-- Executing [0372xxxxxx@default:1] Set("SIP/Acc_0-0000004f", "CALLERID(num)=0344xxxxxx") in new stack -- Executing [0372xxxxxx@default:2] Dial("SIP/Acc_0-0000004f", "SIP/0372xxxxxx@Telekom") in new stack == Using SIP RTP TOS bits 160 == Using SIP RTP CoS mark 5 Audio is at 19462 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x800000000000 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 217.0.23.100:5060: INVITE sip:0372xxxxxx@tel.t-online.de SIP/2.0 Via: SIP/2.0/UDP 192.168.80.201:5060;branch=z9hG4bK47d1b1b2 Max-Forwards: 70 From: "Sekretariat" ;tag=as68a7d476 To: Contact: Call-ID: 2daed05303ef9b0e0bf921f23564e332@tel.t-online.de CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Date: Mon, 02 Oct 2017 08:52:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Type: application/sdp Content-Length: 303 v=0 o=root 1339013114 1339013114 IN IP4 192.168.80.201 s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 c=IN IP4 192.168.80.201 t=0 0 m=audio 19462 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called SIP/0372xxxxxx@Telekom Retransmitting #1 (no NAT) to 217.0.23.100:5060: INVITE sip:0372xxxxxx@tel.t-online.de SIP/2.0 Via: SIP/2.0/UDP 192.168.80.201:5060;branch=z9hG4bK47d1b1b2 Max-Forwards: 70 From: "Sekretariat" ;tag=as68a7d476 To: Contact: Call-ID: 2daed05303ef9b0e0bf921f23564e332@tel.t-online.de CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Date: Mon, 02 Oct 2017 08:52:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Type: application/sdp Content-Length: 303 v=0 o=root 1339013114 1339013114 IN IP4 192.168.80.201 s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 c=IN IP4 192.168.80.201 t=0 0 m=audio 19462 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:217.0.23.100:5060 ---> SIP/2.0 407 Proxy Authentication Required 02035034C Via: SIP/2.0/UDP 192.168.80.201:5060;received=79.213.47.119;branch=z9hG4bK47d1b1b2 To: ;tag=h7g4Esbg_92cd5f8d035a2a9080cbe8b22eb17cb5 From: "Sekretariat" ;tag=as68a7d476 Call-ID: 2daed05303ef9b0e0bf921f23564e332@tel.t-online.de CSeq: 102 INVITE Content-Length: 0 Proxy-Authenticate: Digest nonce="CF9B9CAC50FED1590000000030A12A63",realm="tel.t-online.de",algorithm=MD5,qop="auth",stale=true <-------------> --- (8 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 217.0.23.100:5060 Transmitting (no NAT) to 217.0.23.100:5060: ACK sip:0372xxxxxx@tel.t-online.de SIP/2.0 Via: SIP/2.0/UDP 192.168.80.201:5060;branch=z9hG4bK47d1b1b2 Max-Forwards: 70 From: "Sekretariat" ;tag=as68a7d476 To: ;tag=h7g4Esbg_92cd5f8d035a2a9080cbe8b22eb17cb5 Contact: Call-ID: 2daed05303ef9b0e0bf921f23564e332@tel.t-online.de CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Content-Length: 0 --- Audio is at 19462 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x800000000000 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 217.0.23.100:5060: INVITE sip:0372xxxxxx@tel.t-online.de SIP/2.0 Via: SIP/2.0/UDP 192.168.80.201:5060;branch=z9hG4bK7f7b1d06 Max-Forwards: 70 From: "Sekretariat" ;tag=as68a7d476 To: Contact: Call-ID: 2daed05303ef9b0e0bf921f23564e332@tel.t-online.de CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Proxy-Authorization: Digest username="anonymous@t-online.de", realm="tel.t-online.de", algorithm=MD5, uri="sip:0372xxxxxx@tel.t-online.de", nonce="CF9B9CAC50FED1590000000030A12A63", response="6cd4f57337897b35cb8b5de8cb53b275", qop=auth, cnonce="3543510d", nc=00000001 Date: Mon, 02 Oct 2017 08:52:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Type: application/sdp Content-Length: 303 v=0 o=root 1339013114 1339013115 IN IP4 192.168.80.201 s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 c=IN IP4 192.168.80.201 t=0 0 m=audio 19462 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:217.0.23.100:5060 ---> SIP/2.0 407 Proxy Authentication Required 02035034C Via: SIP/2.0/UDP 192.168.80.201:5060;received=79.213.47.119;branch=z9hG4bK47d1b1b2 To: ;tag=h7g4Esbg_92cd5f8d035a2a9080cbe8b22eb17cb5 From: "Sekretariat" ;tag=as68a7d476 Call-ID: 2daed05303ef9b0e0bf921f23564e332@tel.t-online.de CSeq: 102 INVITE Content-Length: 0 Proxy-Authenticate: Digest nonce="CF9B9CAC50FED1590000000030A12A63",realm="tel.t-online.de",algorithm=MD5,qop="auth",stale=true <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to 217.0.23.100:5060: ACK sip:0372xxxxxx@tel.t-online.de SIP/2.0 Via: SIP/2.0/UDP 192.168.80.201:5060;branch=z9hG4bK7f7b1d06 Max-Forwards: 70 From: "Sekretariat" ;tag=as68a7d476 To: Contact: Call-ID: 2daed05303ef9b0e0bf921f23564e332@tel.t-online.de CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Content-Length: 0 --- Retransmitting #1 (no NAT) to 217.0.23.100:5060: INVITE sip:0372xxxxxx@tel.t-online.de SIP/2.0 Via: SIP/2.0/UDP 192.168.80.201:5060;branch=z9hG4bK7f7b1d06 Max-Forwards: 70 From: "Sekretariat" ;tag=as68a7d476 To: Contact: Call-ID: 2daed05303ef9b0e0bf921f23564e332@tel.t-online.de CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Proxy-Authorization: Digest username="anonymous@t-online.de", realm="tel.t-online.de", algorithm=MD5, uri="sip:0372xxxxxx@tel.t-online.de", nonce="CF9B9CAC50FED1590000000030A12A63", response="6cd4f57337897b35cb8b5de8cb53b275", qop=auth, cnonce="3543510d", nc=00000001 Date: Mon, 02 Oct 2017 08:52:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Type: application/sdp Content-Length: 303 v=0 o=root 1339013114 1339013115 IN IP4 192.168.80.201 s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 c=IN IP4 192.168.80.201 t=0 0 m=audio 19462 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:217.0.23.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.80.201:5060;received=79.213.47.119;branch=z9hG4bK7f7b1d06 To: From: "Sekretariat" ;tag=as68a7d476 Call-ID: 2daed05303ef9b0e0bf921f23564e332@tel.t-online.de CSeq: 103 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:217.0.23.100:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.80.201:5060;received=79.213.47.119;branch=z9hG4bK7f7b1d06 To: ;tag=h7g4Esbg_p65549t1506934340m760513c1882918197s1_2115287555-533452429 From: "Sekretariat" ;tag=as68a7d476 Call-ID: 2daed05303ef9b0e0bf921f23564e332@tel.t-online.de CSeq: 103 INVITE Contact: Record-Route: P-Early-Media: sendrecv Supported: timer Content-Type: application/sdp Content-Length: 177 Allow: UPDATE, NOTIFY, PRACK, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER Accept: application/sdp Accept: application/isup Accept: application/xml Accept: application/media_control+xml Accept: application/vnd.etsi.cug+xml Accept: application/vnd.etsi.sci+xml v=0 o=- 1013222434 2116328301 IN IP4 217.0.23.100 s=IMSS c=IN IP4 217.0.4.167 t=0 0 m=audio 45208 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=sendrecv a=ptime:20 <-------------> --- (19 headers 9 lines) --- list_route: hop: Found RTP audio format 8 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), 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 217.0.4.167:45208 -- SIP/Telekom-00000050 is making progress passing it to SIP/Acc_0-0000004f Audio is at 13966 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 192.168.80.21:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.80.21:5060;branch=z9hG4bK_7C2F8020CEF2_T7FF1AB3F;received=192.168.80.21;rport=5060 From: ;tag=7C2F8020CEF2_T1406040966 To: ;tag=as25f57c01 Call-ID: CALL_ID96_7C2F8020CEF2_T602588363@192.168.80.21 CSeq: 2 INVITE Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 280 v=0 o=root 1604381273 1604381273 IN IP4 192.168.80.201 s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 c=IN IP4 192.168.80.201 t=0 0 m=audio 13966 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:217.0.23.100:5060 ---> SIP/2.0 606 Not Acceptable Via: SIP/2.0/UDP 192.168.80.201:5060;received=79.213.47.119;branch=z9hG4bK7f7b1d06 To: ;tag=h7g4Esbg_p65549t1506934340m760513c1882918197s1_2115287555-533452429 From: "Sekretariat" ;tag=as68a7d476 Call-ID: 2daed05303ef9b0e0bf921f23564e332@tel.t-online.de CSeq: 103 INVITE Contact: Reason: Q.850;cause=88;text="5" Supported: timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 217.0.23.100:5060 Transmitting (no NAT) to 217.0.23.100:5060: ACK sip:sgc_c@217.0.23.100;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.80.201:5060;branch=z9hG4bK7f7b1d06 Max-Forwards: 70 From: "Sekretariat" ;tag=as68a7d476 To: ;tag=h7g4Esbg_p65549t1506934340m760513c1882918197s1_2115287555-533452429 Contact: Call-ID: 2daed05303ef9b0e0bf921f23564e332@tel.t-online.de CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Content-Length: 0 --- Scheduling destruction of SIP dialog '2daed05303ef9b0e0bf921f23564e332@tel.t-online.de' in 6400 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/Acc_0-0000004f' status is 'CONGESTION'