<--- SIP read from UDP:10.211.3.12:5060 ---> INVITE sip:0885250001@172.16.1.99:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 From: ;tag=ymmyzz4w-CC-33 To: CSeq: 1 INVITE Max-Forwards: 68 Contact: Min-SE: 90 Session-Expires: 1800 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER User-Agent: Huawei SoftX3000 V300R010 Supported: 100rel,timer Content-Length: 331 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 6370380 6370380 IN IP4 10.211.3.12 s=Sip Call c=IN IP4 10.211.3.12 t=0 0 m=audio 65280 RTP/AVP 8 0 18 4 105 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:105 CLEARMODE/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> --- (15 headers 14 lines) --- Sending to 10.211.3.12:5060 (no NAT) Sending to 10.211.3.12:5060 (no NAT) Using INVITE request as basis request - sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 Found peer 'pldt-in' for '09175348234' from 10.211.3.12:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 105 Found RTP audio format 127 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found unknown media description format CLEARMODE for ID 105 Found audio description format telephone-event for ID 127 Capabilities: us - (alaw|ulaw|g729|gsm), peer - audio=(ulaw|g723|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) > 0x7efc2c082760 -- Strict RTP learning after remote address set to: 10.211.3.12:65280 Peer audio RTP is at port 10.211.3.12:65280 Looking for 0885250001 in inbound (domain 172.16.1.99) sip_route_dump: route/path hop: <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> -- Executing [0885250001@inbound:1] Answer("SIP/pldt-in-00000069", "") in new stack Audio is at 11574 Adding codec alaw to SDP Adding codec ulaw to SDP Adding codec g729 to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296537 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv <------------> <--- SIP read from UDP:10.211.3.12:5060 ---> INVITE sip:0885250001@172.16.1.99:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 From: ;tag=ymmyzz4w-CC-33 To: CSeq: 1 INVITE Max-Forwards: 68 Contact: Min-SE: 90 Session-Expires: 1800 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER User-Agent: Huawei SoftX3000 V300R010 Supported: 100rel,timer Content-Length: 331 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 6370380 6370380 IN IP4 10.211.3.12 s=Sip Call c=IN IP4 10.211.3.12 t=0 0 m=audio 65280 RTP/AVP 8 0 18 4 105 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:105 CLEARMODE/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> --- (15 headers 14 lines) --- Ignoring this INVITE request <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> Audio is at 11574 Adding codec alaw to SDP Adding codec ulaw to SDP Adding codec g729 to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296538 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv <------------> Retransmitting #1 (no NAT) to 10.211.3.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296537 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv --- -- Executing [0885250001@inbound:2] Dial("SIP/pldt-in-00000069", "SIP/103,30,r") in new stack == Using SIP RTP CoS mark 5 Audio is at 19408 Adding codec gsm to SDP Adding codec ulaw to SDP Adding codec g723 to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.20.0.15:5060: INVITE sip:103@10.20.0.15:5060 SIP/2.0 Via: SIP/2.0/UDP 10.20.0.3:5060;branch=z9hG4bK1f974a16;rport Max-Forwards: 70 From: ;tag=as21d004ab To: Contact: Call-ID: 3066d6f60ca6277904efb1704ff47764@10.20.0.3:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.6.1 Date: Sun, 05 Jan 2020 13:55:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "09175348234" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 296 v=0 o=root 36783387 36783387 IN IP4 10.20.0.3 s=Asterisk PBX 16.6.1 c=IN IP4 10.20.0.3 t=0 0 m=audio 19408 RTP/AVP 3 0 4 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- -- Called SIP/103 <--- SIP read from UDP:10.20.0.15:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.20.0.3:5060;branch=z9hG4bK1f974a16;rport=5060 From: ;tag=as21d004ab To: Call-ID: 3066d6f60ca6277904efb1704ff47764@10.20.0.3:5060 CSeq: 102 INVITE