Sip read: INVITE sip:*2@10.131.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacASHFEti Max-Forwards: 70 From: ;tag=1c85381269 To: Call-ID: 791917431CQPV@10.131.2.1 CSeq: 1 INVITE Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006 Content-Type: application/sdp Content-Length: 242 v=0 o=AudiocodesGW 623122 509909 IN IP4 10.131.2.1 s=Phone-Call c=IN IP4 10.131.2.1 t=0 0 m=audio 6000 RTP/AVP 8 0 96 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv 13 headers, 12 lines Using latest request as basis request Sending to 10.131.2.1 : 5060 (non-NAT) Oct 5 09:07:52 DEBUG[6708]: chan_sip.c:5456 check_user_full: Setting NAT on RTP to 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacASHFEti From: ;tag=1c85381269 To: ;tag=as1bf838c9 Call-ID: 791917431CQPV@10.131.2.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="2ae094d7" Content-Length: 0 to 10.131.2.1:5060 Scheduling destruction of call '791917431CQPV@10.131.2.1' in 15000 ms Found user '070001' localhost*CLI> Sip read: ACK sip:*2@10.131.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacASHFEti Max-Forwards: 70 From: ;tag=1c85381269 To: ;tag=as1bf838c9 Call-ID: 791917431CQPV@10.131.2.1 CSeq: 1 ACK Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006 Content-Length: 0 localhost*CLI> 12 headers, 0 lines Oct 5 09:07:52 DEBUG[6708]: chan_sip.c:841 __sip_ack: Stopping retransmission on '791917431CQPV@10.131.2.1' of Response 1: Found localhost*CLI> Sip read: INVITE sip:*2@10.131.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKaczZXMLAy Max-Forwards: 70 From: ;tag=1c85381269 To: Call-ID: 791917431CQPV@10.131.2.1 CSeq: 2 INVITE Proxy-Authorization: Digest username="070001",realm="asterisk", nonce="2ae094d7",uri="sip:*2@10.131.0.1",algorithm=MD5,response="d4a5b7841d627b24279644779bedc6a5" Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006 Content-Type: application/sdp Content-Length: 242 v=0 o=AudiocodesGW 623122 509909 IN IP4 10.131.2.1 s=Phone-Call c=IN IP4 10.131.2.1 t=0 0 m=audio 6000 RTP/AVP 8 0 96 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv 14 headers, 12 lines Using latest request as basis request Sending to 10.131.2.1 : 5060 (non-NAT) Oct 5 09:07:52 DEBUG[6708]: chan_sip.c:5456 check_user_full: Setting NAT on RTP to 0 Found user '070001' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 96 Peer audio RTP is at port 10.131.2.1:6000 Oct 5 09:07:52 DEBUG[6708]: chan_sip.c:2729 process_sdp: Peer audio RTP is at port 10.131.2.1:6000 Found description format pcma Found description format pcmu Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Oct 5 09:07:52 DEBUG[6708]: chan_sip.c:7354 handle_request: Check for res for 070001 Oct 5 09:07:52 DEBUG[6708]: chan_sip.c:1623 update_user_counter: Call from user '070001' is 1 out of 0 Looking for *2 in telelet Oct 5 09:07:52 WARNING[6708]: pbx.c:798 pbx_find_extension: No such switch 'realtime' Oct 5 09:07:52 DEBUG[6708]: chan_sip.c:4643 build_route: build_route: Contact hop: list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKaczZXMLAy From: ;tag=1c85381269 To: Call-ID: 791917431CQPV@10.131.2.1 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.131.2.1:5060 Oct 5 09:07:52 WARNING[6718]: pbx.c:798 pbx_find_extension: No such switch 'realtime' Oct 5 09:07:52 WARNING[6718]: pbx.c:798 pbx_find_extension: No such switch 'realtime' Oct 5 09:07:52 WARNING[6718]: pbx.c:798 pbx_find_extension: No such switch 'realtime' -- Executing VoiceMailMain("SIP/070001-a908", "s070001") in new stack We're at 10.131.0.1 port 15774 Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKaczZXMLAy From: ;tag=1c85381269 To: ;tag=as5f25037b Call-ID: 791917431CQPV@10.131.2.1 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 254 v=0 o=root 6718 6718 IN IP4 10.131.0.1 s=session c=IN IP4 10.131.0.1 t=0 0 m=audio 15774 RTP/AVP 3 0 8 96 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - to 10.131.2.1:5060 Oct 5 09:07:52 DEBUG[6718]: rtp.c:1195 ast_rtp_write: Ooh, format changed from unknown to ulaw Oct 5 09:07:52 DEBUG[6718]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'vm-login' (language 'en') localhost*CLI> Sip read: ACK sip:*2@10.131.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacbUaPONB Max-Forwards: 70 From: ;tag=1c85381269 To: ;tag=as5f25037b Call-ID: 791917431CQPV@10.131.2.1 CSeq: 2 ACK Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006 Content-Length: 0 12 headers, 0 lines Oct 5 09:07:52 DEBUG[6708]: chan_sip.c:841 __sip_ack: Stopping retransmission on '791917431CQPV@10.131.2.1' of Response 2: Found Oct 5 09:07:54 DEBUG[6718]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Oct 5 09:07:54 DEBUG[6718]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals localhost*CLI> Sip read: BYE sip:*2@10.131.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacOgVTaHN Max-Forwards: 70 From: ;tag=1c85381269 To: ;tag=as5f25037b Call-ID: 791917431CQPV@10.131.2.1 CSeq: 3 BYE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006 Content-Length: 0 12 headers, 0 lines Sending to 10.131.2.1 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKacOgVTaHN From: ;tag=1c85381269 To: ;tag=as5f25037b Call-ID: 791917431CQPV@10.131.2.1 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.131.2.1:5060 Oct 5 09:07:55 WARNING[6718]: app_voicemail.c:3352 vm_execmain: Couldn't read username == Spawn extension (telelet, *2, 1) exited non-zero on 'SIP/070001-a908' Oct 5 09:07:55 WARNING[6718]: pbx.c:798 pbx_find_extension: No such switch 'realtime' Oct 5 09:07:56 DEBUG[6718]: chan_sip.c:1726 sip_hangup: update_user_counter(070001) - decrement inUse counter Destroying call '791917431CQPV@10.131.2.1'