-- Executing Macro("SIP/801-8f04", "qu|othello|204|othello|18|0||Main Number||0") in new stack www*CLI> -- Executing NoOp("SIP/801-8f04", "") in new stack www*CLI> -- Executing Set("SIP/801-8f04", "_ALERT_INFO=AA") in new stack www*CLI> -- Executing GotoIf("SIP/801-8f04", "0?60") in new stack www*CLI> -- Executing Dial("SIP/801-8f04", "Sip/202") in new stack www*CLI> We're at 192.168.1.50 port 13738 www*CLI> Adding codec 0x4 (ulaw) to SDP www*CLI> Adding codec 0x2 (gsm) to SDP www*CLI> Adding codec 0x8 (alaw) to SDP www*CLI> Adding non-codec 0x1 (telephone-event) to SDP www*CLI> 14 headers, 12 lines www*CLI> Reliably Transmitting (no NAT) to 192.168.1.196:5060: INVITE sip:202@192.168.1.196 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK09833371;rport From: "CIDNAME" ;tag=as35a6e510 To: Contact: Call-ID: 443ee1d71a63a0766efa264771714d6b@192.168.1.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 19 Jan 2006 12:08:41 GMT Alert-Info: AA Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 31824 31824 IN IP4 192.168.1.50 s=session c=IN IP4 192.168.1.50 t=0 0 m=audio 13738 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- www*CLI> -- Called 202 www*CLI> <-- SIP read from 192.168.1.196:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK09833371;rport From: "CIDNAME" ;tag=as35a6e510 To: ;tag=168345F5-AEDCAF8A CSeq: 102 INVITE Call-ID: 443ee1d71a63a0766efa264771714d6b@192.168.1.50 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.3.0067 Content-Length: 0 --- (9 headers 0 lines)--- www*CLI> <-- SIP read from 192.168.1.196:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK09833371;rport From: "CIDNAME" ;tag=as35a6e510 To: ;tag=168345F5-AEDCAF8A CSeq: 102 INVITE Call-ID: 443ee1d71a63a0766efa264771714d6b@192.168.1.50 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.3.0067 Allow-Events: talk,hold,conference Content-Length: 0 --- (10 headers 0 lines)--- www*CLI> -- SIP/202-5449 is ringing www*CLI> <-- SIP read from 192.168.1.196:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK09833371;rport From: "CIDNAME" ;tag=as35a6e510 To: ;tag=168345F5-AEDCAF8A CSeq: 102 INVITE Call-ID: 443ee1d71a63a0766efa264771714d6b@192.168.1.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.3.0067 Content-Type: application/sdp Content-Length: 189 v=0 o=- 1137715594 1137715594 IN IP4 192.168.1.196 s=Polycom IP Phone c=IN IP4 192.168.1.196 t=0 0 m=audio 2244 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (11 headers 8 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.196:2244 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.196, port 5060 Transmitting (no NAT) to 192.168.1.196:5060: ACK sip:202@192.168.1.196 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK09da68e3;rport From: "CIDNAME" ;tag=as35a6e510 To: ;tag=168345F5-AEDCAF8A Contact: Call-ID: 443ee1d71a63a0766efa264771714d6b@192.168.1.50 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- www*CLI> -- SIP/202-5449 answered SIP/801-8f04 -- Attempting native bridge of SIP/801-8f04 and SIP/202-5449 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.196, port 5060 We're at 192.168.1.50 port 13738 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.1.196:5060: INVITE sip:202@192.168.1.196 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK2ff2f1a2;rport From: "CIDNAME" ;tag=as35a6e510 To: ;tag=168345F5-AEDCAF8A Contact: Call-ID: 443ee1d71a63a0766efa264771714d6b@192.168.1.50 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 218 v=0 o=root 31824 31825 IN IP4 192.168.1.101 s=session c=IN IP4 192.168.1.101 t=0 0 m=audio 16458 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- www*CLI> www*CLI> <-- SIP read from 192.168.1.196:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK2ff2f1a2;rport From: "CIDNAME" ;tag=as35a6e510 To: ;tag=168345F5-AEDCAF8A CSeq: 103 INVITE Call-ID: 443ee1d71a63a0766efa264771714d6b@192.168.1.