Supported: replaces, path, timer User-Agent: Grandstream GXP2170 1.0.9.69 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:10.20.0.15:5060 ---> SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.20.0.3:5060;branch=z9hG4bK1f974a16;rport=5060 From: ;tag=as21d004ab To: ;tag=1462030114 Call-ID: 3066d6f60ca6277904efb1704ff47764@10.20.0.3:5060 CSeq: 102 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP2170 1.0.9.69 Diversion: ;reason=unconditional Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- -- Got SIP response 302 "Moved Temporarily" back from 10.20.0.15:5060 RDNIS for this call is 103 (reason unconditional) Transmitting (NAT) to 10.20.0.15:5060: ACK sip:103@10.20.0.15:5060 SIP/2.0 Via: SIP/2.0/UDP 10.20.0.3:5060;branch=z9hG4bK1f974a16;rport Max-Forwards: 70 From: ;tag=as21d004ab To: ;tag=1462030114 Contact: Call-ID: 3066d6f60ca6277904efb1704ff47764@10.20.0.3:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.6.1 Content-Length: 0 --- -- Now forwarding SIP/pldt-in-00000069 to 'Local/100@admin' (thanks to SIP/103-0000006a) [Jan 5 13:55:55] NOTICE[2217][C-00000034]: app_dial.c:1006 do_forward: Not accepting call completion offers from call-forward recipient Local/100@admin-00000005;1 -- Executing [100@admin:1] Set("Local/100@admin-00000005;2", "rno=09175348234") in new stack -- Executing [100@admin:2] Set("Local/100@admin-00000005;2", "calltype=Internal") in new stack > Found no rows [SELECT name_display FROM sippeers WHERE room_number = 09175348234] -- Executing [100@admin:3] Set("Local/100@admin-00000005;2", "CALLERID(name)=") in new stack -- Executing [100@admin:4] Dial("Local/100@admin-00000005;2", "SIP/100,30,r") in new stack == Using SIP RTP CoS mark 5 Really destroying SIP dialog '3066d6f60ca6277904efb1704ff47764@10.20.0.3:5060' Method: INVITE Audio is at 17054 Adding codec gsm to SDP Adding codec ulaw to SDP Adding codec g723 to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.20.0.30:5060: INVITE sip:100@10.20.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 10.20.0.3:5060;branch=z9hG4bK1351962d;rport Max-Forwards: 70 From: ;tag=as00b30ac7 To: Contact: Call-ID: 7ba1d0712f6f48eb5ea12be23eec2a1b@10.20.0.3:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.6.1 Date: Sun, 05 Jan 2020 13:55:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "09175348234" ;party=calling;privacy=off;screen=no Diversion: ;reason=unconditional Content-Type: application/sdp Content-Length: 300 v=0 o=root 1948569399 1948569399 IN IP4 10.20.0.3 s=Asterisk PBX 16.6.1 c=IN IP4 10.20.0.3 t=0 0 m=audio 17054 RTP/AVP 3 0 4 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv --- -- Called SIP/100 -- Local/100@admin-00000005;1 is ringing <--- SIP read from UDP:10.20.0.30:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.20.0.3:5060;branch=z9hG4bK1351962d;rport=5060 From: ;tag=as00b30ac7 To: Call-ID: 7ba1d0712f6f48eb5ea12be23eec2a1b@10.20.0.3:5060 CSeq: 102 INVITE Supported: replaces, path, timer User-Agent: Grandstream GXP1615 1.0.4.138 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:10.20.0.30:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.20.0.3:5060;branch=z9hG4bK1351962d;rport=5060 From: ;tag=as00b30ac7 To: ;tag=1394868544 Call-ID: 7ba1d0712f6f48eb5ea12be23eec2a1b@10.20.0.3:5060 CSeq: 102 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP1615 1.0.4.138 Allow-Events: talk, hold Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- sip_route_dump: route/path hop: -- SIP/100-0000006b is ringing -- Local/100@admin-00000005;1 is ringing <--- SIP read from UDP:10.211.3.12:5060 ---> INVITE sip:0885250001@172.16.1.99:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 From: ;tag=ymmyzz4w-CC-33 To: CSeq: 1 INVITE Max-Forwards: 68 Contact: Min-SE: 90 Session-Expires: 1800 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER User-Agent: Huawei SoftX3000 V300R010 Supported: 100rel,timer Content-Length: 331 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 6370380 6370380 IN IP4 10.211.3.12 s=Sip Call c=IN IP4 10.211.3.12 t=0 0 m=audio 65280 RTP/AVP 8 0 18 4 105 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:105 CLEARMODE/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> --- (15 headers 14 lines) --- Ignoring this INVITE request <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> Audio is at 11574 Adding codec alaw to SDP Adding codec ulaw to SDP Adding codec g729 to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296539 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv <------------> Retransmitting #2 (no NAT) to 10.211.3.