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laurent

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Posts posted by laurent

  1. SIP trunk with SRTP

    in Trunk Setup

    Posted

    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 ?

  2. SIP trunk with SRTP

    in Trunk Setup

    Posted

    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

  4. SIP trunk with SRTP

    in Trunk Setup

    Posted

    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

  5. SIP trunk with SRTP

    in Trunk Setup

    Posted

    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

  6. SIP trunk with SRTP

    in Trunk Setup

    Posted

    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

  7. sip trunk with TLS

    in Security

    Posted

    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

  8. sip trunk with TLS

    in Security

    Posted

    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

  9. Snom one with S821 and Vision

    in snom Phones

    Posted

    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

  10. Snom one with S821 and Vision

    in snom Phones

    Posted

    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

  11. one way audio only with snom meeting point

    in Firewalls and NAT

    Posted

    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

  13. registration error

    in Extension Setup

    Posted

    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

  15. Administratively login these agents

    in Agent Group Setup

    Posted

    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

  16. Administratively login these agents

    in Agent Group Setup

    Posted

    ok, is that possible to always include an extension (so without doing any login ) ? I have tryed to put them in "All agents for this ACD " but I see that I must do an "Administratively login " so they start to get call.

     

    I also see that we can use the mobile twining with agent group, just I see that it's not working if the extension is not Registered, if the extension is not registered the group say that no agent is available and he don't use the mobile .

     

    Regards

     

    Laurent

  19. IVR patern

    in IVR Node Setup

    Posted

    Hello,

     

    I see on this forum that we can use E or T as input patern but I don't see any info about this in the doc of snomone.

     

    I need to define a destination if a timeout hapends and maybe to have the possibilty to replay the message 1 time and then if again a timeout hapends to send the call to an operator .

     

    Regards

     

    Laurent

  20. Reboot of the SnomOne plus

    in General Setup

    Posted

    Hello,

     

    I have installed a new SnomOne plus (blue) and I see that when I try to do a reboot of the device from the webinterface (webmin hardware interface for the snomOne plus ) then he start the shutdown be he never restart. I must remove the electrical plug to correctly restart them.

     

    one more question, if I ask for a restart of the snom one application, so not from the admin hardware interface for snomone plus, but from usual pbx interface, he also do a complet reboot of the PBX (at lest it seems) ?

     

    Regards

     

    Laurent

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