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Valerio

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Posts posted by Valerio

  3. Smon360 address book & language issue

    in snom Phones

    Posted

    I am still having problems with the address book. It only displays the first 9 entries out of 20. I have even tried upgrading phones to 7.1.33

     

    It have made a trace and it is the PBX that generates an incorrect XML reply to the phone request.

     

    I have attached the network trace and its ascii equivalent. Please remove the .txt from the wireshark pcap file as uploading is not permitted.

     

     

    ----------------

     

    Frame 27 GET /snom/adrbook.xml?user=2&auth=82141aa666e0325758cdb2170d33b003 HTTP/1.1

     

    Host: 192.168.230.3

     

    Accept: */*

     

    Accept-Language: en-us

     

    Connection: Keep-Alive

     

    Keep-Alive: 5

     

    User-Agent: Mozilla/4.0 (compatible; snom360-SIP 7.1.33)

     

     

     

    HTTP/1.1 200 Ok

     

    Date: Tue, 17 Jun 2008 00:00:00 GMT

     

    Content-Type: text/html

     

    Cache-Control: no-cache

     

    Cache-Control: no-store

     

    Server: PBX/2.1.10.2472 (Win32)

     

    Content-Length: 2832

     

     

     

    <?xml version="1.0" encoding="utf-8"?>

    <SnomIPPhoneDirectory>

    <Title>Address Book</Title>

    <Prompt>Prompt</Prompt>

     

    <DirectoryEntry>

    <Name>PAC (pac)</Name>

    <Telephone>pac</Telephone>

    </DirectoryEntry>

     

    <DirectoryEntry>

    <Name>UtenteA (31)</Name>

    <Telephone>31</Telephone>

    </DirectoryEntry>

     

    <DirectoryEntry>

    <Name>UtenteB (32)</Name>

    <Telephone>32</Telephone>

    </DirectoryEntry>

     

    <DirectoryEntry>

    <Name>Alex Amodio (21)</Name>

    <Telephone>21</Telephone>

    </DirectoryEntry>

     

    <DirectoryEntry>

    <Name>Ivo Amodio (20)</Name>

    <Telephone>20</Telephone>

    </DirectoryEntry>

     

    <DirectoryEntry>

    <Name>Nadia Amodio (26)</Name>

    <Telephone>26</Telephone>

    </DirectoryEntry>

     

    <DirectoryEntry>

    <Name>Nicola Barresi (28)</Name>

    <Telephone>28</Telephone>

    </DirectoryEntry>

     

    <DirectoryEntry>

    <Name>Antonella Coppola (23)</Name>

    <Telephone>23</Telephone>

    </DirectoryEntry>

     

    <DirectoryEntry>

    <Name>Pino Coppola (29)</Name>

    <Telephone>29</Telephone>

    </DirectoryEntry>

     

    <SoftKeyItem>

    <Name>*</Name>

    <URL>http://192.168.230.3/snom/adrbook.xml?user=2&auth=82141aa666e0325758cdb2170d33b003&match=</URL>

    </SoftKeyItem>

    <SoftKeyItem>

    <Name>1</Name>

    <URL>http://192.168.230.3/snom/adrbook.xml?user=2&auth=82141aa666e0325758cdb2170d33b003&match=1</URL>

    </SoftKeyItem>

    <SoftKeyItem>

    <Name>2</Name>

    <URL>http://192.168.230.3/snom/adrbook.xml?user=2&auth=82141aa666e0325758cdb2170d33b003&match=2</URL>

    </SoftKeyItem>

    <SoftKeyItem>

    <Name>3</Name>

    <URL>http://192.168.230.3/snom/adrbook.xml?user=2&auth=82141aa666e0325758cdb2170d33b003&match=3</URL>

    </SoftKeyItem>

    <SoftKeyItem>

    <Name>4</Name>

    <URL>http://192.168.230.3/snom/adrbook.xml?user=2&auth=82141aa666e0325758cdb2170d33b003&match=4</URL>

    </SoftKeyItem>

    <SoftKeyItem>

    <Name>5</Name>

    <URL>http://192.168.230.3/snom/adrbook.xml?user=2&auth=82141aa666e0325758cdb2170d33b003&match=5</URL>

    </SoftKeyItem>

    <SoftKeyItem>

    <Name>6</Name>

    <URL>http://192.168.230.3/snom/adrbook.xml?user=2&auth=82141aa666e0325758cdb2170d33b003&match=6</URL>

    </SoftKeyItem>

    <SoftKeyItem>

    <Name>7</Name>

    <URL>http://192.168.230.3/snom/adrbook.xml?user=2&auth=82141aa666e0325758cdb2170d33b003&match=7</URL>

    </SoftKeyItem>

    <SoftKeyItem>

    <Name>8</Name>

    <URL>http://192.168.230.3/snom/adrbook.xml?user=2&auth=82141aa666e0325758cdb2170d33b003&match=8</URL>

    </SoftKeyItem>

    <SoftKeyItem>

    <Name>9</Name>

    <URL>http://192.168.230.3/snom/adrbook.xml?user=2&auth=82141aa666e0325758cdb2170d33b003&match=9</URL>

    </SoftKeyItem>

    <SoftKeyItem>

    <Name>0</Name>

    <URL>http://192.168.230.3/snom/adrbook.xml?user=2&auth=82141aa666e0325758cdb2170d33b003&match=0</URL>

    </SoftKeyItem>

    </SnomIPPhoneDirectory>

    Address_book.txt

    addresso_book.pcap.txt

  4. Passing Parameters to Snom Phones

    in Other best practices

    Posted

    That still does not work, have copied the file "snom_3xx_phone.xml" from one of the MAC directory and copied it to TFTP.

     

    I have removed all the extension specific info and the file only contains

     

    <?xml version="1.0" encoding="utf-8"?>

    <phone-settings>

    <utc_offset perm="RW">3600</utc_offset>

    <timezone perm="RW">ITA+1</timezone>

    <user_server_type idx="1" perm="RW">pbxnsip</user_server_type>

    <dst perm="RW">3600 3.5.7 2:0:0 10.5.7 2:0:0</dst>

    <with_flash perm="RW">off</with_flash>

    <language perm="RW">Italiano</language>

    <web_language perm="RW">Italiano</web_language>

    <tone_scheme perm="RW">Italia</tone_scheme>

    <time_24_format perm="RW">on</time_24_format>

    <date_us_format perm="RW">off</date_us_format>

    <codec_tos perm="RW">184</codec_tos>

    </phone-settings>

     

    I have reset one of the phone to factory default, rebooted and the custom settings are not applied it gets the standard provisionig as per the log file I included in previous post!!!!

     

    Phone firmware is 7.1.30 and pbxnsip is 2.1.10.2474.

     

    Please help.