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296537 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:10.211.3.12:5060 ---> INVITE sip:0885250001@172.16.1.99:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 From: ;tag=ymmyzz4w-CC-33 To: CSeq: 1 INVITE Max-Forwards: 68 Contact: Min-SE: 90 Session-Expires: 1800 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER User-Agent: Huawei SoftX3000 V300R010 Supported: 100rel,timer Content-Length: 331 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 6370380 6370380 IN IP4 10.211.3.12 s=Sip Call c=IN IP4 10.211.3.12 t=0 0 m=audio 65280 RTP/AVP 8 0 18 4 105 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:105 CLEARMODE/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> --- (15 headers 14 lines) --- Ignoring this INVITE request <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> Audio is at 11574 Adding codec alaw to SDP Adding codec ulaw to SDP Adding codec g729 to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296540 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv <------------> Retransmitting #3 (no NAT) to 10.211.3.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296537 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:10.211.3.12:5060 ---> INVITE sip:0885250001@172.16.1.99:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 From: ;tag=ymmyzz4w-CC-33 To: CSeq: 1 INVITE Max-Forwards: 68 Contact: Min-SE: 90 Session-Expires: 1800 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER User-Agent: Huawei SoftX3000 V300R010 Supported: 100rel,timer Content-Length: 331 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 6370380 6370380 IN IP4 10.211.3.12 s=Sip Call c=IN IP4 10.211.3.12 t=0 0 m=audio 65280 RTP/AVP 8 0 18 4 105 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:105 CLEARMODE/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> --- (15 headers 14 lines) --- Ignoring this INVITE request <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> Audio is at 11574 Adding codec alaw to SDP Adding codec ulaw to SDP Adding codec g729 to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296541 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv <------------> Retransmitting #4 (no NAT) to 10.211.3.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296537 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:10.211.3.12:5060 ---> INVITE sip:0885250001@172.16.1.99:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 From: ;tag=ymmyzz4w-CC-33 To: CSeq: 1 INVITE Max-Forwards: 68 Contact: Min-SE: 90 Session-Expires: 1800 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER User-Agent: Huawei SoftX3000 V300R010 Supported: 100rel,timer Content-Length: 331 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 6370380 6370380 IN IP4 10.211.3.12 s=Sip Call c=IN IP4 10.211.3.12 t=0 0 m=audio 65280 RTP/AVP 8 0 18 4 105 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:105 CLEARMODE/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> --- (15 headers 14 lines) --- Ignoring this INVITE request <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> Audio is at 11574 Adding codec alaw to SDP Adding codec ulaw to SDP Adding codec g729 to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296542 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv <------------> Retransmitting #5 (no NAT) to 10.211.3.