laurent

Yes the gateway play back/correctly decode the RTP stream, so I have voice only in one way. I will also check with the audiocode team but is that possible to get more log from the snomone ? I mean to get the expected value and the current value ?

so for incoming call I have changed the config to only have SDP with SRTP and now I have exactly the same error that for outgoing calls, SnomOne drop SRTP packet comming from the M1000 and indicate this error: [4] 2012/07/12 11:35:36:Dropped 10 SRTP packets with wrong MAC [4] 2012/07/12 11:35:38:Dropped 100 SRTP packets with wrong MAC [4] 2012/07/12 11:35:56: Dropped 1000 SRTP packets with wrong MAC regarding the MAC , I see that it's Message Authentication Code used in SRTP, how can we debug this ? Laurent [9] 2012/07/12 11:35:35: Last message repeated 8 times [8] 2012/07/12 11:35:35: Received SIP connection 3 from 95.128.80.120:36212 [9] 2012/07/12 11:35:35: SIP 95.128.80.120:36212: process_client_hello(03014ffea8763d866d889fa9269add497ee328fe4ab0ad93b4aea2cab69b6e8673e600003400c10066006500640063006200390038003500330032002f001600150014001300120011000a00090008000600050004000300ff0100000400230000) [9] 2012/07/12 11:35:35: SIP 95.128.80.120:36212: [a50e8e58] send_server_hello(03014ffe9a67b20f9479c4b77ce12741f0ccca7e481e33721501bb534ef58fed8d3100000500) [9] 2012/07/12 11:35:35: SIP 95.128.80.120:36212: [a50e8e58] send_certificate(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) [9] 2012/07/12 11:35:35: SIP 95.128.80.120:36212: [a50e8e58] send_hello_done() [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] process_client_key_exchange(008013dfeb8f63a8ab71745cf80e252a42893ceae33bef26c0a64a71ce4930b5c545fb965f8df941f722f55b74f6a01475976b0f60767075fda7761c6e4666ecb8ba46f63c16acd6d47d560fe823ea444a7ae32abb80607e532c05e38f722c71d23e2aa9d03ddb36ca70aef46dc63271fa99aed9ca293a1606be79ab52f039d4d09a) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] Pre Master Secret(02e32caa4201d1103ef913c75b1f062a58196ba558be96ec7ded891f4ebbd93c566ed9b24bcbcca6aeea3a60ad17f2d46a61b735e8dc3ca7eac23420fa9c5b5431405968e8069ba4d55891b52ef9000301dc3f8d82fb4aecfd353740ea8519fbcede143df6100ba1f3671f7505ba3f0f4ffc114001b8b3cbc22bcd453460c2) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] Client Random(4ffea8763d866d889fa9269add497ee328fe4ab0ad93b4aea2cab69b6e8673e6) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] Server Random(4ffe9a67b20f9479c4b77ce12741f0ccca7e481e33721501bb534ef58fed8d31) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] Client Random(4ffea8763d866d889fa9269add497ee328fe4ab0ad93b4aea2cab69b6e8673e6) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] Server Random(4ffe9a67b20f9479c4b77ce12741f0ccca7e481e33721501bb534ef58fed8d31) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] Master Secret(3b709cddb1dfe1a570d69534507830742fc3d7c3f501241c0875a56d9ad10b556d0f469be78b59520c5088a3f549c200) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] process_change_cipher_spec(01) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] send_change_cipher_spec(01) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] perform_change_cipher_spec(0005) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] Key Block(ff0c51b54c9eb7a9dddca2da5368614456a5f5c7fb794993382ebf035d0b2bb86c785111a2156ab851feaa74c1421f1238ea08f188aa87b155d71e91dabf7d236a24ddf54c526a8d) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] Client Write MAC Secret(ff0c51b54c9eb7a9dddca2da5368614456a5f5c7) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] Server Write MAC Secret(fb794993382ebf035d0b2bb86c785111a2156ab8) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] Client Write Key(51feaa74c1421f1238ea08f188aa87b1) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] Server Write Key(55d71e91dabf7d236a24ddf54c526a8d) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] process_finished(9edac49384bf73b1fb34e154) [9] 2012/07/12 11:35:36: SIP 95.128.80.120:36212: [a50e8e58] send_finished(f7a0318d6366144cbb5a348e) [5] 2012/07/12 11:35:36: SIP Rx tls:95.128.80.120:36212: INVITE sip:96956110529@95.128.80.120 SIP/2.0 Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1659423642;alias Max-Forwards: 10 From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446 To: <sip:0325520053@95.128.80.120> Call-ID: 16585798691272012113534@95.128.80.120 CSeq: 1 INVITE Contact: <sip:96956110529@95.128.80.120:5067;transport=tls> Allow: ACK,CANCEL,BYE,INFO User-Agent: Mediant 1000 - MSBG/v.6.60A.011.001 Content-Type: application/sdp Content-Length: 531 x-changeuri: 1 v=0 o=Dialogic_SDP 1658080552 1658080517 IN IP4 95.128.80.120 s=Dialogic-SIP c=IN IP4 95.128.80.120 t=0 0 m=audio 8160 RTP/SAVP 8 0 98 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:98 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XiksbNTW1XnGA5ZaqbacEUR5aWwY6Nd9G+wNukYT|2^31|58:1 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:0dPFOIy0LfD62d7JP/MMLlOhw23LED2M61Sd/ibD|2^31|220:1 [8] 2012/07/12 11:35:36: Allocating for call port 0, SIP call id 16585798691272012113534@95.128.80.120 [9] 2012/07/12 11:35:36: UDP(IPv4): Opening socket on 0.0.0.0:50160 [9] 2012/07/12 11:35:36: UDP(IPv4): Opening socket on 0.0.0.0:50161 [9] 2012/07/12 11:35:36: UDP(IPv6): Opening socket on [::]:50160 [9] 2012/07/12 11:35:36: UDP(IPv6): Opening socket on [::]:50161 [5] 2012/07/12 11:35:36: Identify trunk (IP address and domain match) 2 [5] 2012/07/12 11:35:36: SIP Tx tls:95.128.80.120:36212: SIP/2.0 100 Trying Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1659423642;alias From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446 To: <sip:0325520053@95.128.80.120>;tag=3f880a4652 Call-ID: 16585798691272012113534@95.128.80.120 CSeq: 1 INVITE Content-Length: 0 [6] 2012/07/12 11:35:36: Call-leg 0: Sending RTP for 16585798691272012113534@95.128.80.120 to 95.128.80.120:8160, codec not set yet [8] 2012/07/12 11:35:36: Incoming call: Request URI sip:96956110529@95.128.80.120, To is <sip:0325520053@95.128.80.120> [8] 2012/07/12 11:35:36: Call from a trunk 2 [8] 2012/07/12 11:35:36: Trunk Peoplefone@pbx.company.com has country code not set, area code not set [9] 2012/07/12 11:35:36: Incoming: formatted From is = "0763770377" <sip:+0763770377@95.128.80.91;user=phone> [9] 2012/07/12 11:35:36: Incoming: formatted To is = <sip:0325520053@95.128.80.120;user=phone> [9] 2012/07/12 11:35:36: Incoming: formatted URI is = sip:96956110529@pbx.company.com;user=phone [8] 2012/07/12 11:35:36: To is <sip:0325520053@95.128.80.120;user=phone>, user 0, domain 1 [8] 2012/07/12 11:35:36: Send call to extension ERE returned 40 [5] 2012/07/12 11:35:36: Domain trunk Peoplefone@pbx.company.com sends call to 40 in domain pbx.company.com [8] 2012/07/12 11:35:36: Set the To domain based on To user 40@pbx.company.com [8] 2012/07/12 11:35:36: Call state for call object 1: idle [7] 2012/07/12 11:35:36: Call port 0: set_codecs for 16585798691272012113534@95.128.80.120 codecs "", codec_preference count 6 [8] 2012/07/12 11:35:36: Call state for call object 1: alerting [8] 2012/07/12 11:35:36: Play audio_moh/noise.wav, caching true [8] 2012/07/12 11:35:36: Allocating for call port 1, SIP call id 451bca7c@pbx [9] 2012/07/12 11:35:36: UDP(IPv4): Opening socket on 0.0.0.0:60638 [9] 2012/07/12 11:35:36: UDP(IPv4): Opening socket on 0.0.0.0:60639 [9] 2012/07/12 11:35:36: UDP(IPv6): Opening socket on [::]:60638 [9] 2012/07/12 11:35:36: UDP(IPv6): Opening socket on [::]:60639 [7] 2012/07/12 11:35:36: Call port 1: set_codecs for 451bca7c@pbx codecs "", codec_preference count 6 [9] 2012/07/12 11:35:36: Using outbound proxy sip:192.168.1.41:4901;transport=tls because of flow-label [8] 2012/07/12 11:35:36: call port 1: state code from 0 to 100 [9] 2012/07/12 11:35:36: Call port 1: update_codecs for 451bca7c@pbx: adding codec pcmu/8000 to available list [9] 2012/07/12 11:35:36: Call port 1: update_codecs for 451bca7c@pbx: adding codec pcma/8000 to available list [9] 2012/07/12 11:35:36: Call port 1: update_codecs for 451bca7c@pbx: adding codec g722/8000 to available list [9] 2012/07/12 11:35:36: Call port 1: update_codecs for 451bca7c@pbx: adding codec g726-32/8000 to available list [9] 2012/07/12 11:35:36: Call port 1: update_codecs for 451bca7c@pbx: adding codec gsm/8000 to available list [9] 2012/07/12 11:35:36: Call port 1: update_codecs for 451bca7c@pbx: codec_preference size 6, available codecs size 6 [5] 2012/07/12 11:35:36: SIP Tx tls:192.168.1.41:4901: INVITE sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-158f5432f392d55d4e2d63bae9ebbf85;rport From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071 To: "Forty Zero" <sip:40@pbx.company.com> Call-ID: 451bca7c@pbx CSeq: 7946 INVITE Max-Forwards: 70 Contact: <sip:40@192.168.1.201:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Alert-Info: <http://127.0.0.1/Bellcore-dr3> Content-Type: application/sdp Content-Length: 421 v=0 o=- 759227115 759227115 IN IP4 192.168.1.201 s=- c=IN IP4 192.168.1.201 t=0 0 m=audio 60638 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8Gi8jkVgW8DzeqpYxFJSOEeTo+NNGp7kJAW/A/mQ a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [8] 2012/07/12 11:35:36: call port 0: state code from 0 to 100 [9] 2012/07/12 11:35:36: Call port 0: update_codecs for 16585798691272012113534@95.128.80.120: adding codec pcmu/8000 to available list [9] 2012/07/12 11:35:36: Call port 0: update_codecs for 16585798691272012113534@95.128.80.120: adding codec pcma/8000 to available list [9] 2012/07/12 11:35:36: Call port 0: update_codecs for 16585798691272012113534@95.128.80.120: Other side does not support codec g722/8000 [9] 2012/07/12 11:35:36: Call port 0: update_codecs for 16585798691272012113534@95.128.80.120: adding codec g726-32/8000 to available list [9] 2012/07/12 11:35:36: Call port 0: update_codecs for 16585798691272012113534@95.128.80.120: Other side does not support codec gsm/8000 [9] 2012/07/12 11:35:36: Call port 0: update_codecs for 16585798691272012113534@95.128.80.120: codec_preference size 6, available codecs size 4 [5] 2012/07/12 11:35:36: SIP Rx tls:192.168.1.41:4901: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-158f5432f392d55d4e2d63bae9ebbf85;rport=5061 From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071 To: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey Call-ID: 451bca7c@pbx CSeq: 7946 INVITE Contact: <sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt>;reg-id=1 Content-Length: 0 [9] 2012/07/12 11:35:36: Message repetition, packet dropped [5] 2012/07/12 11:35:36: SIP Rx tls:192.168.1.41:4901: SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-158f5432f392d55d4e2d63bae9ebbf85;rport=5061 From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071 To: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey Call-ID: 451bca7c@pbx CSeq: 7946 INVITE Contact: <sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt>;reg-id=1 Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Content-Length: 0 [5] 2012/07/12 11:35:36: SIP Tx tls:192.168.1.41:4901: PRACK sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-f12868f813612b8534cc8634c840c740;rport From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071 To: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey Call-ID: 451bca7c@pbx CSeq: 7947 PRACK Max-Forwards: 70 Contact: <sip:40@192.168.1.201:5061;transport=tls> RAck: 1 7946 INVITE Content-Length: 0 [8] 2012/07/12 11:35:36: Play audio_en/ringback.wav, caching true [8] 2012/07/12 11:35:36: call port 0: state code from 100 to 183 [6] 2012/07/12 11:35:36: Call-leg 0: Codec pcmu/8000 is chosen for call id 16585798691272012113534@95.128.80.120 [5] 2012/07/12 11:35:36: set codec: codec pcmu/8000 is set to call-leg 0 [5] 2012/07/12 11:35:36: SIP Tx tls:95.128.80.120:36212: SIP/2.0 183 Session Progress Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1659423642;alias From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446 To: <sip:0325520053@95.128.80.120>;tag=3f880a4652 Call-ID: 16585798691272012113534@95.128.80.120 CSeq: 1 INVITE Contact: <sip:99999@192.168.1.201:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Type: application/sdp Content-Length: 379 v=0 o=- 1200740305 1200740305 IN IP4 192.168.1.201 s=- c=IN IP4 192.168.1.201 t=0 0 m=audio 50160 RTP/SAVP 0 8 98 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Kr2UKk1H96d5DB/dk6xD1Yz2DPyX0Z7do817kx/q a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:98 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2012/07/12 11:35:36: SIP Rx tls:192.168.1.41:4901: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-f12868f813612b8534cc8634c840c740;rport=5061 From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071 To: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey Call-ID: 451bca7c@pbx CSeq: 7947 PRACK Contact: <sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt>;reg-id=1 Content-Length: 0 [7] 2012/07/12 11:35:36: Call 451bca7c@pbx: Clear last request [4] 2012/07/12 11:35:36: Dropped 10 SRTP packets with wrong MAC [4] 2012/07/12 11:35:38: Dropped 100 SRTP packets with wrong MAC [5] 2012/07/12 11:35:38: SIP Rx tls:192.168.1.41:4901: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-158f5432f392d55d4e2d63bae9ebbf85;rport=5061 From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071 To: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey Call-ID: 451bca7c@pbx CSeq: 7946 INVITE Contact: <sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt>;reg-id=1 User-Agent: snom821/8.4.35 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 439 v=0 o=root 988953686 988953687 IN IP4 192.168.1.41 s=call c=IN IP4 192.168.1.41 t=0 0 m=audio 58908 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:JiC3R+xWuA39wKgFmj6ISEdD6jCZrW5I0ZKYOhum a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [7] 2012/07/12 11:35:38: Call 451bca7c@pbx: Clear last INVITE [6] 2012/07/12 11:35:38: Call-leg 1: Codec pcmu/8000 is chosen for call id 451bca7c@pbx [6] 2012/07/12 11:35:38: Call-leg 1: Sending RTP for 451bca7c@pbx to 192.168.1.41:58908, codec pcmu/8000 [5] 2012/07/12 11:35:38: set codec: codec pcmu/8000 is set to call-leg 1 [5] 2012/07/12 11:35:38: SIP Tx tls:192.168.1.41:4901: ACK sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt SIP/2.0 Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-e2f6ec7271b167a4dba2f36d1e619818;rport From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071 To: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey Call-ID: 451bca7c@pbx CSeq: 7946 ACK Max-Forwards: 70 Contact: <sip:40@192.168.1.201:5061;transport=tls> Content-Length: 0 [7] 2012/07/12 11:35:38: Determine pass-through mode after receiving response [8] 2012/07/12 11:35:38: Call state for call object 1: connected [8] 2012/07/12 11:35:38: call port 1: state code from 100 to 200 [8] 2012/07/12 11:35:38: call port 0: state code from 183 to 200 [5] 2012/07/12 11:35:38: SIP Tx tls:95.128.80.120:36212: SIP/2.0 200 Ok Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1659423642;alias From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446 To: <sip:0325520053@95.128.80.120>;tag=3f880a4652 Call-ID: 16585798691272012113534@95.128.80.120 CSeq: 1 INVITE Contact: <sip:99999@192.168.1.201:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Type: application/sdp Content-Length: 379 v=0 o=- 1200740305 1200740305 IN IP4 192.168.1.201 s=- c=IN IP4 192.168.1.201 t=0 0 m=audio 50160 RTP/SAVP 0 8 98 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Kr2UKk1H96d5DB/dk6xD1Yz2DPyX0Z7do817kx/q a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:98 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [7] 2012/07/12 11:35:38: 16585798691272012113534@95.128.80.120: RTP pass-through mode [7] 2012/07/12 11:35:38: 451bca7c@pbx: RTP pass-through mode [5] 2012/07/12 11:35:39: SIP Rx tls:95.128.80.120:36212: ACK sip:99999@192.168.1.201:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1185360173;alias Max-Forwards: 10 From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446 To: <sip:0325520053@95.128.80.120>;tag=3f880a4652 Call-ID: 16585798691272012113534@95.128.80.120 CSeq: 1 ACK Contact: <sip:96956110529@95.128.80.120:5067;transport=tls> User-Agent: Mediant 1000 - MSBG/v.6.60A.011.001 Content-Length: 0 [8] 2012/07/12 11:35:50: Packet authenticated by transport layer [5] 2012/07/12 11:35:52: SIP Rx tls:192.168.1.41:4901: BYE sip:40@192.168.1.201:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.41:4901;branch=z9hG4bK-hf8mg4lqk05q;rport From: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey To: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071 Call-ID: 451bca7c@pbx CSeq: 1 BYE Max-Forwards: 70 Contact: <sip:40@192.168.1.41:4901;transport=tls;line=2duu6ljt>;reg-id=1 User-Agent: snom821/8.4.35 RTP-RxStat: Total_Rx_Pkts=0,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=652,Tx_Pkts=652,Remote_Tx_Pkts=0 Proxy-Require: buttons Content-Length: 0 [8] 2012/07/12 11:35:52: Packet authenticated by transport layer [5] 2012/07/12 11:35:52: SIP Tx tls:192.168.1.41:4901: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.41:4901;branch=z9hG4bK-hf8mg4lqk05q;rport=4901 From: "Forty Zero" <sip:40@pbx.company.com>;tag=amvok8c4ey To: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=409778071 Call-ID: 451bca7c@pbx CSeq: 1 BYE Contact: <sip:40@192.168.1.201:5061;transport=tls> User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Length: 0 [7] 2012/07/12 11:35:52: 16585798691272012113534@95.128.80.120: Media-aware pass-through mode [8] 2012/07/12 11:35:52: Clearing call port 1, SIP call id 451bca7c@pbx [8] 2012/07/12 11:35:52: call port 0: state code from 200 to 486 [5] 2012/07/12 11:35:52: SIP Tx tls:95.128.80.120:36212: BYE sip:96956110529@95.128.80.120:5067;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-33dee76e478a1d151aa2114665c1c711;rport From: <sip:0325520053@95.128.80.120>;tag=3f880a4652 To: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446 Call-ID: 16585798691272012113534@95.128.80.120 CSeq: 4306 BYE Max-Forwards: 70 Contact: <sip:99999@192.168.1.201:5061;transport=tls> Content-Length: 0 [8] 2012/07/12 11:35:52: Remove leg 2: call port 1, SIP call id 451bca7c@pbx [8] 2012/07/12 11:35:52: Hangup: Call 1 not found [8] 2012/07/12 11:35:52: Last message repeated 2 times [5] 2012/07/12 11:35:52: SIP 95.128.80.120:36212: Alert(2, 20) [4] 2012/07/12 11:35:56: Dropped 1000 SRTP packets with wrong MAC [5] 2012/07/12 11:35:58: SIP Rx tls:95.128.80.120:36212: BYE sip:99999@192.168.1.201:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac2125828410;alias Max-Forwards: 10 From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446 To: <sip:0325520053@95.128.80.120>;tag=3f880a4652 Call-ID: 16585798691272012113534@95.128.80.120 CSeq: 2 BYE User-Agent: Mediant 1000 - MSBG/v.6.60A.011.001 Content-Length: 0 [5] 2012/07/12 11:35:58: SIP Tx tls:95.128.80.120:36212: SIP/2.0 200 Ok Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac2125828410;alias From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446 To: <sip:0325520053@95.128.80.120>;tag=3f880a4652 Call-ID: 16585798691272012113534@95.128.80.120 CSeq: 2 BYE Contact: <sip:99999@192.168.1.201:5061;transport=tls> User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Length: 0 [8] 2012/07/12 11:35:58: Clearing call port 0, SIP call id 16585798691272012113534@95.128.80.120 [8] 2012/07/12 11:35:58: Remove leg 1: call port 0, SIP call id 16585798691272012113534@95.128.80.120 [8] 2012/07/12 11:35:58: Hangup: Call 0 not found [5] 2012/07/12 11:35:58: SIP Rx tls:95.128.80.120:36212: SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.1.201;branch=z9hG4bK-33dee76e478a1d151aa2114665c1c711;received=62.12.196.104;rport=5061 From: <sip:0325520053@95.128.80.120>;tag=3f880a4652 To: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1658692446 Call-ID: 16585798691272012113534@95.128.80.120 CSeq: 4306 BYE Contact: <sip:96956110529@95.128.80.120:5067;transport=tls> Server: Mediant 1000 - MSBG/v.6.60A.011.001 Content-Length: 0

