Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log of such a call. Any help would be high appreciated.
regards t. asterix*CLI> sip debug SIP Debugging enabled asterix*CLI> <-- SIP read from 217.10.79.9:5060: --- (0 headers 0 lines) Nat keepalive --- asterix*CLI> <-- SIP read from 192.168.150.51:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 70 Content-Length: 293 Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK6abcc4ace Call-ID: 94cba353ee1163b From: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd To: sip:[EMAIL PROTECTED] CSeq: 2078851383 INVITE Supported: timer Session-Expires: 7200 Allow-Events: talk, hold, conference Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO Content-Type: application/sdp Contact: Steffen <sip:[EMAIL PROTECTED]:5060;transport=udp> Supported: replaces User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 v=0 o=MxSIP 0 1730916047 IN IP4 192.168.150.51 s=SIP Call c=IN IP4 192.168.150.51 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (16 headers 13 lines) --- Using INVITE request as basis request - 94cba353ee1163b Sending to 192.168.150.51 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.150.51:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK6abcc4ace;received=192.168.150.51 From: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd To: sip:[EMAIL PROTECTED];tag=as06838deb Call-ID: 94cba353ee1163b CSeq: 2078851383 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7e84319d" Content-Length: 0 --- Scheduling destruction of call '94cba353ee1163b' in 15000 ms Found user '116' asterix*CLI> <-- SIP read from 192.168.150.51:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK6abcc4ace Call-ID: 94cba353ee1163b From: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd To: sip:[EMAIL PROTECTED];tag=as06838deb CSeq: 2078851383 ACK User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 --- (9 headers 0 lines) --- asterix*CLI> <-- SIP read from 192.168.150.51:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 70 Content-Length: 293 Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK2e49558fa Call-ID: 94cba353ee1163b From: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd To: sip:[EMAIL PROTECTED] CSeq: 2078851384 INVITE Supported: timer Session-Expires: 7200 Allow-Events: talk, hold, conference Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO Content-Type: application/sdp Proxy-Authorization:Digest response="ea33e742f1b16d49344c67d8cc980a16",username="116",realm="asterisk",nonce="7e84319d",algorithm=MD5,uri="sip:[EMAIL PROTECTED]" Supported: replaces Contact: Steffen <sip:[EMAIL PROTECTED]:5060;transport=udp> User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 v=0 o=MxSIP 0 1730916047 IN IP4 192.168.150.51 s=SIP Call c=IN IP4 192.168.150.51 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (17 headers 13 lines) --- Using INVITE request as basis request - 94cba353ee1163b Sending to 192.168.150.51 : 5060 (non-NAT) Found user '116' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 9 Found RTP audio format 101 Peer audio RTP is at port 192.168.150.51:5004 Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G722 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw| g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 119 in default (domain 192.168.150.151) list_route: hop: <sip:[EMAIL PROTECTED]:5060;transport=udp> Transmitting (no NAT) to 192.168.150.51:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK2e49558fa;received=192.168.150.51 From: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd To: sip:[EMAIL PROTECTED] Call-ID: 94cba353ee1163b CSeq: 2078851384 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- We're at 192.168.150.151 port 16216 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.150.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK2e49558fa;received=192.168.150.51 From: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd To: sip:[EMAIL PROTECTED];tag=as36ac8450 Call-ID: 94cba353ee1163b CSeq: 2078851384 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 246 v=0 o=root 24922 24922 IN IP4 192.168.150.151 s=session c=IN IP4 192.168.150.151 t=0 0 m=audio 16216 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- We're at 192.168.150.151 port 7280 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 192.168.150.11:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport From: "Steffen" <sip:[EMAIL PROTECTED]>;tag=as085e9ac6 To: <sip:[EMAIL PROTECTED]:5060> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 20 Nov 2007 09:41:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 245 v=0 o=root 24922 24922 IN IP4 192.168.150.151 s=session c=IN IP4 192.168.150.151 t=0 0 m=audio 7280 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterix*CLI> <-- SIP read from 192.168.150.11:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport From: "Steffen" <sip:[EMAIL PROTECTED]>;tag=as085e9ac6 