  5. Passing Parameters to Snom Phones

    in Other best practices

    Posted

    I have tried coping the {MAC} directory under tftp and under http but it does not work!!!!

     

    looking at the phone log I can see that it tries to load a file -{MAC}.xml that according to the order in which it is loaded should contain the custom settings....but the phone gets a http error 302

     

    I have tried creating such file under tftp and http in the main folders and under the {MAC} folders but the phone never find it !!!!

     

    How can I pass custom settings to the phones ??????

     

    please help !!!!!

     

     

    6]24/12/2001 00:00:08: I take GUI language English from: /mnt/snomlang/gui_lang_EN.xml

    [6]24/12/2001 00:00:09: I take WEB language English from: /mnt/snomlang/web_lang_EN.xml

    [5]24/12/2001 00:00:09: read_xml_settings: found dial-plan XML header

    [9]24/12/2001 00:00:17: initiate_pnp_settings[8]24/12/2001 00:00:17: Random Seed: 1925290349

    [8]24/12/2001 00:00:17: Connect host: 127.0.0.1

    [5]24/12/2001 00:00:17: Fetching URL: https://192.168.230.3:443/provisioning/snom...00413292D3C.htm

    [8]24/12/2001 00:00:17: route_pending_packet 1000000: entry=url udp 192.168.230.3:5060

    [8]24/12/2001 00:00:17: route_pending_packet 1000000: entry=udp 192.168.230.3 5060

    [8]24/12/2001 00:00:17: Send Packet 200

    [7]24/12/2001 00:00:18: Conf setup: finished false: code: 200, host: 192.168.230.3:443, file: /provisioning/snom360-000413292D3C.htm

    [5]24/12/2001 00:00:18: read_xml_settings: found setting-files XML header

    [5]24/12/2001 00:00:18: read_xml_settings: added URL: https://192.168.230.3:443/provisioning/snom...l?model=snom360

    [5]24/12/2001 00:00:18: read_xml_settings: added URL: https://192.168.230.3:443/provisioning/-000413292D3C.xml

    [5]24/12/2001 00:00:18: read_xml_settings: added URL: https://192.168.230.3:443/provisioning/snom...00413292D3C.xml

    [5]24/12/2001 00:00:18: read_xml_settings: added URL: http://192.168.230.3:80/provisioning/snom_web_lang.xml

    [5]24/12/2001 00:00:18: read_xml_settings: added URL: http://192.168.230.3:80/provisioning/snom_gui_lang.xml

    [5]24/12/2001 00:00:18: Conf setup: found xml style settings

    [5]24/12/2001 00:00:18: Fetching URL: https://192.168.230.3:443/provisioning/snom...l?model=snom360

    [7]24/12/2001 00:00:18: Conf setup: finished false: code: 200, host: 192.168.230.3:443, file: /provisioning/snom_3xx_phone-000413292D3C.xml?model=snom360

    [5]24/12/2001 00:00:18: read_xml_settings: found phone-settings XML header

    [5]24/12/2001 00:00:18: Conf setup: found xml style settings

     

    [5]24/12/2001 00:00:18: Fetching URL: https://192.168.230.3:443/provisioning/-000413292D3C.xml

    [7]24/12/2001 00:00:19: Conf setup: finished false: code: 302, host: 192.168.230.3:443, file: /provisioning/-000413292D3C.xml

    ^^^^

    [5]24/12/2001 00:00:19: Fetching URL: https://192.168.230.3:443/provisioning/snom...00413292D3C.xml

    [7]24/12/2001 00:00:19: Conf setup: finished false: code: 200, host: 192.168.230.3:443, file: /provisioning/snom_3xx_fkeys-000413292D3C.xml

    [5]24/12/2001 00:00:19: read_xml_settings: found function-keys XML header

    [5]24/12/2001 00:00:19: Conf setup: found xml style settings

    [5]24/12/2001 00:00:19: Fetching URL: http://192.168.230.3:80/provisioning/snom_web_lang.xml

    [7]24/12/2001 00:00:19: Conf setup: finished false: code: 200, host: 192.168.230.3:80, file: /provisioning/snom_web_lang.xml

    [5]24/12/2001 00:00:19: read_xml_settings: found web-languages XML header

    [5]24/12/2001 00:00:19: Conf setup: found xml style settings

    [5]24/12/2001 00:00:19: Fetching URL: http://192.168.230.3:80/provisioning/snom_gui_lang.xml

    [7]24/12/2001 00:00:19: Conf setup: finished false: code: 200, host: 192.168.230.3:80, file: /provisioning/snom_gui_lang.xml

    [5]24/12/2001 00:00:19: read_xml_settings: found gui-languages XML header

    [5]24/12/2001 00:00:19: Conf setup: found xml style settings

    [5]24/12/2001 00:00:19: Fetching URL: http://127.0.0.1/dummy.htm

    [7]24/12/2001 00:00:19: Conf setup: all setting urls attempted

    [7]24/12/2001 00:00:19: Conf setup: finished false: code: 500, host: 127.0.0.1:80, file: /dummy.htm

    [7]24/12/2001 00:00:19: Conf setup: finished true

  7. Passing Parameters to Snom Phones

    in Other best practices

    Posted

    Hi I have the same problem, I need to provide the parameters shown in the picture.

     

    I have tried addind then in the PnP section as shown in the picture but it does not work.

     

    I have tried creating a snom_3xx_phone.xml file and putting it in the TFTP directory and it does not work.

     

    I have tried creating a snom_3xx_phone.xml file and putting it in the HTML directory and the phone gets only the parameters in the file but not all the user info it normally gets during provisioning.

     

    What I am doing wrong ????

     

    thanks,

    Valerio

    post-237-1213291080_thumb.jpg

  8. Smon360 address book & language issue

    in snom Phones

    Posted

    I have reset the phone to factory defaults and reprovisoined them, I have even tried to install the pbx on a new clean machine but the phone always get the first 9 entry of the address book but not the remaining.

     

    About the language issue, the lang files are in place and the phones load them, but I need to select and use the Italian when the phone gets provisoned and not have the phone proviosned with other langages but still defaulting to english.

     

    thanks

  9. Smon360 address book & language issue

    in snom Phones

    Posted

    Hello,

    I have two problem with pbxnsip 2.1.10.2474 e snom 360 fw 7.1.30, I can autoprovision the phones but when I press the directory button on the phone I get only the fist 9 entries of the domain address book (about 20 entries) and the following message in the logs:

     

    2008/05/23 12:18:59: Web Server: File snom/adrbook.xml?user=1&auth=557fa927fedc754de431e50f3688e89e not found

    2008/05/23 12:18:59: Remote site closed the connection

     

    Second problem is with the language, I can provision the phones with the available languanges but I dont know how to set it to Italian on autoproviosion I have to manually select it.