ok I will see if it's possible. and regarding outgoing call, any idea about the problom of "Dropped 1000 SRTP packets with wrong MAC" Laurent

I try it but it's not ok. and phone are in the same LAN that the Snomone , on Internet they is only the M1000 and he is configured to act as a border controller to solve NAT issue. for incoming calls the voice is ok just it seems that the SRTP is not correctly negotiated by the snomone. (see my second post) Laurent

and for Incoming calls, as you can see in the log bellow the M1000 propose the SRTP option in the INVITE SDP but in the 200 OK sendback from the Snomone PBX to the M1000 they is no SRTP proposition. (for incoming calls RTP is working in both way) any idea how to force snomone to use SRTP ? Laurent [8] 2012/07/09 12:23:46: Last message repeated 2 times [5] 2012/07/09 12:23:46: SIP Rx tls:95.128.80.120:54209: INVITE sip:96956110529@95.128.80.120 SIP/2.0 Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1202004276;alias Max-Forwards: 10 From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1201025120 To: <sip:0325520053@95.128.80.120> Call-ID: 1200926091972012122345@95.128.80.120 CSeq: 1 INVITE Contact: <sip:96956110529@95.128.80.120:5067;transport=tls> Allow: ACK,CANCEL,BYE,INFO User-Agent: Mediant 1000 - MSBG/v.6.60A.011.001 Content-Type: application/sdp Content-Length: 759 x-changeuri: 1 v=0 o=Dialogic_SDP 1200505965 1200505930 IN IP4 95.128.80.120 s=Dialogic-SIP c=IN IP4 95.128.80.120 t=0 0 m=audio 8070 RTP/AVP 8 0 98 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:98 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - m=audio 8070 RTP/SAVP 8 0 98 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:98 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pmP4qAsGAw1ojK/EYz6yU076pauMb79BPc95T+iw|2^31|3:1 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:1MkcGr7NlCSAXuecmu3HCWuZUtbfoTfqDqAGkOPX|2^31|82:1 [8] 2012/07/09 12:23:46: Allocating for call port 23, SIP call id 1200926091972012122345@95.128.80.120 [9] 2012/07/09 12:23:46: UDP(IPv4): Opening socket on 0.0.0.0:58306 [9] 2012/07/09 12:23:46: UDP(IPv4): Opening socket on 0.0.0.0:58307 [9] 2012/07/09 12:23:46: UDP(IPv6): Opening socket on [::]:58306 [9] 2012/07/09 12:23:46: UDP(IPv6): Opening socket on [::]:58307 [5] 2012/07/09 12:23:46: Identify trunk (IP address and domain match) 2 [5] 2012/07/09 12:23:46: SIP Tx tls:95.128.80.120:54209: SIP/2.0 100 Trying Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1202004276;alias From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1201025120 To: <sip:0325520053@95.128.80.120>;tag=096ec469b7 Call-ID: 1200926091972012122345@95.128.80.120 CSeq: 1 INVITE Content-Length: 0 [6] 2012/07/09 12:23:46: Call-leg 23: Sending RTP for 1200926091972012122345@95.128.80.120 to 95.128.80.120:8070, codec not set yet [8] 2012/07/09 12:23:46: Incoming call: Request URI sip:96956110529@95.128.80.120, To is <sip:0325520053@95.128.80.120> [8] 2012/07/09 12:23:46: Call from a trunk 2 [8] 2012/07/09 12:23:46: Trunk Peoplefone@pbx.company.com has country code not set, area code not set [9] 2012/07/09 12:23:46: Incoming: formatted From is = "0763770377" <sip:+0763770377@95.128.80.91;user=phone> [9] 2012/07/09 12:23:46: Incoming: formatted To is = <sip:0325520053@95.128.80.120;user=phone> [9] 2012/07/09 12:23:46: Incoming: formatted URI is = sip:96956110529@pbx.company.com;user=phone [8] 2012/07/09 12:23:46: To is <sip:0325520053@95.128.80.120;user=phone>, user 0, domain 1 [8] 2012/07/09 12:23:46: Send call to extension ERE returned 40 [5] 2012/07/09 12:23:46: Domain trunk Peoplefone@pbx.company.com sends call to 40 in domain pbx.company.com [8] 2012/07/09 12:23:46: Set the To domain based on To user 40@pbx.company.com [8] 2012/07/09 12:23:46: Call state for call object 13: idle [7] 2012/07/09 12:23:46: Call port 23: set_codecs for 1200926091972012122345@95.128.80.120 codecs "", codec_preference count 6 [8] 2012/07/09 12:23:46: Call state for call object 13: alerting [8] 2012/07/09 12:23:46: Play audio_moh/noise.wav, caching true [8] 2012/07/09 12:23:46: Allocating for call port 24, SIP call id c580351d@pbx [9] 2012/07/09 12:23:46: UDP(IPv4): Opening socket on 0.0.0.0:50840 [9] 2012/07/09 12:23:46: UDP(IPv4): Opening socket on 0.0.0.0:50841 [9] 2012/07/09 12:23:46: UDP(IPv6): Opening socket on [::]:50840 [9] 2012/07/09 12:23:46: UDP(IPv6): Opening socket on [::]:50841 [7] 2012/07/09 12:23:46: Call port 24: set_codecs for c580351d@pbx codecs "", codec_preference count 6 [9] 2012/07/09 12:23:46: Using outbound proxy sip:192.168.1.41:4041;transport=tls because of flow-label [8] 2012/07/09 12:23:46: call port 24: state code from 0 to 100 [9] 2012/07/09 12:23:46: Call port 24: update_codecs for c580351d@pbx: adding codec pcmu/8000 to available list [9] 2012/07/09 12:23:46: Call port 24: update_codecs for c580351d@pbx: adding codec pcma/8000 to available list [9] 2012/07/09 12:23:46: Call port 24: update_codecs for c580351d@pbx: adding codec g722/8000 to available list [9] 2012/07/09 12:23:46: Call port 24: update_codecs for c580351d@pbx: adding codec g726-32/8000 to available list [9] 2012/07/09 12:23:46: Call port 24: update_codecs for c580351d@pbx: adding codec gsm/8000 to available list [9] 2012/07/09 12:23:46: Call port 24: update_codecs for c580351d@pbx: codec_preference size 6, available codecs size 6 [5] 2012/07/09 12:23:46: SIP Tx tls:192.168.1.41:4041: INVITE sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit SIP/2.0 Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-c4f2c9e63f7663c66ea459f7bca7d152;rport From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984 To: "Forty Zero" <sip:40@pbx.company.com> Call-ID: c580351d@pbx CSeq: 1774 INVITE Max-Forwards: 70 Contact: <sip:40@192.168.1.201:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Alert-Info: <http://127.0.0.1/Bellcore-dr3> Content-Type: application/sdp Content-Length: 423 v=0 o=- 1293832005 1293832005 IN IP4 192.168.1.201 s=- c=IN IP4 192.168.1.201 t=0 0 m=audio 50840 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:hyK1Q5IAZUcPpSQujETIKGmweDxVup6NqNjBQHsR a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [8] 2012/07/09 12:23:46: call port 23: state code from 0 to 100 [9] 2012/07/09 12:23:46: Call port 23: update_codecs for 1200926091972012122345@95.128.80.120: adding codec pcmu/8000 to available list [9] 2012/07/09 12:23:46: Call port 23: update_codecs for 1200926091972012122345@95.128.80.120: adding codec pcma/8000 to available list [9] 2012/07/09 12:23:46: Call port 