To: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream BT200 1.1.1.14 Content-Length: 0 --- (8 headers 0 lines) --- asterix*CLI> <-- SIP read from 192.168.150.11:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport From: "Steffen" <sip:[EMAIL PROTECTED]>;tag=as085e9ac6 To: <sip:[EMAIL PROTECTED]:5060>;tag=1fc925a8554a4f7a Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream BT200 1.1.1.14 Contact: <sip:[EMAIL PROTECTED]:5060> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 --- (10 headers 0 lines) --- asterix*CLI> <-- SIP read from 192.168.150.51:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 192.168.150.51:5060;branch=z9hG4bK40bfb60e5 Call-ID: 94cba353ee1163b From: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd To: sip:[EMAIL PROTECTED];tag=as36ac8450 CSeq: 2078851384 ACK Proxy-Authorization:Digest response="ea33e742f1b16d49344c67d8cc980a16",username="116",realm="asterisk",nonce="7e84319d",algorithm=MD5,uri="sip:[EMAIL PROTECTED]" User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 --- (10 headers 0 lines) --- asterix*CLI> <-- SIP read from 192.168.150.51:5060: --- (0 headers 0 lines) Nat keepalive --- Destroying call '[EMAIL PROTECTED]' asterix*CLI> <-- SIP read from 192.168.150.43:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.150.43;branch=z9hG4bK4c66703c19E1401F From: "Thomas Stein" <sip:[EMAIL PROTECTED]>;tag=ED14397B-9031B306 To: <sip:[EMAIL PROTECTED]> CSeq: 1 INVITE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.1.0037 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 253 v=0 o=- 1195480065 1195480065 IN IP4 192.168.150.43 s=Polycom IP Phone c=IN IP4 192.168.150.43 t=0 0 m=audio 2222 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 11 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.150.43 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.150.43:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.150.43;branch=z9hG4bK4c66703c19E1401F;received=192.168.150.43 From: "Thomas Stein" <sip:[EMAIL PROTECTED]>;tag=ED14397B-9031B306 To: <sip:[EMAIL PROTECTED]>;tag=as41909f67 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="059c6efc" Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '118' asterix*CLI> <-- SIP read from 192.168.150.43:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.150.43;branch=z9hG4bK4c66703c19E1401F From: "Thomas Stein" <sip:[EMAIL PROTECTED]>;tag=ED14397B-9031B306 To: <sip:[EMAIL PROTECTED]>;tag=as41909f67 CSeq: 1 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.1.0037 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines) --- asterix*CLI> <-- SIP read from 192.168.150.43:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.150.43;branch=z9hG4bKf571a0598596EE34 From: "Thomas Stein" <sip:[EMAIL PROTECTED]>;tag=ED14397B-9031B306 To: <sip:[EMAIL PROTECTED]> CSeq: 2 INVITE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.1.0037 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="118", realm="asterisk", nonce="059c6efc", uri="sip:[EMAIL PROTECTED]:5060", response="3025cdb052cbf156933a61f6470dd21d", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 253 v=0 o=- 1195480065 1195480065 IN IP4 192.168.150.43 s=Polycom IP Phone c=IN IP4 192.168.150.43 t=0 0 m=audio 2222 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.150.43 : 5060 (non-NAT) Found user '118' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.150.43:2222 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw| g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for *8 in default (domain 192.168.150.151) list_route: hop: <sip:[EMAIL PROTECTED]> Transmitting (no NAT) to 192.168.150.43:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.150.43;branch=z9hG4bKf571a0598596EE34;received=192.168.150.43 From: "Thomas Stein" <sip:[EMAIL PROTECTED]>;tag=ED14397B-9031B306 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- We're at 192.168.150.151 port 13978 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.150.43:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.150.43;branch=z9hG4bKf571a0598596EE34;received=192.168.150.43 From: "Thomas Stein" <sip:[EMAIL PROTECTED]>;tag=ED14397B-9031B306 To: <sip:[EMAIL PROTECTED]>;tag=as45b8e08e Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 246 v=0 o=root 24922 24922 IN IP4 192.168.150.151 s=session c=IN IP4 192.168.150.151 t=0 0 m=audio 13978 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Scheduling destruction of call '[EMAIL PROTECTED]' in 32000 ms Reliably Transmitting (no NAT) to 192.168.150.11:5060: CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport From: "Steffen" <sip:[EMAIL PROTECTED]>;tag=as085e9ac6 To: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing <sip:[EMAIL PROTECTED]:5060;transport=udp> for address/port to send to set_destination: set destination to 192.168.150.51, port 5060 We're at 192.168.150.151 port 16216 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 13 lines Reliably Transmitting (no NAT) to 192.168.150.51:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK78e7cf84;rport From: sip:[EMAIL PROTECTED];tag=as36ac8450 To: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd Contact: <sip:[EMAIL PROTECTED]> Call-ID: 94cba353ee1163b CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 290 v=0 o=root 24922 24923 IN IP4 192.168.150.43 s=session c=IN IP4 192.168.150.43 t=0 0 m=audio 2222 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterix*CLI> <-- SIP read from 192.168.150.