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.3.0067 Content-Type: application/sdp Content-Length: 189 v=0 o=- 1137715595 1137715595 IN IP4 192.168.1.196 s=Polycom IP Phone c=IN IP4 192.168.1.196 t=0 0 m=audio 2244 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (11 headers 8 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.196:2244 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.196, port 5060 Transmitting (no NAT) to 192.168.1.196:5060: ACK sip:202@192.168.1.196 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK61eb716c;rport From: "CIDNAME" ;tag=as35a6e510 T www*CLI> o: ;tag=168345F5-AEDCAF8A Contact: Call-ID: 443ee1d71a63a0766efa264771714d6b@192.168.1.50 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- www*CLI> www*CLI> www*CLI> www*CLI> www*CLI> www*CLI> www*CLI> www*CLI> www*CLI> www*CLI> www*CLI> www*CLI> www*CLI> www*CLI> rtp debug ip www*CLI> RTP Debugging Enabled www*CLI> rtp no debug Got RTP packet from 192.168.1.101:16458 (type 0, seq 2189, ts 129954313, len 240) Sent RTP packet to 192.168.1.196:2244 (type 0, seq 14195, ts 97920, len 160) www*CLI> rtp no debug Got RTP packet from 192.168.1.101:16458 (type 0, seq 2190, ts 129954553, len 240) Sent RTP packet to 192.168.1.196:2244 (type 0, seq 14196, ts 98080, len 160) Sent RTP packet to 192.168.1.196:2244 (type 0, seq 14197, ts 98240, len 160) www*CLI> rtp no debug Got RTP packet from 192.168.1.101:16458 (type 0, seq 2191, ts 129954793, len 240) Sent RTP packet to 192.168.1.196:2244 (type 0, seq 14198, ts 98400, len 160) www*CLI> rtp no debug www*CLI> RTP Debugging Disabled www*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.196, port 5060 We're at 192.168.1.50 port 13738 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 192.168.1.196:5060: INVITE sip:202@192.168.1.196 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK3f3c4f4a;rport From: "CIDNAME" ;tag=as35a6e510 To: ;tag=168345F5-AEDCAF8A Contact: Call-ID: 443ee1d71a63a0766efa264771714d6b@192.168.1.50 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 263 v=0 o=root 31824 31826 IN IP4 192.168.1.50 s=session c=IN IP4 192.168.1.50 t=0 0 m=audio 13738 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- == Spawn extension (macro-qu, s, 4) exited non-zero on 'SIP/801-8f04' in macro 'qu' == Spawn extension (othellox, s, 1) exited non-zero on 'SIP/801-8f04' www*CLI> <-- SIP read from 192.168.1.196:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK3f3c4f4a;rport From: "CIDNAME" ;tag=as35a6e510 To: ;tag=168345F5-AEDCAF8A CSeq: 104 INVITE Call-ID: 443ee1d71a63a0766efa264771714d6b@192.168.1.50 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.3.0067 Content-Type: application/sdp Content-Length: 189 v=0 o=- 1137715596 1137715596 IN IP4 192.168.1.196 s=Polycom IP Phone c=IN IP4 192.168.1.196 t=0 0 m=audio 2244 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (11 headers 8 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.196:2244 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.196, port 5060 Transmitting (no NAT) to 192.168.1.196:5060: ACK sip:202@192.168.1.196 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK6213f421;rport From: "CIDNAME" ;tag=as35a6e510 T www*CLI> o: ;tag=168345F5-AEDCAF8A Contact: Call-ID: 443ee1d71a63a0766efa264771714d6b@192.168.1.50 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.196, port 5060 Reliably Transmitting (no NAT) to 192.168.1.196:5060: BYE sip:202@192.168.1.196 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK6b45d021;rport From: "CIDNAME" ;tag=as35a6e510 To: ;tag=168345F5-AEDCAF8A Contact: Call-ID: 443ee1d71a63a0766efa264771714d6b@192.168.1.50 CSeq: 105 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- www*CLI> <-- SIP read from 192.168.1.196:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK6b45d021;rport From: "CIDNAME" ;tag=as35a6e510 To: ;tag=168345F5-AEDCAF8A CSeq: 105 BYE Call-ID: 443ee1d71a63a0766efa264771714d6b@192.168.1.50 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.3.0067 Content-Length: 0 --- (9 headers 0 lines)--- Destroying call '443ee1d71a63a0766efa264771714d6b@192.168.1.50' www*CLI>