23: update_codecs for 1200926091972012122345@95.128.80.120: Other side does not support codec g722/8000 [9] 2012/07/09 12:23:46: Call port 23: update_codecs for 1200926091972012122345@95.128.80.120: adding codec g726-32/8000 to available list [9] 2012/07/09 12:23:46: Call port 23: update_codecs for 1200926091972012122345@95.128.80.120: Other side does not support codec gsm/8000 [9] 2012/07/09 12:23:46: Call port 23: update_codecs for 1200926091972012122345@95.128.80.120: codec_preference size 6, available codecs size 4 [5] 2012/07/09 12:23:46: SIP Rx tls:192.168.1.41:4041: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-c4f2c9e63f7663c66ea459f7bca7d152;rport=5061 From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984 To: "Forty Zero" <sip:40@pbx.company.com>;tag=tc04sjvufr Call-ID: c580351d@pbx CSeq: 1774 INVITE Contact: <sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit>;reg-id=1 Content-Length: 0 [9] 2012/07/09 12:23:46: Message repetition, packet dropped [5] 2012/07/09 12:23:46: SIP Rx tls:192.168.1.41:4041: SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-c4f2c9e63f7663c66ea459f7bca7d152;rport=5061 From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984 To: "Forty Zero" <sip:40@pbx.company.com>;tag=tc04sjvufr Call-ID: c580351d@pbx CSeq: 1774 INVITE Contact: <sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit>;reg-id=1 Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Content-Length: 0 [5] 2012/07/09 12:23:46: SIP Tx tls:192.168.1.41:4041: PRACK sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit SIP/2.0 Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-9005154d23fdbcef188a71ccd3fc2287;rport From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984 To: "Forty Zero" <sip:40@pbx.company.com>;tag=tc04sjvufr Call-ID: c580351d@pbx CSeq: 1775 PRACK Max-Forwards: 70 Contact: <sip:40@192.168.1.201:5061;transport=tls> RAck: 1 1774 INVITE Content-Length: 0 [8] 2012/07/09 12:23:46: Play audio_en/ringback.wav, caching true [8] 2012/07/09 12:23:46: call port 23: state code from 100 to 183 [6] 2012/07/09 12:23:46: Call-leg 23: Codec pcmu/8000 is chosen for call id 1200926091972012122345@95.128.80.120 [5] 2012/07/09 12:23:46: set codec: codec pcmu/8000 is set to call-leg 23 [5] 2012/07/09 12:23:46: SIP Tx tls:95.128.80.120:54209: SIP/2.0 183 Session Progress Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1202004276;alias From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1201025120 To: <sip:0325520053@95.128.80.120>;tag=096ec469b7 Call-ID: 1200926091972012122345@95.128.80.120 CSeq: 1 INVITE Contact: <sip:99999@192.168.1.201:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Type: application/sdp Content-Length: 294 v=0 o=- 1705764368 1705764368 IN IP4 192.168.1.201 s=- c=IN IP4 192.168.1.201 t=0 0 m=audio 58306 RTP/AVP 0 8 98 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:98 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2012/07/09 12:23:46: SIP Rx tls:192.168.1.41:4041: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-9005154d23fdbcef188a71ccd3fc2287;rport=5061 From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984 To: "Forty Zero" <sip:40@pbx.company.com>;tag=tc04sjvufr Call-ID: c580351d@pbx CSeq: 1775 PRACK Contact: <sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit>;reg-id=1 Content-Length: 0 [7] 2012/07/09 12:23:46: Call c580351d@pbx: Clear last request [5] 2012/07/09 12:23:48: SIP Rx tls:192.168.1.41:4041: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-c4f2c9e63f7663c66ea459f7bca7d152;rport=5061 From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984 To: "Forty Zero" <sip:40@pbx.company.com>;tag=tc04sjvufr Call-ID: c580351d@pbx CSeq: 1774 INVITE Contact: <sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit>;reg-id=1 User-Agent: snom821/8.4.35 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 439 v=0 o=root 761033547 761033548 IN IP4 192.168.1.41 s=call c=IN IP4 192.168.1.41 t=0 0 m=audio 53224 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:gthWqnKybO3EFXOfhrP0Y2ubFNPmXed6fqdCCVEt a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [7] 2012/07/09 12:23:48: Call c580351d@pbx: Clear last INVITE [6] 2012/07/09 12:23:48: Call-leg 24: Codec pcmu/8000 is chosen for call id c580351d@pbx [6] 2012/07/09 12:23:48: Call-leg 24: Sending RTP for c580351d@pbx to 192.168.1.41:53224, codec pcmu/8000 [5] 2012/07/09 12:23:48: set codec: codec pcmu/8000 is set to call-leg 24 [5] 2012/07/09 12:23:48: SIP Tx tls:192.168.1.41:4041: ACK sip:40@192.168.1.41:4041;transport=tls;line=x1obyfit SIP/2.0 Via: SIP/2.0/TLS 192.168.1.201:5061;branch=z9hG4bK-acfd9fd54aa58f898d9bca5a8194a2c9;rport From: "0763770377" <sip:+0763770377@pbx.company.com;user=phone>;tag=1479292984 To: "Forty Zero" <sip:40@pbx.company.com>;tag=tc04sjvufr Call-ID: c580351d@pbx CSeq: 1774 ACK Max-Forwards: 70 Contact: <sip:40@192.168.1.201:5061;transport=tls> Content-Length: 0 [7] 2012/07/09 12:23:48: Determine pass-through mode after receiving response [8] 2012/07/09 12:23:48: Call state for call object 13: connected [8] 2012/07/09 12:23:48: call port 24: state code from 100 to 200 [8] 2012/07/09 12:23:48: call port 23: state code from 183 to 200 [5] 2012/07/09 12:23:48: SIP Tx tls:95.128.80.120:54209: SIP/2.0 200 Ok Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac1202004276;alias From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1201025120 To: <sip:0325520053@95.128.80.120>;tag=096ec469b7 Call-ID: 1200926091972012122345@95.128.80.120 CSeq: 1 INVITE Contact: <sip:99999@192.168.1.201:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Type: application/sdp Content-Length: 294 v=0 o=- 1705764368 1705764368 IN IP4 192.168.1.201 s=- c=IN IP4 192.168.1.201 t=0 0 m=audio 58306 RTP/AVP 0 8 98 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:98 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [7] 2012/07/09 12:23:48: 1200926091972012122345@95.128.80.120: RTP pass-through mode [7] 2012/07/09 12:23:48: c580351d@pbx: RTP pass-through mode [5] 2012/07/09 12:23:48: SIP Rx tls:95.128.80.120:54209: ACK sip:99999@192.168.1.201:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 95.128.80.120:5067;branch=z9hG4bKac199010494;alias Max-Forwards: 10 From: 0763770377 <sip:+0763770377@95.128.80.91>;tag=1c1201025120 To: <sip:0325520053@95.128.80.120>;tag=096ec469b7 Call-ID: 1200926091972012122345@95.128.80.120 CSeq: 1 ACK Contact: <sip:96956110529@95.128.80.120:5067;transport=tls> User-Agent: Mediant 1000 - MSBG/v.6.60A.011.001 Content-Length: 0

