Phone Seesion Expires = 180s Asterisk Seesion Expires = 300s astrid-test*CLI> sip set debug peer myphone SIP Debugging Enabled for IP: 192.168.100.1 <--- SIP read from UDP:192.168.100.1:5060 ---> INVITE sip:8512@192.168.100.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bKdb36442361a716ae From: ;tag=ec7e04e91bc954ac To: Contact: Supported: replaces, timer, path X-Grandstream-PBX: true P-Early-Media: Supported Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 51338 INVITE User-Agent: Grandstream GXP2010 1.2.5.3 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 364 v=0 o=myphone 8000 8000 IN IP4 192.168.100.1 s=SIP Call c=IN IP4 192.168.100.1 t=0 0 m=audio 5068 RTP/AVP 8 0 3 18 2 4 9 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (15 headers 17 lines) --- ending to 192.168.100.1:5060 (NAT) Using INVITE request as basis request - 95f040f05fc678e7@192.168.100.1 Found peer 'myphone' for 'myphone' from 192.168.100.1:5060 <--- Reliably Transmitting (no NAT) to 192.168.100.1:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bKdb36442361a716ae;received=192.168.100.1 From: ;tag=ec7e04e91bc954ac To: ;tag=as4300a9f7 Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 51338 INVITE Server: Asterisk PBX SVN-branch-1.8-r385916 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="01850fcf" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '95f040f05fc678e7@192.168.100.1' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.100.1:5060 ---> ACK sip:8512@192.168.100.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bKdb36442361a716ae From: ;tag=ec7e04e91bc954ac To: ;tag=as4300a9f7 Contact: Supported: path X-Grandstream-PBX: true Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 51338 ACK User-Agent: Grandstream GXP2010 1.2.5.3 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> --- (13 headers 0 lines) --- <--- SIP read from UDP:192.168.100.1:5060 ---> INVITE sip:8512@192.168.100.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK8e5660a6cd338b9e From: ;tag=ec7e04e91bc954ac To: Contact: Supported: replaces, timer, path X-Grandstream-PBX: true P-Early-Media: Supported Authorization: Digest username="myphone", realm="asterisk", algorithm=MD5, uri="sip:8512@192.168.100.254", nonce="01850fcf", response="48daaac877e0abb0b4be4ab4d21734a6" Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 51339 INVITE User-Agent: Grandstream GXP2010 1.2.5.3 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 364 v=0 o=myphone 8000 8001 IN IP4 192.168.100.1 s=SIP Call c=IN IP4 192.168.100.1 t=0 0 m=audio 5068 RTP/AVP 8 0 3 18 2 4 9 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (16 headers 17 lines) --- Sending to 192.168.100.1:5060 (no NAT) Using INVITE request as basis request - 95f040f05fc678e7@192.168.100.1 Found peer 'myphone' for 'myphone' from 192.168.100.1:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 9 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 2 Found audio description format G723 for ID 4 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.100.1:5068 Looking for 8512 in phones (domain 192.168.100.254) list_route: hop: <--- Transmitting (no NAT) to 192.168.100.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK8e5660a6cd338b9e;received=192.168.100.1 From: ;tag=ec7e04e91bc954ac To: Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 51339 INVITE Server: Asterisk PBX SVN-branch-1.8-r385916 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 300;refresher=uas Contact: Content-Length: 0 <------------> -- Executing [8512@phones:1] Dial("SIP/myphone-00000000", "IAX2/astrid2/8512") in new stack -- Called IAX2/astrid2/8512 -- Call accepted by 192.168.5.40 (format alaw) -- Format for call is alaw -- IAX2/astrid2-17576 is proceeding passing it to SIP/myphone-00000000 <--- Transmitting (no NAT) to 192.168.100.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK8e5660a6cd338b9e;received=192.168.100.1 From: ;tag=ec7e04e91bc954ac To: Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 51339 INVITE Server: Asterisk PBX SVN-branch-1.8-r385916 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 300;refresher=uas Contact: Content-Length: 0 <------------> -- IAX2/astrid2-17576 is ringing <--- Transmitting (no NAT) to 192.168.100.1:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK8e5660a6cd338b9e;received=192.168.100.1 From: ;tag=ec7e04e91bc954ac To: ;tag=as24a74542 Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 51339 INVITE Server: Asterisk PBX SVN-branch-1.8-r385916 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 300;refresher=uas Contact: Content-Length: 0 <------------> -- IAX2/astrid2-17576 is ringing -- IAX2/astrid2-17576 stopped sounds -- IAX2/astrid2-17576 answered SIP/myphone-00000000 Audio is at 19754 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.100.