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport From: "Steffen" <sip:[EMAIL PROTECTED]>;tag=as085e9ac6 To: <sip:[EMAIL PROTECTED]:5060>;tag=1fc925a8554a4f7a Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent: Grandstream BT200 1.1.1.14 Contact: <sip:[EMAIL PROTECTED]:5060> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines) --- asterix*CLI> <-- SIP read from 192.168.150.11:5060: SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport From: "Steffen" <sip:[EMAIL PROTECTED]>;tag=as085e9ac6 To: <sip:[EMAIL PROTECTED]:5060>;tag=1fc925a8554a4f7a Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream BT200 1.1.1.14 Content-Length: 0 --- (8 headers 0 lines) --- Transmitting (no NAT) to 192.168.150.11:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 ia: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK41bcdf63;rport From: "Steffen" <sip:[EMAIL PROTECTED]>;tag=as085e9ac6 To: <sip:[EMAIL PROTECTED]:5060>;tag=1fc925a8554a4f7a Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' asterix*CLI> <-- SIP read from 192.168.150.43:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.150.43;branch=z9hG4bKdccbff09857C5B3 From: "Thomas Stein" <sip:[EMAIL PROTECTED]>;tag=ED14397B-9031B306 To: <sip:[EMAIL PROTECTED]>;tag=as45b8e08e CSeq: 2 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.1.0037 Proxy-Authorization: Digest username="118", realm="asterisk", nonce="059c6efc", uri="sip:[EMAIL PROTECTED]:5060", response="3025cdb052cbf156933a61f6470dd21d", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 --- (12 headers 0 lines) --- set_destination: Parsing <sip:[EMAIL PROTECTED]> for address/port to send to set_destination: set destination to 192.168.150.43, port 5060 We're at 192.168.150.151 port 13978 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 15 lines Reliably Transmitting (no NAT) to 192.168.150.43:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK7055cc60;rport From: <sip:[EMAIL PROTECTED]>;tag=as45b8e08e To: "Thomas Stein" <sip:[EMAIL PROTECTED]>;tag=ED14397B-9031B306 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 334 v=0 o=root 24922 24923 IN IP4 192.168.150.51 s=session c=IN IP4 192.168.150.51 t=0 0 m=audio 5004 RTP/AVP 8 0 4 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterix*CLI> <-- SIP read from 192.168.150.43:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK7055cc60;rport From: <sip:[EMAIL PROTECTED]>;tag=as45b8e08e To: "Thomas Stein" <sip:[EMAIL PROTECTED]>;tag=ED14397B-9031B306 CSeq: 102 INVITE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.1.0037 Content-Type: application/sdp Content-Length: 191 v=0 o=- 1195480065 1195480066 IN IP4 192.168.150.43 s=Polycom IP Phone c=IN IP4 192.168.150.43 t=0 0 m=audio 2222 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (11 headers 8 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.150.43:2222 Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:[EMAIL PROTECTED]> set_destination: Parsing <sip:[EMAIL PROTECTED]> for address/port to send to set_destination: set destination to 192.168.150.43, port 5060 Transmitting (no NAT) to 192.168.150.43:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK55ee2bfa;rport From: <sip:[EMAIL PROTECTED]>;tag=as45b8e08e To: "Thomas Stein" <sip:[EMAIL PROTECTED]>;tag=ED14397B-9031B306 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterix*CLI> <-- SIP read from 192.168.150.51:5060: SIP/2.0 100 Trying Call-ID: 94cba353ee1163b CSeq: 102 INVITE From: sip:[EMAIL PROTECTED];tag=as36ac8450 To: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK78e7cf84;rport Content-Length: 0 User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 --- (8 headers 0 lines) --- set_destination: Parsing <sip:[EMAIL PROTECTED]:5060;transport=udp> for address/port to send to set_destination: set destination to 192.168.150.51, port 5060 We're at 192.168.150.151 port 16216 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.150.51:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK53eb19ca;rport From: sip:[EMAIL PROTECTED];tag=as36ac8450 To: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd Contact: <sip:[EMAIL PROTECTED]> Call-ID: 94cba353ee1163b CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 219 v=0 o=root 24922 24924 IN IP4 192.168.150.43 s=session c=IN IP4 192.168.150.43 t=0 0 m=audio 2222 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterix*CLI> <-- SIP read from 