Dear all, I try to setup a trunk from a snomone to an audiocode M1000 with TLS and SRTP. both element are not on the same network, the M1000 has a public IP on internet and the snomone is behind a NAT with a private IP I have defined a SIP proxy with the IP of the M1000 as proxy (I'm doing IP authentication so no Register) the TLs part is OK but I have a problem with RTP/SRTP. I have voice only in one way (only outgoing voice is ok). I have think about a NAT issue but it's not the case as I see RTP in both way between the audiocode and snomone PBX (trace done on snomone server) but I see that snomone is not forwarding the RTP to the final phone. I have checked the log and I see that snomone has blocked RTP : Dropped 1000 SRTP packets with wrong MAC how can I solve this issue ? Best regards Laurent

Hello, Yes I see that the TLS connection can not be started so the GW dont get the SIP message. I have laready tryed to change the trunk type but same problem. how can I debug the TLS negotiation ? Laurent

Hello all, I try to setup a sip trunk from snomone to an audiocode M1000 with tls. we have generated a certificate for the M1000 and we have imported into the snomone the Root certificate. We have done the same process (generation of a certificate) for the snome one as you can see in the log bellow, the snomeone try to initiate the tls layer but it's not working. bellow, log of the snom one (level 9) and attached the wireshark capture tls_snomone_debug.zip any idea ? Laurent [7] 2012/06/21 17:54:50: UDP(IPv6): Opening socket on [::] [8] 2012/06/21 17:54:50: Trunk 2: sending discover message for sips.peoplefone.com [5] 2012/06/21 17:54:50: Set process affinity to 1 [9] 2012/06/21 17:54:50: Resolve 1: discover 95.128.80.120 [8] 2012/06/21 17:54:50: Trunk 2: Received reply for discover method [8] 2012/06/21 17:54:50: Trunk 2 (peoplefone) is associated with the following addresses: 95.128.80.120 [8] 2012/06/21 17:54:50: Trunk peoplefone: Sending registration to sips.peoplefone.com [9] 2012/06/21 17:54:50: Resolve 2: url sip:sips.peoplefone.com [9] 2012/06/21 17:54:50: Resolve 2: naptr sips.peoplefone.com [8] 2012/06/21 17:54:50: DNS: Use DNS server 62.12.130.66 [8] 2012/06/21 17:54:50: DNS: Request sips.peoplefone.com from server 62.12.130.66 [7] 2012/06/21 17:54:50: UDP(IPv4): Opening socket on 0.0.0.0:5060 [8] 2012/06/21 17:54:50: Joined multicast group 224.0.1.75 [7] 2012/06/21 17:54:50: UDP(IPv6): Opening socket on [::]:5060 [7] 2012/06/21 17:54:50: TCP(IPv4): Opening socket on 0.0.0.0:5060 [7] 2012/06/21 17:54:50: TCP(IPv6): Opening socket on [::]:5060 [7] 2012/06/21 17:54:50: TCP(IPv4): Opening socket on 0.0.0.0:5061 [7] 2012/06/21 17:54:50: TCP(IPv6): Opening socket on [::]:5061 [8] 2012/06/21 17:54:51: DNS: Add NAPTR sips.peoplefone.com (ttl=10800) [9] 2012/06/21 17:54:51: DNS: erasing NAPTR sips.peoplefone.com, id 1 retry count 1, [9] 2012/06/21 17:54:51: Resolve 2: naptr sips.peoplefone.com [9] 2012/06/21 17:54:51: Resolve 2: srv tls _sips._tcp.sips.peoplefone.com [8] 2012/06/21 17:54:51: DNS: Request _sips._tcp.sips.peoplefone.com from server 62.12.130.66 [8] 2012/06/21 17:54:51: DNS: Add SRV _sips._tcp.sips.peoplefone.com 4 4 sips.peoplefone.com 5067 (ttl=2400) [9] 2012/06/21 17:54:51: DNS: erasing SRV _sips._tcp.sips.peoplefone.com, id 2 retry count 0, [9] 2012/06/21 17:54:51: Resolve 2: srv tls _sips._tcp.sips.peoplefone.com [9] 2012/06/21 17:54:51: Resolve 2: a tls sips.peoplefone.com 5067 [8] 2012/06/21 17:54:51: DNS: Request sips.peoplefone.com from server 62.12.130.66 [8] 2012/06/21 17:54:51: DNS: Add A sips.peoplefone.com 95.128.80.120 (ttl=2400) [9] 2012/06/21 17:54:51: DNS: erasing A sips.peoplefone.com, id 3 retry count 0, [9] 2012/06/21 17:54:51: Resolve 2: a tls sips.peoplefone.com 5067 [9] 2012/06/21 17:54:51: Resolve 2: tls 95.128.80.120 5067 [8] 2012/06/21 17:54:51: Received SIP connection 1 from 95.128.80.120:5067 [5] 2012/06/21 17:54:51: SIP Tx tls:95.128.80.120:5067: REGISTER sip:sips.peoplefone.com SIP/2.0 Via: SIP/2.0/TLS 192.168.1.46:15865;branch=z9hG4bK-64e099b010dd6030e1d4f0afb4a79f21;rport From: "90543373418" <sip:90543373418@sips.peoplefone.com>;tag=13381 To: "90543373418" <sip:90543373418@sips.peoplefone.com> Call-ID: hfx5bq0f@pbx CSeq: 18149 REGISTER Max-Forwards: 70 Contact: <sip:90543373418@192.168.1.46:15865;transport=tls;line=c81e728d>;+sip.instance="<urn:uuid:30f55c9a-396c-425e-9aa0-f8fe602dd1f2>" User-Agent: snomONE/4.5.0.1075 Delta Aurigids Supported: outbound Expires: 3600 Content-Length: 0 [9] 2012/06/21 17:54:51: SIP 95.128.80.120:5067: send_client_hello(03014fe343cb2fd047c4b9641603a25e2c4ee845c285ccb0d89f3a54fa6ee39ec1a0000004000400050100001c000000180016000013736970732e70656f706c65666f6e652e636f6d) [1] 2012/06/21 17:54:51: TCP: TOS could not be set, code 0 [5] 2012/06/21 17:54:51: SIP 95.128.80.120:5067: Alert(2, 100) [5] 2012/06/21 17:55:00: Last message repeated 2 times [5] 2012/06/21 17:55:00: Table cdrt: Finished reading 6 rows [5] 2012/06/21 17:55:00: Table cdre: Finished reading 15 rows [5] 2012/06/21 17:55:00: Table cdri: Finished reading 6 rows [5] 2012/06/21 17:55:22: Registration on trunk 2 (peoplefone) failed with code 408. Retry in 60 seconds [2] 2012/06/21 17:55:22: Trunk status peoplefone (2) changed to "408 Request Timeout" (Registration failed, retry after 60 seconds) [6] 2012/06/21 17:55:51: SIP TCP/TLS timeout on 95.128.80.120:5067, closing connection [9] 2012/06/21 17:55:51: SIP 95.128.80.120:5067: send_alert(0100) [8] 2012/06/21 17:55:51: Release SIP thread 1 [0] 2012/06/21 17:55:58: Administrator logged in from IP address 127.0.0.1, session 