     

     

    Thanks.

    valerio

  11. Trunk problem between OCS and Pbxnsip

    in Microsoft OCS

    Posted

    wow.......it works.........

     

    For your conveinince I am adding the log from the successful call.....so that you can check any possible side effects....

     

    thanks....great support and great product

     

    [7] 2008/01/15 00:34:42: SIP Rx tcp:10.10.10.53:49770:

    INVITE sip:300@10.10.10.202;user=phone SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.53:49770;branch=z9hG4bK-zgpkxwhj1qo0;rport

    From: "3300" <sip:3300@10.10.10.202>;tag=c1madbtcuv

    To: <sip:300@10.10.10.202;user=phone>

    Call-ID: 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF

    CSeq: 1 INVITE

    Max-Forwards: 70

    Contact: <sip:3300@10.10.10.53:49770;transport=tcp;line=ojn9itpa>;flow-id=1

    P-Key-Flags: resolution="31x13", keys="4"

    User-Agent: snomSoft/5.3

    Accept: application/sdp

    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

    Allow-Events: talk, hold, refer

    Supported: timer, 100rel, replaces, callerid

    Session-Expires: 3600;refresher=uas

    Content-Type: application/sdp

    Content-Length: 362

     

    v=0

    o=root 12550 12550 IN IP4 10.10.10.53

    s=call

    c=IN IP4 10.10.10.53

    t=0 0

    m=audio 49292 RTP/AVP 0 8 3 101

    a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:pbOLHpmc0hJJvhfCrYgp4xvN5ERRG11O7OPRWNYc

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=encryption:optional

    a=sendrecv

     

    [7] 2008/01/15 00:34:42: UDP: Opening socket on port 49632

    [7] 2008/01/15 00:34:42: UDP: Opening socket on port 49633

    [8] 2008/01/15 00:34:42: Could not find a trunk (3 trunks)

    [9] 2008/01/15 00:34:42: Using outbound proxy sip:10.10.10.53:49770;transport=tcp because of flow-label

    [9] 2008/01/15 00:34:42: Resolve 39: tcp 10.10.10.53 49770

    [7] 2008/01/15 00:34:42: SIP Tx tcp:10.10.10.53:49770:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 10.10.10.53:49770;branch=z9hG4bK-zgpkxwhj1qo0;rport=49770

    From: "3300" <sip:3300@10.10.10.202>;tag=c1madbtcuv

    To: <sip:300@10.10.10.202;user=phone>;tag=67331960ee

    Call-ID: 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF

    CSeq: 1 INVITE

    Content-Length: 0

     

     

    [7] 2008/01/15 00:34:42: Set packet length to 20

    [6] 2008/01/15 00:34:42: Sending RTP for 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF#67331960ee to 10.10.10.53:49292

    [9] 2008/01/15 00:34:42: Dialplan: Evaluating !^(3[0-9]{2})@.*!sip:\+39081999999\1@\r;user=phone!i against 300@10.10.10.202

    [5] 2008/01/15 00:34:42: Dialplan IDC DP: Match 300@10.10.10.202 to <sip:+39081999999300@srv-dc1.idc.it;user=phone> on trunk OCS Mediation

    [5] 2008/01/15 00:34:42: Charge user 3300 for redirecting calls

    [8] 2008/01/15 00:34:42: Play audio_moh/noise.wav

    [7] 2008/01/15 00:34:42: UDP: Opening socket on port 55444

    [7] 2008/01/15 00:34:42: UDP: Opening socket on port 55445

    [9] 2008/01/15 00:34:42: Resolve 40: url sip:10.10.10.201:5060;transport=tcp

    [9] 2008/01/15 00:34:42: Resolve 40: a tcp 10.10.10.201 5060

    [9] 2008/01/15 00:34:42: Resolve 40: tcp 10.10.10.201 5060

    [7] 2008/01/15 00:34:42: SIP Tx tcp:10.10.10.201:5060:

    INVITE sip:+39081999999300@srv-dc1.idc.it;user=phone SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.202:3249;branch=z9hG4bK-ca0f19c6ab3a7f0a080a8f5841571692;rport

    From: "3300" <sip:3300@10.10.10.202>;tag=14880

    To: <sip:+39081999999300@srv-dc1.idc.it;user=phone>

    Call-ID: bc1d47e0@pbx

    CSeq: 13764 INVITE

    Max-Forwards: 70

    Contact: <sip:Anonymous@10.10.10.202:3249;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2367

    P-Asserted-Identity: "Valerio Capodacqua" <sip:3300@localhost>

    Content-Type: application/sdp

    Content-Length: 214

     

    v=0

    o=- 2237 2237 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 55444 RTP/AVP 8 0 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

     

    [7] 2008/01/15 00:34:42: Set packet length to 20

    [9] 2008/01/15 00:34:42: Resolve 41: tcp 10.10.10.53 49770

    [7] 2008/01/15 00:34:42: SIP Tx tcp:10.10.10.53:49770:

    SIP/2.0 183 Ringing

    Via: SIP/2.0/TCP 10.10.10.53:49770;branch=z9hG4bK-zgpkxwhj1qo0;rport=49770

    From: "3300" <sip:3300@10.10.10.202>;tag=c1madbtcuv

    To: <sip:300@10.10.10.202;user=phone>;tag=67331960ee

    Call-ID: 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF

    CSeq: 1 INVITE

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2367

    Require: 100rel

    RSeq: 1

    Content-Type: application/sdp

    Content-Length: 228

     

    v=0

    o=- 53196 53196 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 49632 RTP/AVP 8 0 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=sendrecv

     

    [7] 2008/01/15 00:34:42: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 100 Trying

    FROM: "3300"<sip:3300@10.10.10.202>;tag=14880

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>

    CSEQ: 13764 INVITE

    CALL-ID: bc1d47e0@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:3249;branch=z9hG4bK-ca0f19c6ab3a7f0a080a8f5841571692;rport

    CONTENT-LENGTH: 0

     

     

    [7] 2008/01/15 00:34:43: SIP Rx tcp:10.10.10.53:49770:

    PRACK sip:3300@10.10.10.202:5160;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.53:49770;branch=z9hG4bK-xo1o7cjsoavd;rport

    From: "3300" <sip:3300@10.10.10.202>;tag=c1madbtcuv

    To: <sip:300@10.10.10.202;user=phone>;tag=67331960ee

    Call-ID: 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF

    CSeq: 2 PRACK

    Max-Forwards: 70

    Contact: <sip:3300@10.10.10.53:49770;transport=tcp;line=ojn9itpa>;flow-id=1

    RAck: 1 1 INVITE

    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

    Allow-Events: talk, hold, refer

    Content-Length: 0

     

     

