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- <------------>
- set_destination: Parsing <sip:+390633040237@87.248.56.100> for address/port to send to
- set_destination: set destination to 87.248.56.100:5060
- Audio is at 10028
- Adding codec 100008 (g729) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (no NAT) to 87.248.56.100:5060:
- INVITE sip:+390633040237@87.248.56.100 SIP/2.0
- Via: SIP/2.0/UDP 87.248.56.102:5060;branch=z9hG4bK2ab7dbb7
- Max-Forwards: 70
- From: <sip:0287250030@87.248.56.102>;tag=as77930702
- To: <sip:+390633040237@87.248.56.100;user=phone>;tag=3f8d7cfe0000be8c
- Contact: <sip:0287250030@87.248.56.102:5060>
- Call-ID: 04eca337110c4dbc1c8f2fd439bcc2d2@87.248.56.102:5060
- CSeq: 105 INVITE
- User-Agent: Asterisk PBX 12.4.0
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- Supported: replaces, timer
- X-asterisk-Info: SIP re-invite (External RTP bridge)
- Content-Type: application/sdp
- Content-Length: 276
- v=0
- o=root 1953170441 1953170444 IN IP4 87.248.56.102
- s=Asterisk PBX 12.4.0
- c=IN IP4 87.248.56.102
- t=0 0
- m=audio 10028 RTP/AVP 18 101
- a=rtpmap:18 G729/8000
- a=fmtp:18 annexb=no
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=maxptime:230
- a=sendrecv
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