59wqvzj3mbs3wgkbqitx [9] 2012/06/21 17:55:59: Remote site 127.0.0.1 closed the connection [9] 2012/06/21 17:56:06: Last message repeated 3 times [5] 2012/06/21 17:56:06: Could not send 32960 bytes to 127.0.0.1, error code 10054 [9] 2012/06/21 17:56:06: Remote site 127.0.0.1 closed the connection [9] 2012/06/21 17:56:22: Last message repeated 2 times [8] 2012/06/21 17:56:22: Trunk 2: Preparing for re-registration [8] 2012/06/21 17:56:22: Trunk 2: sending discover message for sips.peoplefone.com [9] 2012/06/21 17:56:22: Resolve 3: discover 95.128.80.120 [8] 2012/06/21 17:56:22: Trunk 2: Received reply for discover method [8] 2012/06/21 17:56:22: Trunk 2 (peoplefone) is associated with the following addresses: 95.128.80.120 [8] 2012/06/21 17:56:22: Trunk peoplefone: Sending registration to sips.peoplefone.com [9] 2012/06/21 17:56:22: Resolve 4: url sip:sips.peoplefone.com [9] 2012/06/21 17:56:22: Resolve 4: naptr sips.peoplefone.com [9] 2012/06/21 17:56:22: Resolve 4: srv tls _sips._tcp.sips.peoplefone.com [9] 2012/06/21 17:56:22: Resolve 4: a tls sips.peoplefone.com 5067 [9] 2012/06/21 17:56:22: Resolve 4: tls 95.128.80.120 5067 [8] 2012/06/21 17:56:22: Received SIP connection 2 from 95.128.80.120:5067 [5] 2012/06/21 17:56:22: SIP Tx tls:95.128.80.120:5067: REGISTER sip:sips.peoplefone.com SIP/2.0 Via: SIP/2.0/TLS 192.168.1.46:24771;branch=z9hG4bK-cbfe8b3e1d290c5c811789a9ff69417f;rport From: "90543373418" <sip:90543373418@sips.peoplefone.com>;tag=13381 To: "90543373418" <sip:90543373418@sips.peoplefone.com> Call-ID: hfx5bq0f@pbx CSeq: 18150 REGISTER Max-Forwards: 70 Contact: <sip:90543373418@192.168.1.46:24771;transport=tls;line=c81e728d>;+sip.instance="<urn:uuid:30f55c9a-396c-425e-9aa0-f8fe602dd1f2>" User-Agent: snomONE/4.5.0.1075 Delta Aurigids Supported: outbound Expires: 3600 Content-Length: 0 [9] 2012/06/21 17:56:22: SIP 95.128.80.120:5067: send_client_hello(03014fe34426679aea4ba212708e2e8e2442776eeff79e216839c50ba6a37ac6e8a4000004000400050100001c000000180016000013736970732e70656f706c65666f6e652e636f6d) [1] 2012/06/21 17:56:22: TCP: TOS could not be set, code 0 [5] 2012/06/21 17:56:22: SIP 95.128.80.120:5067: Alert(2, 100) [5] 2012/06/21 17:56:27: Last message repeated 2 times [9] 2012/06/21 17:56:27: Remote site 127.0.0.1 closed the connection [9] 2012/06/21 17:56:54: Last message repeated 2 times [5] 2012/06/21 17:56:54: Registration on trunk 2 (peoplefone) failed with code 408. Retry in 60 seconds [6] 2012/06/21 17:57:22: SIP TCP/TLS timeout on 95.128.80.120:5067, closing connection [9] 2012/06/21 17:57:22: SIP 95.128.80.120:5067: send_alert(0100) [8] 2012/06/21 17:57:22: Release SIP thread 2 [8] 2012/06/21 17:57:54: Trunk 2: Preparing for re-registration [8] 2012/06/21 17:57:54: Trunk 2: sending discover message for sips.peoplefone.com [9] 2012/06/21 17:57:54: Resolve 5: discover 95.128.80.120 [8] 2012/06/21 17:57:54: Trunk 2: Received reply for discover method [8] 2012/06/21 17:57:54: Trunk 2 (peoplefone) is associated with the following addresses: 95.128.80.120 [8] 2012/06/21 17:57:54: Trunk peoplefone: Sending registration to sips.peoplefone.com [9] 2012/06/21 17:57:54: Resolve 6: url sip:sips.peoplefone.com [9] 2012/06/21 17:57:54: Resolve 6: naptr sips.peoplefone.com [9] 2012/06/21 17:57:54: Resolve 6: srv tls _sips._tcp.sips.peoplefone.com [9] 2012/06/21 17:57:54: Resolve 6: a tls sips.peoplefone.com 5067 [9] 2012/06/21 17:57:54: Resolve 6: tls 95.128.80.120 5067 [8] 2012/06/21 17:57:54: Received SIP connection 3 from 95.128.80.120:5067 [5] 2012/06/21 17:57:54: SIP Tx tls:95.128.80.120:5067: REGISTER sip:sips.peoplefone.com SIP/2.0 Via: SIP/2.0/TLS 192.168.1.46:33753;branch=z9hG4bK-d4ee6ce7c65657f6ba23064c0b4d378b;rport From: "90543373418" <sip:90543373418@sips.peoplefone.com>;tag=13381 To: "90543373418" <sip:90543373418@sips.peoplefone.com> Call-ID: hfx5bq0f@pbx CSeq: 18151 REGISTER Max-Forwards: 70 Contact: <sip:90543373418@192.168.1.46:33753;transport=tls;line=c81e728d>;+sip.instance="<urn:uuid:30f55c9a-396c-425e-9aa0-f8fe602dd1f2>" User-Agent: snomONE/4.5.0.1075 Delta Aurigids Supported: outbound Expires: 3600 Content-Length: 0 [9] 2012/06/21 17:57:54: SIP 95.128.80.120:5067: send_client_hello(03014fe34482230c38b414df1934914e2c735fd3f3f9a821212f217fcf08deff4d3a000004000400050100001c000000180016000013736970732e70656f706c65666f6e652e636f6d) [1] 2012/06/21 17:57:54: TCP: TOS could not be set, code 0 [5] 2012/06/21 17:57:54: SIP 95.128.80.120:5067: Alert(2, 100) [5] 2012/06/21 17:58:26: Last message repeated 2 times [5] 2012/06/21 17:58:26: Registration on trunk 2 (peoplefone) failed with code 408. Retry in 60 seconds [6] 2012/06/21 17:58:54: SIP TCP/TLS timeout on 95.128.80.120:5067, closing connection [9] 2012/06/21 17:58:54: SIP 95.128.80.120:5067: send_alert(0100) [8] 2012/06/21 17:58:54: Release SIP thread 3 [8] 2012/06/21 17:59:26: Trunk 2: Preparing for re-registration [8] 2012/06/21 17:59:26: Trunk 2: sending discover message for sips.peoplefone.com [9] 2012/06/21 17:59:26: Resolve 7: discover 95.128.80.120 [8] 2012/06/21 17:59:26: Trunk 2: Received reply for discover method [8] 2012/06/21 17:59:26: Trunk 2 (peoplefone) is associated with the following addresses: 95.128.80.120 [8] 2012/06/21 17:59:26: Trunk peoplefone: Sending registration to sips.peoplefone.com [9] 2012/06/21 17:59:26: Resolve 8: url sip:sips.peoplefone.com [9] 2012/06/21 17:59:26: Resolve 8: naptr sips.peoplefone.com [9] 2012/06/21 17:59:26: Resolve 8: srv tls _sips._tcp.sips.peoplefone.com [9] 2012/06/21 17:59:26: Resolve 8: a tls sips.peoplefone.com 5067 [9] 2012/06/21 17:59:26: Resolve 8: tls 95.128.80.120 