asterdev1*CLI> <-- SIP read from 69.67.248.51:5060: INVITE sip:3155798378;npdi=yes@66.218.0.21 SIP/2.0 Max-Forwards: 69 Session-Expires: 3600;Refresher=uac Supported: timer To: From: ;tag=gK0e7cbc6b Call-ID: 5583493-3364989205-384142@nextone2.usadatanet.net CSeq: 1 INVITE Via: SIP/2.0/UDP 69.67.248.51:5060;branch=a391ee313d01432339da76e64a427321 Contact: Content-Type: application/sdp Content-Length: 246 v=0 o=NexTone-MSW 1234 8089 IN IP4 69.67.248.5 s=sip call c=IN IP4 69.67.248.5 t=0 0 m=audio 47218 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:10 Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3357 parse_request: Header 0: INVITE sip:3155798378;npdi=yes@66.218.0.21 SIP/2.0 (50) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3357 parse_request: Header 1: Max-Forwards: 69 (16) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3357 parse_request: Header 2: Session-Expires: 3600;Refresher=uac (35) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3357 parse_request: Header 3: Supported: timer (16) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3357 parse_request: Header 4: To: (38) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3357 parse_request: Header 5: From: ;tag=gK0e7cbc6b (96) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3357 parse_request: Header 6: Call-ID: 5583493-3364989205-384142@nextone2.usadatanet.net (58) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3357 parse_request: Header 7: CSeq: 1 INVITE (14) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3357 parse_request: Header 8: Via: SIP/2.0/UDP 69.67.248.51:5060;branch=a391ee313d01432339da76e64a427321 (74) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3357 parse_request: Header 9: Contact: (86) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3357 parse_request: Header 10: Content-Type: application/sdp (29) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3357 parse_request: Header 11: Content-Length: 246 (19) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3357 parse_request: Header 12: (0) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3389 parse_request: Line: v=0 (3) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3389 parse_request: Line: o=NexTone-MSW 1234 8089 IN IP4 69.67.248.5 (42) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3389 parse_request: Line: s=sip call (10) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3389 parse_request: Line: c=IN IP4 69.67.248.5 (20) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3389 parse_request: Line: t=0 0 (5) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3389 parse_request: Line: m=audio 47218 RTP/AVP 0 18 101 (30) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3389 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3389 parse_request: Line: a=rtpmap:18 G729/8000 (21) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3389 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3389 parse_request: Line: a=fmtp:101 0-15 (15) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3389 parse_request: Line: a=sendrecv (10) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3389 parse_request: Line: a=maxptime:10 (13) --- (12 headers 12 lines)--- Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3141 sip_alloc: Allocating new SIP dialog for 5583493-3364989205-384142@nextone2.usadatanet.net - INVITE (With RTP) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:11139 handle_request: **** Received INVITE (5) - Command in SIP INVITE Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:1004 parse_sip_options: Begin: parsing SIP "Supported: timer" Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:1016 parse_sip_options: Found SIP option: -timer- Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:1022 parse_sip_options: Matched SIP option: timer Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:1033 parse_sip_options: * SIP extension value: 4 for call 5583493-3364989205-384142@nextone2.usadatanet.net Using INVITE request as basis request - 5583493-3364989205-384142@nextone2.usadatanet.net Sending to 69.67.248.51 : 5060 (non-NAT) Found peer 'nextone' Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:7236 check_user_full: Setting NAT on RTP to 0 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 69.67.248.5:47218 Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:3598 process_sdp: Peer audio RTP is at port 69.67.248.5:47218 Found description format PCMU Found description format G729 Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:10496 handle_request_invite: Checking SIP call limits for device Aug 19 12:58:37 DEBUG[21208]: chan_sip.c:2206 update_call_counter: Updating call counter for incoming call Looking for s in default (domain 3155798378)