    [9] 2008/01/15 00:34:43: Resolve 42: tcp 10.10.10.53 49770

    [7] 2008/01/15 00:34:43: SIP Tx tcp:10.10.10.53:49770:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 10.10.10.53:49770;branch=z9hG4bK-xo1o7cjsoavd;rport=49770

    From: "3300" <sip:3300@10.10.10.202>;tag=c1madbtcuv

    To: <sip:300@10.10.10.202;user=phone>;tag=67331960ee

    Call-ID: 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF

    CSeq: 2 PRACK

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    User-Agent: pbxnsip-PBX/2.1.6.2367

    Content-Length: 0

     

     

    [7] 2008/01/15 00:34:43: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 183 Session Progress

    FROM: "3300"<sip:3300@10.10.10.202>;tag=14880

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=cbd4ff715

    CSEQ: 13764 INVITE

    CALL-ID: bc1d47e0@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:3249;branch=z9hG4bK-ca0f19c6ab3a7f0a080a8f5841571692;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

     

    [8] 2008/01/15 00:34:43: Play audio_it/ringback.wav

    [7] 2008/01/15 00:34:43: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 180 Ringing

    FROM: "3300"<sip:3300@10.10.10.202>;tag=14880

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=cbd4ff715

    CSEQ: 13764 INVITE

    CALL-ID: bc1d47e0@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:3249;branch=z9hG4bK-ca0f19c6ab3a7f0a080a8f5841571692;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

     

    [7] 2008/01/15 00:34:43: Last message repeated 2 times

    [9] 2008/01/15 00:34:43: Message repetition, packet dropped

    [7] 2008/01/15 00:34:45: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 180 Ringing

    FROM: "3300"<sip:3300@10.10.10.202>;tag=14880

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=cbd4ff715

    CSEQ: 13764 INVITE

    CALL-ID: bc1d47e0@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:3249;branch=z9hG4bK-ca0f19c6ab3a7f0a080a8f5841571692;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

     

    [9] 2008/01/15 00:34:45: Message repetition, packet dropped

    [7] 2008/01/15 00:34:47: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 200 OK

    FROM: "3300"<sip:3300@10.10.10.202>;tag=14880

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=cbd4ff715

    CSEQ: 13764 INVITE

    CALL-ID: bc1d47e0@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:3249;branch=z9hG4bK-ca0f19c6ab3a7f0a080a8f5841571692;rport

    CONTACT: <sip:srv-dc1.idc.it:5060;transport=Tcp;maddr=10.10.10.201>

    CONTENT-LENGTH: 253

    SUPPORTED: 100rel

    CONTENT-TYPE: application/sdp; charset=utf-8

    ALLOW: UPDATE

    SERVER: RTCC/3.0.0.0 MediationServer

    ALLOW: Ack, Cancel, Bye,Invite

     

    v=0

    o=- 0 0 IN IP4 10.10.10.201

    s=session

    c=IN IP4 10.10.10.201

    b=CT:1000

    t=0 0

    m=audio 62804 RTP/AVP 8 101

    c=IN IP4 10.10.10.201

    a=rtcp:62805

    a=label:Audio

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 2008/01/15 00:34:47: Call bc1d47e0@pbx#14880: Clear last INVITE

    [7] 2008/01/15 00:34:47: Set packet length to 20

    [6] 2008/01/15 00:34:47: Sending RTP for bc1d47e0@pbx#14880 to 10.10.10.201:62804

    [9] 2008/01/15 00:34:47: Resolve 43: aaaa tcp 10.10.10.201 5060

    [9] 2008/01/15 00:34:47: Resolve 43: a tcp 10.10.10.201 5060

    [9] 2008/01/15 00:34:47: Resolve 43: tcp 10.10.10.201 5060

    [7] 2008/01/15 00:34:47: SIP Tx tcp:10.10.10.201:5060:

    ACK sip:srv-dc1.idc.it:5060;transport=Tcp;maddr=10.10.10.201 SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.202:3249;branch=z9hG4bK-40ef049621cb7a6ebdc62db475e994e9;rport

    From: "3300" <sip:3300@10.10.10.202>;tag=14880

    To: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;tag=cbd4ff715

    Call-ID: bc1d47e0@pbx

    CSeq: 13764 ACK

    Max-Forwards: 70

    Contact: <sip:Anonymous@10.10.10.202:3249;transport=tcp>

    P-Asserted-Identity: "Valerio Capodacqua" <sip:3300@localhost>

    Content-Length: 0

     

     

    [7] 2008/01/15 00:34:47: Determine pass-through mode after receiving response

    [9] 2008/01/15 00:34:47: Resolve 44: tcp 10.10.10.53 49770

    [7] 2008/01/15 00:34:47: SIP Tx tcp:10.10.10.53:49770:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 10.10.10.53:49770;branch=z9hG4bK-zgpkxwhj1qo0;rport=49770

    From: "3300" <sip:3300@10.10.10.202>;tag=c1madbtcuv

    To: <sip:300@10.10.10.202;user=phone>;tag=67331960ee

    Call-ID: 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF

    CSeq: 1 INVITE

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2367

    Content-Type: application/sdp

    Content-Length: 228

     

    v=0

    o=- 53196 53196 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 49632 RTP/AVP 8 0 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=sendrecv

     

    [7] 2008/01/15 00:34:47: bc1d47e0@pbx#14880: RTP pass-through mode

    [7] 2008/01/15 00:34:47: 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF#67331960ee: RTP pass-through mode

    [7] 2008/01/15 00:34:47: SIP Rx tcp:10.10.10.53:49770:

    ACK sip:3300@10.10.10.202:5160;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.53:49770;branch=z9hG4bK-0w6xu1xf1l8q;rport

    From: "3300" <sip:3300@10.10.10.202>;tag=c1madbtcuv

    To: <sip:300@10.10.10.202;user=phone>;tag=67331960ee

    Call-ID: 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF

    CSeq: 1 ACK

    Max-Forwards: 70

    Contact: <sip:3300@10.10.10.53:49770;transport=tcp;line=ojn9itpa>;flow-id=1

    Content-Length: 0

     

     

    [7] 2008/01/15 00:34:49: SIP Rx tcp:10.10.10.201:5060:

    INVITE sip:Anonymous@10.10.10.202:3249;transport=tcp SIP/2.0

    FROM: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=cbd4ff715

    TO: <sip:3300@10.10.10.202>;tag=14880

    CSEQ: 1 INVITE

    CALL-ID: bc1d47e0@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK97ed4d1

    CONTACT: <sip:srv-dc1.idc.it:5060;transport=Tcp;maddr=10.10.10.201;ms-opaque=077dec9431d90ce3>