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296537 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:10.211.3.12:5060 ---> INVITE sip:0885250001@172.16.1.99:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 From: ;tag=ymmyzz4w-CC-33 To: CSeq: 1 INVITE Max-Forwards: 68 Contact: Min-SE: 90 Session-Expires: 1800 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER User-Agent: Huawei SoftX3000 V300R010 Supported: 100rel,timer Content-Length: 331 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 6370380 6370380 IN IP4 10.211.3.12 s=Sip Call c=IN IP4 10.211.3.12 t=0 0 m=audio 65280 RTP/AVP 8 0 18 4 105 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:105 CLEARMODE/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> --- (15 headers 14 lines) --- Ignoring this INVITE request <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> Audio is at 11574 Adding codec alaw to SDP Adding codec ulaw to SDP Adding codec g729 to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296543 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv <------------> Retransmitting #6 (no NAT) to 10.211.3.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296537 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:10.211.3.12:5060 ---> INVITE sip:0885250001@172.16.1.99:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 From: ;tag=ymmyzz4w-CC-33 To: CSeq: 1 INVITE Max-Forwards: 68 Contact: Min-SE: 90 Session-Expires: 1800 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER User-Agent: Huawei SoftX3000 V300R010 Supported: 100rel,timer Content-Length: 331 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 6370380 6370380 IN IP4 10.211.3.12 s=Sip Call c=IN IP4 10.211.3.12 t=0 0 m=audio 65280 RTP/AVP 8 0 18 4 105 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:105 CLEARMODE/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> --- (15 headers 14 lines) --- Ignoring this INVITE request <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> Audio is at 11574 Adding codec alaw to SDP Adding codec ulaw to SDP Adding codec g729 to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296544 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv <------------> Retransmitting #7 (no NAT) to 10.211.3.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296537 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:10.211.3.12:5060 ---> INVITE sip:0885250001@172.16.1.99:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 From: ;tag=ymmyzz4w-CC-33 To: CSeq: 1 INVITE Max-Forwards: 68 Contact: Min-SE: 90 Session-Expires: 1800 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER User-Agent: Huawei SoftX3000 V300R010 Supported: 100rel,timer Content-Length: 331 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 6370380 6370380 IN IP4 10.211.3.12 s=Sip Call c=IN IP4 10.211.3.12 t=0 0 m=audio 65280 RTP/AVP 8 0 18 4 105 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:105 CLEARMODE/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> --- (15 headers 14 lines) --- Ignoring this INVITE request <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> Audio is at 11574 Adding codec alaw to SDP Adding codec ulaw to SDP Adding codec g729 to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296545 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv <------------> Retransmitting #8 (no NAT) to 10.211.3.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296537 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv --- <--- SIP read from UDP:10.211.3.12:5060 ---> INVITE sip:0885250001@172.16.1.99:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 From: ;tag=ymmyzz4w-CC-33 To: CSeq: 1 INVITE Max-Forwards: 68 Contact: Min-SE: 90 Session-Expires: 1800 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER User-Agent: Huawei SoftX3000 V300R010 Supported: 100rel,timer Content-Length: 331 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 6370380 6370380 IN IP4 10.211.3.12 s=Sip Call c=IN IP4 10.211.3.12 t=0 0 m=audio 65280 RTP/AVP 8 0 18 4 105 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:105 CLEARMODE/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> --- (15 headers 14 lines) --- Ignoring this INVITE request <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> Audio is at 11574 Adding codec alaw to SDP Adding codec ulaw to SDP Adding codec g729 to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.211.3.