5067 [8] 2012/06/21 17:59:26: Received SIP connection 4 from 95.128.80.120:5067 [5] 2012/06/21 17:59:26: SIP Tx tls:95.128.80.120:5067: REGISTER sip:sips.peoplefone.com SIP/2.0 Via: SIP/2.0/TLS 192.168.1.46:42695;branch=z9hG4bK-5d88db26b7512642f482708035915439;rport From: "90543373418" <sip:90543373418@sips.peoplefone.com>;tag=13381 To: "90543373418" <sip:90543373418@sips.peoplefone.com> Call-ID: hfx5bq0f@pbx CSeq: 18152 REGISTER Max-Forwards: 70 Contact: <sip:90543373418@192.168.1.46:42695;transport=tls;line=c81e728d>;+sip.instance="<urn:uuid:30f55c9a-396c-425e-9aa0-f8fe602dd1f2>" User-Agent: snomONE/4.5.0.1075 Delta Aurigids Supported: outbound Expires: 3600 Content-Length: 0 [9] 2012/06/21 17:59:26: SIP 95.128.80.120:5067: send_client_hello(03014fe344de27b514a9f3895056db8d11d13c764c489f406e511a62fc89b447541c000004000400050100001c000000180016000013736970732e70656f706c65666f6e652e636f6d) [1] 2012/06/21 17:59:26: TCP: TOS could not be set, code 0 [5] 2012/06/21 17:59:26: SIP 95.128.80.120:5067: Alert(2, 100) [5] 2012/06/21 17:59:58: Last message repeated 2 times [5] 2012/06/21 17:59:58: Registration on trunk 2 (peoplefone) failed with code 408. Retry in 60 seconds [6] 2012/06/21 18:00:26: SIP TCP/TLS timeout on 95.128.80.120:5067, closing connection [9] 2012/06/21 18:00:26: SIP 95.128.80.120:5067: send_alert(0100) [8] 2012/06/21 18:00:26: Release SIP thread 4 [8] 2012/06/21 18:00:58: Trunk 2: Preparing for re-registration [8] 2012/06/21 18:00:58: Trunk 2: sending discover message for sips.peoplefone.com [9] 2012/06/21 18:00:58: Resolve 9: discover 95.128.80.120 [8] 2012/06/21 18:00:58: Trunk 2: Received reply for discover method [8] 2012/06/21 18:00:58: Trunk 2 (peoplefone) is associated with the following addresses: 95.128.80.120 [8] 2012/06/21 18:00:58: Trunk peoplefone: Sending registration to sips.peoplefone.com [9] 2012/06/21 18:00:58: Resolve 10: url sip:sips.peoplefone.com [9] 2012/06/21 18:00:58: Resolve 10: naptr sips.peoplefone.com [9] 2012/06/21 18:00:58: Resolve 10: srv tls _sips._tcp.sips.peoplefone.com [9] 2012/06/21 18:00:58: Resolve 10: a tls sips.peoplefone.com 5067 [9] 2012/06/21 18:00:58: Resolve 10: tls 95.128.80.120 5067 [8] 2012/06/21 18:00:58: Received SIP connection 5 from 95.128.80.120:5067 [5] 2012/06/21 18:00:58: SIP Tx tls:95.128.80.120:5067: REGISTER sip:sips.peoplefone.com SIP/2.0 Via: SIP/2.0/TLS 192.168.1.46:51681;branch=z9hG4bK-3c0fd8624723f2b9cbce15cb06177aef;rport From: "90543373418" <sip:90543373418@sips.peoplefone.com>;tag=13381 To: "90543373418" <sip:90543373418@sips.peoplefone.com> Call-ID: hfx5bq0f@pbx CSeq: 18153 REGISTER Max-Forwards: 70 Contact: <sip:90543373418@192.168.1.46:51681;transport=tls;line=c81e728d>;+sip.instance="<urn:uuid:30f55c9a-396c-425e-9aa0-f8fe602dd1f2>" User-Agent: snomONE/4.5.0.1075 Delta Aurigids Supported: outbound Expires: 3600 Content-Length: 0 [9] 2012/06/21 18:00:58: SIP 95.128.80.120:5067: send_client_hello(03014fe3453aacb4e33a8202ffa7687d14c384e52a30f0b6495e7f4a001b1c41a054000004000400050100001c000000180016000013736970732e70656f706c65666f6e652e636f6d) [1] 2012/06/21 18:00:58: TCP: TOS could not be set, code 0 [5] 2012/06/21 18:00:58: SIP 95.128.80.120:5067: Alert(2, 100) [5] 2012/06/21 18:01:17: Last message repeated 2 times [9] 2012/06/21 18:01:17: Remote site 127.0.0.1 closed the connection [5] 2012/06/21 18:01:30: Registration on trunk 2 (peoplefone) failed with code 408. Retry in 60 seconds [6] 2012/06/21 18:01:58: SIP TCP/TLS timeout on 95.128.80.120:5067, closing connection [9] 2012/06/21 18:01:58: SIP 95.128.80.120:5067: send_alert(0100) [8] 2012/06/21 18:01:58: Release SIP thread 5 [8] 2012/06/21 18:02:30: Trunk 2: Preparing for re-registration [8] 2012/06/21 18:02:30: Trunk 2: sending discover message for sips.peoplefone.com [9] 2012/06/21 18:02:30: Resolve 11: discover 95.128.80.120 [8] 2012/06/21 18:02:30: Trunk 2: Received reply for discover method [8] 2012/06/21 18:02:30: Trunk 2 (peoplefone) is associated with the following addresses: 95.128.80.120 [8] 2012/06/21 18:02:30: Trunk peoplefone: Sending registration to sips.peoplefone.com [9] 2012/06/21 18:02:30: Resolve 12: url sip:sips.peoplefone.com [9] 2012/06/21 18:02:30: Resolve 12: naptr sips.peoplefone.com [9] 2012/06/21 18:02:30: Resolve 12: srv tls _sips._tcp.sips.peoplefone.com [9] 2012/06/21 18:02:30: Resolve 12: a tls sips.peoplefone.com 5067 [9] 2012/06/21 18:02:30: Resolve 12: tls 95.128.80.120 5067 [8] 2012/06/21 18:02:30: Received SIP connection 6 from 95.128.80.120:5067 [5] 2012/06/21 18:02:30: SIP Tx tls:95.128.80.120:5067: REGISTER sip:sips.peoplefone.com SIP/2.0 Via: SIP/2.0/TLS 192.168.1.46:60645;branch=z9hG4bK-f5504575306277380e1e52c7e4d31347;rport From: "90543373418" <sip:90543373418@sips.peoplefone.com>;tag=13381 To: "90543373418" <sip:90543373418@sips.peoplefone.com> Call-ID: hfx5bq0f@pbx CSeq: 18154 REGISTER Max-Forwards: 70 Contact: <sip:90543373418@192.168.1.46:60645;transport=tls;line=c81e728d>;+sip.instance="<urn:uuid:30f55c9a-396c-425e-9aa0-f8fe602dd1f2>" User-Agent: snomONE/4.5.0.1075 Delta Aurigids Supported: outbound Expires: 3600 Content-Length: 0 [9] 2012/06/21 18:02:30: SIP 95.128.80.120:5067: send_client_hello(03014fe3459692c679951bb16d23f97146dc84949290fc41df14782440f84c4eb86d000004000400050100001c000000180016000013736970732e70656f706c65666f6e652e636f6d) [1] 2012/06/21 18:02:30: TCP: TOS could not be set, code 0 [5] 2012/06/21 18:02:30: SIP 95.128.80.120:5067: Alert(2, 100) [5] 2012/06/21 18:03:02: Last message repeated 2 times [5] 2012/06/21 18:03:02: Registration on trunk 2 (peoplefone) failed with code 408. Retry in 60 seconds