    CONTENT-LENGTH: 265

    SUPPORTED: 100rel

    USER-AGENT: RTCC/3.0.0.0 MediationServer

    CONTENT-TYPE: application/sdp; charset=utf-8

     

    v=0

    o=- 0 0 IN IP4 10.10.10.201

    s=session

    c=IN IP4 10.10.10.201

    b=CT:1000

    t=0 0

    m=audio 62804 RTP/AVP 8 101

    c=IN IP4 10.10.10.201

    a=rtcp:62805

    a=inactive

    a=label:Audio

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 2008/01/15 00:34:49: Set packet length to 20

    [9] 2008/01/15 00:34:49: Resolve 45: tcp 10.10.10.201 5060

    [7] 2008/01/15 00:34:49: SIP Tx tcp:10.10.10.201:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK97ed4d1

    From: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=cbd4ff715

    To: <sip:3300@10.10.10.202>;tag=14880

    Call-ID: bc1d47e0@pbx

    CSeq: 1 INVITE

    Contact: <sip:Anonymous@10.10.10.202:3249;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2367

    Content-Type: application/sdp

    Content-Length: 202

     

    v=0

    o=- 2237 2237 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 55444 RTP/AVP 8 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=inactive

     

    [7] 2008/01/15 00:34:49: bc1d47e0@pbx#14880: Media-aware pass-through mode

    [6] 2008/01/15 00:34:49: Call hold from trunk

    [7] 2008/01/15 00:34:49: SIP Rx tcp:10.10.10.201:5060:

    ACK sip:Anonymous@10.10.10.202:3249;transport=tcp SIP/2.0

    FROM: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=cbd4ff715

    TO: <sip:3300@10.10.10.202>;tag=14880

    CSEQ: 1 ACK

    CALL-ID: bc1d47e0@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK6126e865

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0 MediationServer

     

     

    [7] 2008/01/15 00:34:53: SIP Rx tcp:10.10.10.201:5060:

    INVITE sip:Anonymous@10.10.10.202:3249;transport=tcp SIP/2.0

    FROM: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=cbd4ff715

    TO: <sip:3300@10.10.10.202>;tag=14880

    CSEQ: 2 INVITE

    CALL-ID: bc1d47e0@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK4aa11b2b

    CONTACT: <sip:srv-dc1.idc.it:5060;transport=Tcp;maddr=10.10.10.201;ms-opaque=077dec9431d90ce3>

    CONTENT-LENGTH: 253

    SUPPORTED: 100rel

    USER-AGENT: RTCC/3.0.0.0 MediationServer

    CONTENT-TYPE: application/sdp; charset=utf-8

     

    v=0

    o=- 0 0 IN IP4 10.10.10.201

    s=session

    c=IN IP4 10.10.10.201

    b=CT:1000

    t=0 0

    m=audio 62804 RTP/AVP 8 101

    c=IN IP4 10.10.10.201

    a=rtcp:62805

    a=label:Audio

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 2008/01/15 00:34:53: Set packet length to 20

    [9] 2008/01/15 00:34:53: Resolve 46: tcp 10.10.10.201 5060

    [7] 2008/01/15 00:34:53: SIP Tx tcp:10.10.10.201:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK4aa11b2b

    From: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=cbd4ff715

    To: <sip:3300@10.10.10.202>;tag=14880

    Call-ID: bc1d47e0@pbx

    CSeq: 2 INVITE

    Contact: <sip:Anonymous@10.10.10.202:3249;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2367

    Content-Type: application/sdp

    Content-Length: 202

     

    v=0

    o=- 2237 2237 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 55444 RTP/AVP 8 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=sendrecv

     

    [7] 2008/01/15 00:34:53: bc1d47e0@pbx#14880: RTP pass-through mode

    [6] 2008/01/15 00:34:53: Call hold from trunk

    [7] 2008/01/15 00:34:53: SIP Rx tcp:10.10.10.201:5060:

    ACK sip:Anonymous@10.10.10.202:3249;transport=tcp SIP/2.0

    FROM: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=cbd4ff715

    TO: <sip:3300@10.10.10.202>;tag=14880

    CSEQ: 2 ACK

    CALL-ID: bc1d47e0@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK9a6e7a0

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0 MediationServer

     

     

    [7] 2008/01/15 00:34:54: SIP Rx tcp:10.10.10.201:5060:

    BYE sip:Anonymous@10.10.10.202:3249;transport=tcp SIP/2.0

    FROM: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=cbd4ff715

    TO: <sip:3300@10.10.10.202>;tag=14880

    CSEQ: 3 BYE

    CALL-ID: bc1d47e0@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK54a4b25f

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0 MediationServer

     

     

    [9] 2008/01/15 00:34:54: Resolve 47: tcp 10.10.10.201 5060

    [7] 2008/01/15 00:34:54: SIP Tx tcp:10.10.10.201:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK54a4b25f

    From: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=cbd4ff715

    To: <sip:3300@10.10.10.202>;tag=14880

    Call-ID: bc1d47e0@pbx

    CSeq: 3 BYE

    Contact: <sip:Anonymous@10.10.10.202:3249;transport=tcp>

    User-Agent: pbxnsip-PBX/2.1.6.2367

    RTP-RxStat: Dur=12,Pkt=18,Oct=3096,Underun=0

    RTP-TxStat: Dur=7,Pkt=188,Oct=32168

    Content-Length: 0

     

     

    [7] 2008/01/15 00:34:54: 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF#67331960ee: Media-aware pass-through mode

    [7] 2008/01/15 00:34:54: Other Ports: 1

    [7] 2008/01/15 00:34:54: Call Port: 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF#67331960ee

    [9] 2008/01/15 00:34:54: Resolve 48: url sip:10.10.10.53:49770;transport=tcp

    [9] 2008/01/15 00:34:54: Resolve 48: a tcp 10.10.10.53 49770

    [9] 2008/01/15 00:34:54: Resolve 48: tcp 10.10.10.53 49770

    [7] 2008/01/15 00:34:54: SIP Tx tcp:10.10.10.53:49770:

    BYE sip:3300@10.10.10.53:49770;transport=tcp;line=ojn9itpa SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.202:5160;branch=z9hG4bK-3b09a6ccd72ed5d1f423e5897ed669fe;rport

    From: <sip:300@10.10.10.202;user=phone>;tag=67331960ee

    To: "3300" <sip:3300@10.10.10.202>;tag=c1madbtcuv

    Call-ID: 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF

    CSeq: 31422 BYE

    Max-Forwards: 70

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    RTP-RxStat: Dur=12,Pkt=581,Oct=99932,Underun=0

    RTP-TxStat: Dur=7,Pkt=257,Oct=44204

    Content-Length: 0

     

     

    [7] 2008/01/15 00:34:54: SIP Rx tcp:10.10.10.53:49770:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 10.10.10.202:5160;branch=z9hG4bK-3b09a6ccd72ed5d1f423e5897ed669fe;rport=5160