192.168.150.51:5060: SIP/2.0 500 Server Internal Error Call-ID: 94cba353ee1163b CSeq: 103 INVITE From: sip:[EMAIL PROTECTED];tag=as36ac8450 To: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK53eb19ca;rport Content-Length: 0 Retry-After: 8 User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 --- (9 headers 0 lines) --- set_destination: Parsing <sip:[EMAIL PROTECTED]:5060;transport=udp> for address/port to send to set_destination: set destination to 192.168.150.51, port 5060 Transmitting (no NAT) to 192.168.150.51:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK53eb19ca;rport From: sip:[EMAIL PROTECTED];tag=as36ac8450 To: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd Contact: <sip:[EMAIL PROTECTED]> Call-ID: 94cba353ee1163b CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 32000 ms set_destination: Parsing <sip:[EMAIL PROTECTED]> for address/port to send to set_destination: set destination to 192.168.150.43, port 5060 Reliably Transmitting (no NAT) to 192.168.150.43:5060: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK3b752b3b;rport From: <sip:[EMAIL PROTECTED]>;tag=as45b8e08e To: "Thomas Stein" <sip:[EMAIL PROTECTED]>;tag=ED14397B-9031B306 Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterix*CLI> <-- SIP read from 192.168.150.43:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK3b752b3b;rport From: <sip:[EMAIL PROTECTED]>;tag=as45b8e08e To: "Thomas Stein" <sip:[EMAIL PROTECTED]>;tag=ED14397B-9031B306 CSeq: 103 BYE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.1.0037 Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' asterix*CLI> <-- SIP read from 192.168.150.51:5060: SIP/2.0 200 OK Call-ID: 94cba353ee1163b CSeq: 102 INVITE From: sip:[EMAIL PROTECTED];tag=as36ac8450 To: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK78e7cf84;rport Content-Length: 257 Session-Expires: 7200;refresher=uas Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO Content-Type: application/sdp Supported: replaces Contact: Steffen <sip:[EMAIL PROTECTED]:5060;transport=udp> User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 v=0 o=MxSIP 0 1730916047 IN IP4 192.168.150.51 s=SIP Call c=IN IP4 192.168.150.51 t=0 0 m=audio 5004 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (13 headers 12 lines) --- asterix*CLI> <-- SIP read from 192.168.150.51:5060: SIP/2.0 200 OK Call-ID: 94cba353ee1163b CSeq: 102 INVITE From: sip:[EMAIL PROTECTED];tag=as36ac8450 To: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK78e7cf84;rport Content-Length: 257 Session-Expires: 7200;refresher=uas Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO Content-Type: application/sdp Supported: replaces Contact: Steffen <sip:[EMAIL PROTECTED]:5060;transport=udp> User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 v=0 o=MxSIP 0 1730916047 IN IP4 192.168.150.51 s=SIP Call c=IN IP4 192.168.150.51 t=0 0 m=audio 5004 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (13 headers 12 lines) --- asterix*CLI> <-- SIP read from 192.168.150.51:5060: SIP/2.0 200 OK Call-ID: 94cba353ee1163b CSeq: 102 INVITE From: sip:[EMAIL PROTECTED];tag=as36ac8450 To: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK78e7cf84;rport Content-Length: 257 Session-Expires: 7200;refresher=uas Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO Content-Type: application/sdp Supported: replaces Contact: Steffen <sip:[EMAIL PROTECTED]:5060;transport=udp> User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 v=0 o=MxSIP 0 1730916047 IN IP4 192.168.150.51 s=SIP Call c=IN IP4 192.168.150.51 t=0 0 m=audio 5004 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (13 headers 12 lines) --- asterix*CLI> <-- SIP read from 217.10.79.9:5060: --- (0 headers 0 lines) Nat keepalive --- asterix*CLI> sip de <-- SIP read from 192.168.150.51:5060: SIP/2.0 200 OK Call-ID: 94cba353ee1163b CSeq: 102 INVITE From: sip:[EMAIL PROTECTED];tag=as36ac8450 To: Steffen <sip:[EMAIL PROTECTED]>;tag=19a39a2dc7a54cd Via: SIP/2.0/UDP 192.168.150.151:5060;branch=z9hG4bK78e7cf84;rport Content-Length: 257 Session-Expires: 7200;refresher=uas Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO Content-Type: application/sdp Supported: replaces Contact: Steffen <sip:[EMAIL PROTECTED]:5060;transport=udp> User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8 v=0 o=MxSIP 0 1730916047 IN IP4 192.168.150.51 s=SIP Call c=IN IP4 192.168.150.51 t=0 0 m=audio 5004 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (13 headers 12 lines) --- asterix*CLI> sip no debug SIP Debugging Disabled -- knowledgeTools® ... managing complexity. -------------------------------------------------- knowledgeTools International GmbH Wallstraße 15 / 15 a 10179 Berlin Fon: +49 30 726 169 20 Fax: +49 30 726 169 249 [EMAIL PROTECTED] www.knowledgetools.de Sitz Berlin, AG Berlin-Charlottenburg, HRB 86378 Geschäftsführer: Oliver Seyboldt, Reinhard Kunz 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