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296546 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv <------------> Retransmitting #9 (no NAT) to 10.211.3.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296537 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv --- -- Nobody picked up in 30000 ms Scheduling destruction of SIP dialog '7ba1d0712f6f48eb5ea12be23eec2a1b@10.20.0.3:5060' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 10.20.0.30:5060: CANCEL sip:100@10.20.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 10.20.0.3:5060;branch=z9hG4bK1351962d;rport Max-Forwards: 70 From: ;tag=as00b30ac7 To: Call-ID: 7ba1d0712f6f48eb5ea12be23eec2a1b@10.20.0.3:5060 CSeq: 102 CANCEL User-Agent: Asterisk PBX 16.6.1 Content-Length: 0 --- Scheduling destruction of SIP dialog '7ba1d0712f6f48eb5ea12be23eec2a1b@10.20.0.3:5060' in 32000 ms (Method: INVITE) == Spawn extension (admin, 100, 4) exited non-zero on 'Local/100@admin-00000005;2' -- Executing [h@admin:1] GotoIf("Local/100@admin-00000005;2", "1?hangup:continue") in new stack -- Goto (admin,h,13) -- Executing [h@admin:13] NoOp("Local/100@admin-00000005;2", "No need to Log this call") in new stack -- Executing [0885250001@inbound:3] Hangup("SIP/pldt-in-00000069", "") in new stack == Spawn extension (inbound, 0885250001, 3) exited non-zero on 'SIP/pldt-in-00000069' Scheduling destruction of SIP dialog 'sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000' in 32000 ms (Method: INVITE) <--- SIP read from UDP:10.20.0.30:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.20.0.3:5060;branch=z9hG4bK1351962d;rport=5060 From: ;tag=as00b30ac7 To: ;tag=1394868544 Call-ID: 7ba1d0712f6f48eb5ea12be23eec2a1b@10.20.0.3:5060 CSeq: 102 CANCEL Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP1615 1.0.4.138 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:10.20.0.30:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.20.0.3:5060;branch=z9hG4bK1351962d;rport=5060 From: ;tag=as00b30ac7 To: ;tag=1394868544 Call-ID: 7ba1d0712f6f48eb5ea12be23eec2a1b@10.20.0.3:5060 CSeq: 102 INVITE Contact: Supported: replaces, path, timer User-Agent: Grandstream GXP1615 1.0.4.138 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (NAT) to 10.20.0.30:5060: ACK sip:100@10.20.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 10.20.0.3:5060;branch=z9hG4bK1351962d;rport Max-Forwards: 70 From: ;tag=as00b30ac7 To: ;tag=1394868544 Contact: Call-ID: 7ba1d0712f6f48eb5ea12be23eec2a1b@10.20.0.3:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 16.6.1 Content-Length: 0 --- Scheduling destruction of SIP dialog '7ba1d0712f6f48eb5ea12be23eec2a1b@10.20.0.3:5060' in 32000 ms (Method: INVITE) <--- SIP read from UDP:10.211.3.12:5060 ---> INVITE sip:0885250001@172.16.1.99:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 From: ;tag=ymmyzz4w-CC-33 To: CSeq: 1 INVITE Max-Forwards: 68 Contact: Min-SE: 90 Session-Expires: 1800 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER User-Agent: Huawei SoftX3000 V300R010 Supported: 100rel,timer Content-Length: 331 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 6370380 6370380 IN IP4 10.211.3.12 s=Sip Call c=IN IP4 10.211.3.12 t=0 0 m=audio 65280 RTP/AVP 8 0 18 4 105 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:105 CLEARMODE/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-15 <-------------> --- (15 headers 14 lines) --- Ignoring this INVITE request Retransmitting #10 (no NAT) to 10.211.3.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.211.3.12:5060;branch=z9hG4bK4vtut80f8nhe4rf6riqm3aqi94;received=10.211.3.12 From: ;tag=ymmyzz4w-CC-33 To: ;tag=as1d4d89f7 Call-ID: sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 CSeq: 1 INVITE Server: Asterisk PBX 16.6.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 306 v=0 o=root 204296537 204296537 IN IP4 172.16.1.99 s=Asterisk PBX 16.6.1 c=IN IP4 172.16.1.99 t=0 0 m=audio 11574 RTP/AVP 8 0 18 127 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=maxptime:150 a=sendrecv --- [Jan 5 13:56:26] WARNING[945]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32001ms with no response Really destroying SIP dialog 'sn5u9mnt34mvm25vs5y89ayata2zsz4m@SoftX3000' Method: INVITE