    From: <sip:300@10.10.10.202;user=phone>;tag=67331960ee

    To: "3300" <sip:3300@10.10.10.202>;tag=c1madbtcuv

    Call-ID: 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF

    CSeq: 31422 BYE

    Contact: <sip:3300@10.10.10.53:49770;transport=tcp;line=ojn9itpa>;flow-id=1

    User-Agent: snomSoft/5.3

    Content-Length: 0

     

     

    [7] 2008/01/15 00:34:54: Call 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF#67331960ee: Clear last request

    [5] 2008/01/15 00:34:54: BYE Response: Terminate 90f18b478847-vob18memj2yl@snomSoft-000413FFFFFF

  12. Trunk problem between OCS and Pbxnsip

    in Microsoft OCS

    Posted

    hello, the new build did not solve the problem

     

    I am attaching a new trace with tihs build and same a call from 3300 (snom softphone) to (300) Office communicator I only trid to put on hold from communicator but same behaviour

     

    [7] 2008/01/14 23:29:06: SIP Rx tcp:10.10.10.110:49302:

    INVITE sip:300@10.10.10.202;user=phone SIP/2.0

    Via: SIP/2.0/TCP 127.0.0.1:49302;branch=z9hG4bK-gpzd3xpxmc3k;rport

    From: "3300" <sip:3300@10.10.10.202>;tag=w529eyt2w6

    To: <sip:300@10.10.10.202;user=phone>

    Call-ID: 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF

    CSeq: 1 INVITE

    Max-Forwards: 70

    Contact: <sip:3300@127.0.0.1:49302;transport=tcp;line=ojn9itpa>;flow-id=1

    P-Key-Flags: resolution="31x13", keys="4"

    User-Agent: snomSoft/5.3

    Accept: application/sdp

    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

    Allow-Events: talk, hold, refer

    Supported: timer, 100rel, replaces, callerid

    Session-Expires: 3600;refresher=uas

    Content-Type: application/sdp

    Content-Length: 358

     

    v=0

    o=root 16036 16036 IN IP4 127.0.0.1

    s=call

    c=IN IP4 127.0.0.1

    t=0 0

    m=audio 63888 RTP/AVP 0 8 3 101

    a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:qkF9dSi9mymDxZnOZbuz1iVJYnbhlBnYqBK6Qooz

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=encryption:optional

    a=sendrecv

     

    [0] 2008/01/14 23:29:06: UDP: Illegal port number

    [7] 2008/01/14 23:29:06: UDP: Opening socket on port 60164

    [7] 2008/01/14 23:29:06: UDP: Opening socket on port 60165

    [8] 2008/01/14 23:29:06: Could not find a trunk (3 trunks)

    [9] 2008/01/14 23:29:06: Using outbound proxy sip:10.10.10.110:49302;transport=tcp because of flow-label

    [9] 2008/01/14 23:29:06: Resolve 63: tcp 10.10.10.110 49302

    [7] 2008/01/14 23:29:06: SIP Tx tcp:10.10.10.110:49302:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 127.0.0.1:49302;branch=z9hG4bK-gpzd3xpxmc3k;rport=49302;received=10.10.10.110

    From: "3300" <sip:3300@10.10.10.202>;tag=w529eyt2w6

    To: <sip:300@10.10.10.202;user=phone>;tag=e47322687a

    Call-ID: 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF

    CSeq: 1 INVITE

    Content-Length: 0

     

     

    [7] 2008/01/14 23:29:06: Set packet length to 20

    [6] 2008/01/14 23:29:06: Sending RTP for 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF#e47322687a to 127.0.0.1:63888

    [9] 2008/01/14 23:29:06: Dialplan: Evaluating !^(3[0-9]{2})@.*!sip:\+39081999999\1@\r;user=phone!i against 300@10.10.10.202

    [5] 2008/01/14 23:29:06: Dialplan IDC DP: Match 300@10.10.10.202 to <sip:+39081999999300@srv-dc1.idc.it;user=phone> on trunk OCS Mediation

    [5] 2008/01/14 23:29:06: Charge user 3300 for redirecting calls

    [8] 2008/01/14 23:29:06: Play audio_moh/noise.wav

    [7] 2008/01/14 23:29:06: UDP: Opening socket on port 57208

    [7] 2008/01/14 23:29:06: UDP: Opening socket on port 57209

    [9] 2008/01/14 23:29:06: Resolve 64: url sip:10.10.10.201:5060;transport=tcp

    [9] 2008/01/14 23:29:06: Resolve 64: a tcp 10.10.10.201 5060

    [9] 2008/01/14 23:29:06: Resolve 64: tcp 10.10.10.201 5060

    [7] 2008/01/14 23:29:06: SIP Tx tcp:10.10.10.201:5060:

    INVITE sip:+39081999999300@srv-dc1.idc.it;user=phone SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.202:1258;branch=z9hG4bK-0fbd0a44dc4b5e376d74a158e068a2ae;rport

    From: "3300" <sip:3300@10.10.10.202>;tag=25139

    To: <sip:+39081999999300@srv-dc1.idc.it;user=phone>

    Call-ID: 2ed7adad@pbx

    CSeq: 3439 INVITE

    Max-Forwards: 70

    Contact: <sip:Anonymous@10.10.10.202:1258;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2366

    P-Asserted-Identity: "Valerio Capodacqua" <sip:3300@localhost>

    Content-Type: application/sdp

    Content-Length: 216

     

    v=0

    o=- 61817 61817 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 57208 RTP/AVP 8 0 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

     

    [7] 2008/01/14 23:29:06: Set packet length to 20

    [9] 2008/01/14 23:29:06: Resolve 65: tcp 10.10.10.110 49302

    [7] 2008/01/14 23:29:06: SIP Tx tcp:10.10.10.110:49302:

    SIP/2.0 183 Ringing

    Via: SIP/2.0/TCP 127.0.0.1:49302;branch=z9hG4bK-gpzd3xpxmc3k;rport=49302;received=10.10.10.110

    From: "3300" <sip:3300@10.10.10.202>;tag=w529eyt2w6

    To: <sip:300@10.10.10.202;user=phone>;tag=e47322687a

    Call-ID: 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF

    CSeq: 1 INVITE

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2366

    Require: 100rel

    RSeq: 1

    Content-Type: application/sdp

    Content-Length: 228

     

    v=0

    o=- 32294 32294 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 60164 RTP/AVP 8 0 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=sendrecv

     

    [7] 2008/01/14 23:29:06: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 100 Trying

    FROM: "3300"<sip:3300@10.10.10.202>;tag=25139

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>

    CSEQ: 3439 INVITE

    CALL-ID: 2ed7adad@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:1258;branch=z9hG4bK-0fbd0a44dc4b5e376d74a158e068a2ae;rport

    CONTENT-LENGTH: 0

     

     