Hello, with the beta , now the led is no more remining red but I see 2 problem. when a phone is ringing the led don't blink on the vision . when a phone is in call and the led is red, the snom phone change the view to display all blf / key event they is key configured. any idea. Laurent

Hello all, I have a snome one configured with snom 821 phone and I try to add a snom vision for BLF. if on the snom vision I configure the button as BLF and as params <sip:USER@SNOMIP;user=phone> then when the USER is bussy the button change to RED but at the end of the call he stay red. if I configure on the snom one web interface a BLF button and I use this button in the vison (with type button and not BLF) then on the vision all is ok (button come RED during a call and he is switched off after the call) but the phone display the vistual key screen and no idea how to disable the display of the virtual screen. regarding the snom visio I have also problem to configure it, many times I change the config from the web interface, select save but nothing is saved !! any idea ? Regards Laurent

Hello all, I have a really strange case ! I have a snom one+ blue with a sonic wall TZ as firewall. Connected to this snom one, I have several snom 821 that are working correctly. I have also a nom meeting point and with this snom meeting point, when I do an outgoing call I have voice only in 1 way ( no incoming voice) but if I do an incoming calls, all is ok. on the TZ , no log message indicating that some RTP traffic is dropped. any Idea? Regards Laurent

Hi, no user blacklisted and I have an autorisation for the full range 130.x.x.x .

Hello all, I have a snomone plus blue with 12 snom 821 phone. the setup was correctly working since 2 month and we disconect 2 phones to move them and now they can't register to the snomone. if we change the config like phone name, they correctly take the new configuration but they can still not register. in the phone log we have: [2] 3/10/2011 17:52:44: SIP: request destination invalid tls:130.x.x.x:5061 4f70263c4542-sri8k165hkae [5] 3/10/2011 17:52:44: SIP: final transport error: 1000002 -> tls:130.x.x.x:5061 [2] 3/10/2011 17:52:44: Transport Error: Pending packet 1000002: generating fake [2] 3/10/2011 17:52:44: Registrar 839@pbx.company.com timed out [5] 3/10/2011 17:53:21: send lldp advertisment [2] 3/10/2011 17:54:16: SIP: request destination invalid tls:130.x.x.x:5061 4f70263c4542-sri8k165hkae [5] 3/10/2011 17:54:16: SIP: final transport error: 1000004 -> we see that the time of the snomone was not correct (hardware time was ok but system time was too late ) so we change it to hardware time but the problem is still the same. other phone are actualy ok but I think if we reboot them we will have same problem with them. any idea ? regards

Hello, How can I send music and not the usual ring tone with a hunt group. I'm using snomone 2011-4.2.1.4025 (Linux) . Regards Laurent

I see that if I create an extension with the mobile destination set , if I want to use them in a agent groups without any real phone, I must set a dummy contact destination in the register page and I must log them manualy ( Administratively login ). but now what will hapends if the PBX is restarted ? is that possible to set destination in a groups without login ? can we use unregistered phone without the need to set dummy regsitration ? Regards Laurent