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK8e5660a6cd338b9e;received=192.168.100.1 From: ;tag=ec7e04e91bc954ac To: ;tag=as24a74542 Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 51339 INVITE Server: Asterisk PBX SVN-branch-1.8-r385916 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 300;refresher=uas <<<<<<<<<<<<<<<<<<<<<<<< This is new since ASTERISK-20787 <<<<<<<<<<<<<<<<<<<< Contact: Content-Type: application/sdp Require: timer Content-Length: 298 v=0 o=root 1575286713 1575286713 IN IP4 192.168.100.254 s=Asterisk PBX SVN-branch-1.8-r385916 c=IN IP4 192.168.100.254 t=0 0 m=audio 19754 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:192.168.100.1:5060 ---> ACK sip:8512@192.168.100.254:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bKe6fda982bd697e55 From: ;tag=ec7e04e91bc954ac To: ;tag=as24a74542 Contact: Supported: path X-Grandstream-PBX: true Authorization: Digest username="myphone", realm="asterisk", algorithm=MD5, uri="sip:8512@192.168.100.254", nonce="01850fcf", response="48daaac877e0abb0b4be4ab4d21734a6" Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 51339 ACK User-Agent: Grandstream GXP2010 1.2.5.3 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> --- (14 headers 0 lines) --- astrid-test*CLI> core set verbose 0 Verbosity is now OFF astrid-test*CLI> astrid-test*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.100.1:5060 Audio is at 19754 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.100.1:5060: INVITE sip:myphone@192.168.100.1:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK76d39f6b Max-Forwards: 70 From: ;tag=as24a74542 To: ;tag=ec7e04e91bc954ac Contact: Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r385916 Session-Expires: 300;refresher=uac <<<<<<<<<<<<<<<<<<<<<<<< re-invite from asterisk 300 seconds <<<<<<<<<<<<<<<<<<<< Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 298 v=0 o=root 1575286713 1575286713 IN IP4 192.168.100.254 s=Asterisk PBX SVN-branch-1.8-r385916 c=IN IP4 192.168.100.254 t=0 0 m=audio 19754 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.100.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK76d39f6b From: ;tag=as24a74542 To: ;tag=ec7e04e91bc954ac Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2010 1.2.5.3 Session-Expires: 180;refresher=uac <<<<<<<<<<<<<<<<<<<<<<<< response from phone 180 seconds <<<<<<<<<<<<<<<<<<<< Min-SE: 180 Require: timer Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 216 v=0 o=myphone 8000 8002 IN IP4 192.168.100.1 s=SIP Call c=IN IP4 192.168.100.1 t=0 0 m=audio 5068 RTP/AVP 8 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (15 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.100.1:5068 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.100.1:5060 Transmitting (no NAT) to 192.168.100.1:5060: ACK sip:myphone@192.168.100.1:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK1f00ee9a Max-Forwards: 70 From: ;tag=as24a74542 To: ;tag=ec7e04e91bc954ac Contact: Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r385916 Content-Length: 0 --- <--- SIP read from UDP:192.168.100.1:5060 ---> BYE sip:8512@192.168.100.254:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK866529a6808641d6 From: ;tag=ec7e04e91bc954ac To: ;tag=as24a74542 Supported: path X-Grandstream-PBX: true Authorization: Digest username="myphone", realm="asterisk", algorithm=MD5, uri="sip:8512@192.168.100.254:5060", nonce="01850fcf", response="84215ae13a03744c8243d9bfcb429244" Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 51340 BYE User-Agent: Grandstream GXP2010 1.2.5.3 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Reason: SIP ;text="Session-timer expired" <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< GAME OVER (180-32) seconds later <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 192.168.100.1:5060 (no NAT) Scheduling destruction of SIP dialog '95f040f05fc678e7@192.168.100.1' in 32000 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.100.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.1:5060;branch=z9hG4bK866529a6808641d6;received=192.168.100.1 From: ;tag=ec7e04e91bc954ac To: ;tag=as24a74542 Call-ID: 95f040f05fc678e7@192.168.100.1 CSeq: 51340 BYE Server: Asterisk PBX SVN-branch-1.8-r385916 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------>