Proxy-Authorization: Digest username="8868231", realm="asterisk", algorithm=MD5, uri="sip:13134453623@72.4.160.99", nonce="0088e93c", response="132cd5d45f1eec878b713ce46c9fb727" Call-ID: 0a3d3337f388d3d5@192.168.1.100 CSeq: 11506 INVITE User-Agent: Grandstream GXP2000 1.1.0.5 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 257 v=0 o=8868231 8000 8001 IN IP4 192.168.1.100 s=SIP Call c=IN IP4 209.19.88.240 t=0 0 m=audio 5004 RTP/AVP 4 8 18 3 0 a=sendrecv a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 --- (14 headers 13 lines) --- Using INVITE request as basis request - 0a3d3337f388d3d5@192.168.1.100 Sending to 209.19.88.240 : 5062 (NAT) Found user '8868231' Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 0 Peer audio RTP is at port 209.19.88.240:5004 Found description format G723 Found description format PCMA Found description format G729 Found description format GSM Found description format PCMU Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 13134453623 in a2billing (domain 72.4.160.99) Reliably Transmitting (NAT) to 209.19.88.240:5062: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bK48cf6ff105f00a6d;received=209.19.88.240 From: "3138875702" ;tag=ed140170ea65a119 To: ;tag=as76d864d4 Call-ID: 0a3d3337f388d3d5@192.168.1.100 CSeq: 11506 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- localhost*CLI> <-- SIP read from 209.19.88.240:5062: ACK sip:13134453623@72.4.160.99 SIP/2.0 Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bK48cf6ff105f00a6d From: "3138875702" ;tag=ed140170ea65a119 To: ;tag=as76d864d4 Contact: Proxy-Authorization: Digest username="8868231", realm="asterisk", algorithm=MD5, uri="sip:13134453623@72.4.160.99", nonce="0088e93c", response="0ca166d65e8fc8abdcda17ffe76a7d17" Call-ID: 0a3d3337f388d3d5@192.168.1.100 CSeq: 11506 ACK User-Agent: Grandstream GXP2000 1.1.0.5 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 --- (12 headers 0 lines) --- Destroying call '0a3d3337f388d3d5@192.168.1.100' localhost*CLI> sip no debug SIP Debugging Disabled localhost*CLI> \ localhost*CLI> ! [root@localhost ~]# clear [root@localhost ~]# clear [root@localhost ~]# asterisk -vr Asterisk 1.2.18, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.18 currently running on localhost (pid = 1971) Verbosity is at least 1 localhost*CLI> sip debug SIP Debugging enabled localhost*CLI> <-- SIP read from 209.19.88.240:5062: INVITE sip:13134453623@72.4.160.99 SIP/2.0 Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bKdeec3dbf1c7d07f4 From: "3138875702" ;tag=62251498eda64ee6 To: Contact: Supported: replaces, timer Call-ID: dc4721988f5a0589@192.168.1.100 CSeq: 24790 INVITE User-Agent: Grandstream GXP2000 1.1.0.5 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 257 v=0 o=8868231 8000 8000 IN IP4 192.168.1.100 s=SIP Call c=IN IP4 209.19.88.240 t=0 0 m=audio 5004 RTP/AVP 4 8 18 3 0 a=sendrecv a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 --- (13 headers 13 lines) --- Using INVITE request as basis request - dc4721988f5a0589@192.168.1.100 Sending to 209.19.88.240 : 5062 (non-NAT) Reliably Transmitting (NAT) to 209.19.88.240:5062: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bKdeec3dbf1c7d07f4;received=209.19.88.240 From: "3138875702" ;tag=62251498eda64ee6 To: ;tag=as64dbe8d0 Call-ID: dc4721988f5a0589@192.168.1.100 CSeq: 24790 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17ce0a18" Content-Length: 0 --- Scheduling destruction of call 'dc4721988f5a0589@192.168.1.100' in 15000 ms Found user '8868231' localhost*CLI> <-- SIP read from 209.19.88.240:5062: ACK sip:13134453623@72.4.160.99 SIP/2.0 Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bKdeec3dbf1c7d07f4 From: "3138875702" ;tag=62251498eda64ee6 To: ;tag=as64dbe8d0 Contact: Call-ID: dc4721988f5a0589@192.168.1.100 CSeq: 24790 ACK User-Agent: Grandstream GXP2000 1.1.0.5 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 --- (11 headers 0 lines) --- localhost*CLI> <-- SIP read from 209.19.88.240:5062: INVITE sip:13134453623@72.4.160.99 SIP/2.0 Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bK919f159680a0db66 From: "3138875702" ;tag=62251498eda64ee6 To: Contact: Supported: replaces, timer Proxy-Authorization: Digest username="8868231", realm="asterisk", algorithm=MD5, uri="sip:13134453623@72.4.160.99", nonce="17ce0a18", response="3a590a215509ad6cc64a7592852cf29a" Call-ID: dc4721988f5a0589@192.168.1.100 CSeq: 24791 INVITE User-Agent: Grandstream GXP2000 1.1.0.5 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 257 v=0 o=8868231 8000 8001 IN IP4 192.168.1.100 s=SIP Call c=IN IP4 209.19.88.240 t=0 0 m=audio 5004 RTP/AVP 4 8 18 3 0 a=sendrecv a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 --- (14 headers 13 lines) --- Using INVITE request as basis request - dc4721988f5a0589@192.168.1.100 Sending to 209.19.88.240 : 5062 (NAT) Found user '8868231' Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 0 Peer audio RTP is at port 209.19.88.240:5004 Found description format G723 Found description format PCMA Found description format G729 Found description format GSM Found description format PCMU Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 13134453623 in a2billing (domain 72.4.160.99) Reliably Transmitting (NAT) to 209.19.88.240:5062: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bK919f159680a0db66;received=209.19.88.240 From: "3138875702" ;tag=62251498eda64ee6 To: ;tag=as64dbe8d0 Call-ID: dc4721988f5a0589@192.168.1.100 CSeq: 24791 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- localhost*CLI> <-- SIP read from 209.19.88.240:5062: ACK sip:13134453623@72.4.160.99 SIP/2.0 Via: SIP/2.0/UDP 209.19.88.240:5062;branch=z9hG4bK919f159680a0db66 From: "3138875702" ;tag=62251498eda64ee6 To: ;tag=as64dbe8d0 Contact: Proxy-Authorization: Digest username="8868231", realm="asterisk", algorithm=MD5, uri="sip:13134453623@72.4.160.99", nonce="17ce0a18", response="90198d838bfb6b5df37ce3095815f5e3" Call-ID: dc4721988f5a0589@192.168.1.100 CSeq: 24791 ACK User-Agent: Grandstream GXP2000 1.1.0.5 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 --- (12 headers 0 lines) --- Destroying call 'dc4721988f5a0589@192.168.1.100' localhost*CLI> sip no debug SIP Debugging Disabled