    [8] 2008/01/14 23:29:06: UDP: recvfrom receives ICMP message

    [8] 2008/01/14 23:29:06: Last message repeated 16 times

     

    [7] 2008/01/14 23:29:06: SIP Rx tcp:10.10.10.110:49302:

    PRACK sip:3300@10.10.10.202:5160;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 127.0.0.1:49302;branch=z9hG4bK-xf336f8k2ata;rport

    From: "3300" <sip:3300@10.10.10.202>;tag=w529eyt2w6

    To: <sip:300@10.10.10.202;user=phone>;tag=e47322687a

    Call-ID: 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF

    CSeq: 2 PRACK

    Max-Forwards: 70

    Contact: <sip:3300@127.0.0.1:49302;transport=tcp;line=ojn9itpa>;flow-id=1

    RAck: 1 1 INVITE

    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

    Allow-Events: talk, hold, refer

    Content-Length: 0

     

     

    [9] 2008/01/14 23:29:06: Resolve 66: tcp 10.10.10.110 49302

    [7] 2008/01/14 23:29:06: SIP Tx tcp:10.10.10.110:49302:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 127.0.0.1:49302;branch=z9hG4bK-xf336f8k2ata;rport=49302;received=10.10.10.110

    From: "3300" <sip:3300@10.10.10.202>;tag=w529eyt2w6

    To: <sip:300@10.10.10.202;user=phone>;tag=e47322687a

    Call-ID: 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF

    CSeq: 2 PRACK

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    User-Agent: pbxnsip-PBX/2.1.6.2366

    Content-Length: 0

     

     

    [8] 2008/01/14 23:29:06: UDP: recvfrom receives ICMP message

    [8] 2008/01/14 23:29:06: Last message repeated 3 times

    [6] 2008/01/14 23:29:06: Sending RTP for 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF#e47322687a to 10.10.10.110:63888

    [7] 2008/01/14 23:29:06: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 183 Session Progress

    FROM: "3300"<sip:3300@10.10.10.202>;tag=25139

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=189634db9

    CSEQ: 3439 INVITE

    CALL-ID: 2ed7adad@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:1258;branch=z9hG4bK-0fbd0a44dc4b5e376d74a158e068a2ae;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

     

    [8] 2008/01/14 23:29:06: Play audio_it/ringback.wav

    [7] 2008/01/14 23:29:07: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 180 Ringing

    FROM: "3300"<sip:3300@10.10.10.202>;tag=25139

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=189634db9

    CSEQ: 3439 INVITE

    CALL-ID: 2ed7adad@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:1258;branch=z9hG4bK-0fbd0a44dc4b5e376d74a158e068a2ae;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

     

    [7] 2008/01/14 23:29:07: Last message repeated 2 times

    [9] 2008/01/14 23:29:07: Message repetition, packet dropped

    [7] 2008/01/14 23:29:08: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 180 Ringing

    FROM: "3300"<sip:3300@10.10.10.202>;tag=25139

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=189634db9

    CSEQ: 3439 INVITE

    CALL-ID: 2ed7adad@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:1258;branch=z9hG4bK-0fbd0a44dc4b5e376d74a158e068a2ae;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

     

    [9] 2008/01/14 23:29:08: Message repetition, packet dropped

    [7] 2008/01/14 23:29:11: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 200 OK

    FROM: "3300"<sip:3300@10.10.10.202>;tag=25139

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=189634db9

    CSEQ: 3439 INVITE

    CALL-ID: 2ed7adad@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:1258;branch=z9hG4bK-0fbd0a44dc4b5e376d74a158e068a2ae;rport

    CONTACT: <sip:srv-dc1.idc.it:5060;transport=Tcp;maddr=10.10.10.201>

    CONTENT-LENGTH: 253

    SUPPORTED: 100rel

    CONTENT-TYPE: application/sdp; charset=utf-8

    ALLOW: UPDATE

    SERVER: RTCC/3.0.0.0 MediationServer

    ALLOW: Ack, Cancel, Bye,Invite

     

    v=0

    o=- 0 0 IN IP4 10.10.10.201

    s=session

    c=IN IP4 10.10.10.201

    b=CT:1000

    t=0 0

    m=audio 61400 RTP/AVP 8 101

    c=IN IP4 10.10.10.201

    a=rtcp:61401

    a=label:Audio

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 2008/01/14 23:29:11: Call 2ed7adad@pbx#25139: Clear last INVITE

    [7] 2008/01/14 23:29:11: Set packet length to 20

    [6] 2008/01/14 23:29:11: Sending RTP for 2ed7adad@pbx#25139 to 10.10.10.201:61400

    [9] 2008/01/14 23:29:11: Resolve 67: aaaa tcp 10.10.10.201 5060

    [9] 2008/01/14 23:29:11: Resolve 67: a tcp 10.10.10.201 5060

    [9] 2008/01/14 23:29:11: Resolve 67: tcp 10.10.10.201 5060

    [7] 2008/01/14 23:29:11: SIP Tx tcp:10.10.10.201:5060:

    ACK sip:srv-dc1.idc.it:5060;transport=Tcp;maddr=10.10.10.201 SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.202:1258;branch=z9hG4bK-81b512562598361a57a3e37c08f64947;rport

    From: "3300" <sip:3300@10.10.10.202>;tag=25139

    To: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;tag=189634db9

    Call-ID: 2ed7adad@pbx

    CSeq: 3439 ACK

    Max-Forwards: 70

    Contact: <sip:Anonymous@10.10.10.202:1258;transport=tcp>

    P-Asserted-Identity: "Valerio Capodacqua" <sip:3300@localhost>

    Content-Length: 0

     

     

    [7] 2008/01/14 23:29:11: Determine pass-through mode after receiving response

    [9] 2008/01/14 23:29:11: Resolve 68: tcp 10.10.10.110 49302

    [7] 2008/01/14 23:29:11: SIP Tx tcp:10.10.10.110:49302:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 127.0.0.1:49302;branch=z9hG4bK-gpzd3xpxmc3k;rport=49302;received=10.10.10.110

    From: "3300" <sip:3300@10.10.10.202>;tag=w529eyt2w6

    To: <sip:300@10.10.10.202;user=phone>;tag=e47322687a

    Call-ID: 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF

    CSeq: 1 INVITE

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2366

    Content-Type: application/sdp

    Content-Length: 228

     

    v=0

    o=- 32294 32294 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 60164 RTP/AVP 8 0 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=sendrecv

     

    [7] 2008/01/14 23:29:11: 2ed7adad@pbx#25139: RTP pass-through mode

    [7] 2008/01/14 23:29:11: 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF#e47322687a: RTP pass-through mode

    [7] 2008/01/14 23:29:11: SIP Rx tcp:10.10.10.110:49302:

