... s=Asterisk PBX 13.18.3 c=IN IP4 192.168.10.4 t=0 0 m=audio 15258 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16

Found RTP audio format 103Found RTP audio format …

https://pastebin.com/zFepwdwN

Found RTP audio format 0Found RTP audio format …

https://pastebin.com/JXFTmPq1

Found RTP audio format 101 Found RTP audio format 100 Peer audio RTP is at port xxx.xxx.xxx.xxx:18756 Found audio description format telephone-event for ID 101 Found unknown media description format X-NSE for ID 100 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) ...

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