    ACK sip:3300@10.10.10.202:5160;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 127.0.0.1:49302;branch=z9hG4bK-46s7vp7naexw;rport

    From: "3300" <sip:3300@10.10.10.202>;tag=w529eyt2w6

    To: <sip:300@10.10.10.202;user=phone>;tag=e47322687a

    Call-ID: 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF

    CSeq: 1 ACK

    Max-Forwards: 70

    Contact: <sip:3300@127.0.0.1:49302;transport=tcp;line=ojn9itpa>;flow-id=1

    Content-Length: 0

     

     

    [7] 2008/01/14 23:29:14: SIP Rx tcp:10.10.10.201:5060:

    INVITE sip:Anonymous@10.10.10.202:1258;transport=tcp SIP/2.0

    FROM: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=189634db9

    TO: <sip:3300@10.10.10.202>;tag=25139

    CSEQ: 1 INVITE

    CALL-ID: 2ed7adad@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bKd66c2746

    CONTACT: <sip:srv-dc1.idc.it:5060;transport=Tcp;maddr=10.10.10.201;ms-opaque=077dec9431d90ce3>

    CONTENT-LENGTH: 265

    SUPPORTED: 100rel

    USER-AGENT: RTCC/3.0.0.0 MediationServer

    CONTENT-TYPE: application/sdp; charset=utf-8

     

    v=0

    o=- 0 0 IN IP4 10.10.10.201

    s=session

    c=IN IP4 10.10.10.201

    b=CT:1000

    t=0 0

    m=audio 61400 RTP/AVP 8 101

    c=IN IP4 10.10.10.201

    a=rtcp:61401

    a=inactive

    a=label:Audio

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 2008/01/14 23:29:14: Set packet length to 20

    [9] 2008/01/14 23:29:14: Resolve 69: tcp 10.10.10.201 5060

    [7] 2008/01/14 23:29:14: SIP Tx tcp:10.10.10.201:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bKd66c2746

    From: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=189634db9

    To: <sip:3300@10.10.10.202>;tag=25139

    Call-ID: 2ed7adad@pbx

    CSeq: 1 INVITE

    Contact: <sip:Anonymous@10.10.10.202:1258;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2366

    Content-Type: application/sdp

    Content-Length: 204

     

    v=0

    o=- 61817 61817 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 57208 RTP/AVP 8 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=sendrecv

     

    [7] 2008/01/14 23:29:14: 2ed7adad@pbx#25139: Media-aware pass-through mode

    [6] 2008/01/14 23:29:14: Call hold from trunk

    [7] 2008/01/14 23:29:14: SIP Rx tcp:10.10.10.201:5060:

    ACK sip:Anonymous@10.10.10.202:1258;transport=tcp SIP/2.0

    FROM: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=189634db9

    TO: <sip:3300@10.10.10.202>;tag=25139

    CSEQ: 1 ACK

    CALL-ID: 2ed7adad@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK81c5ea69

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0 MediationServer

     

     

    [7] 2008/01/14 23:29:44: SIP Rx tcp:10.10.10.201:5060:

    BYE sip:Anonymous@10.10.10.202:1258;transport=tcp SIP/2.0

    FROM: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=189634db9

    TO: <sip:3300@10.10.10.202>;tag=25139

    CSEQ: 2 BYE

    CALL-ID: 2ed7adad@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bKbff31acf

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0 MediationServer

     

     

    [9] 2008/01/14 23:29:44: Resolve 70: tcp 10.10.10.201 5060

    [7] 2008/01/14 23:29:44: SIP Tx tcp:10.10.10.201:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bKbff31acf

    From: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=189634db9

    To: <sip:3300@10.10.10.202>;tag=25139

    Call-ID: 2ed7adad@pbx

    CSeq: 2 BYE

    Contact: <sip:Anonymous@10.10.10.202:1258;transport=tcp>

    User-Agent: pbxnsip-PBX/2.1.6.2366

    RTP-RxStat: Dur=38,Pkt=31,Oct=5332,Underun=0

    RTP-TxStat: Dur=33,Pkt=146,Oct=22592

    Content-Length: 0

     

     

    [7] 2008/01/14 23:29:44: 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF#e47322687a: Media-aware pass-through mode

    [7] 2008/01/14 23:29:44: Other Ports: 1

    [7] 2008/01/14 23:29:44: Call Port: 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF#e47322687a

    [9] 2008/01/14 23:29:44: Resolve 71: url sip:10.10.10.110:49302;transport=tcp

    [9] 2008/01/14 23:29:44: Resolve 71: a tcp 10.10.10.110 49302

    [9] 2008/01/14 23:29:44: Resolve 71: tcp 10.10.10.110 49302

    [7] 2008/01/14 23:29:44: SIP Tx tcp:10.10.10.110:49302:

    BYE sip:3300@127.0.0.1:49302;transport=tcp;line=ojn9itpa SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.202:5160;branch=z9hG4bK-bbdfb6986e256f61af841c8adab1d3de;rport

    From: <sip:300@10.10.10.202;user=phone>;tag=e47322687a

    To: "3300" <sip:3300@10.10.10.202>;tag=w529eyt2w6

    Call-ID: 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF

    CSeq: 26897 BYE

    Max-Forwards: 70

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    RTP-RxStat: Dur=38,Pkt=1881,Oct=323532,Underun=0

    RTP-TxStat: Dur=33,Pkt=293,Oct=50396

    Content-Length: 0

     

     

    [7] 2008/01/14 23:29:44: SIP Rx tcp:10.10.10.110:49302:

    SIP/2.0 200 OK

    Via: SIP/2.0/TCP 10.10.10.202:5160;branch=z9hG4bK-bbdfb6986e256f61af841c8adab1d3de;rport=5160

    From: <sip:300@10.10.10.202;user=phone>;tag=e47322687a

    To: "3300" <sip:3300@10.10.10.202>;tag=w529eyt2w6

    Call-ID: 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF

    CSeq: 26897 BYE

    Contact: <sip:3300@127.0.0.1:49302;transport=tcp;line=ojn9itpa>;flow-id=1

    User-Agent: snomSoft/5.3

    Content-Length: 0

     

     

    [7] 2008/01/14 23:29:44: Call 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF#e47322687a: Clear last request

    [5] 2008/01/14 23:29:44: BYE Response: Terminate 33e28b478f31-yicswmynrfel@snomSoft-000413FFFFFF

  13. Trunk problem between OCS and Pbxnsip

    in Microsoft OCS

    Posted

    There is a serious problem if you establish trunk between pbxnsip and ocs.

     

    If you try to put a call on hold from the office communicator client or office communicator phone edition the client reports putting on hold was unsucessfull and just mutes speaker and microphone, you can ear the moh but after a short while the call drops.

     

    here is attached a trace of the issue.

     

    A linksys phone (3300) logged in pbxnsip starts the call to office communicator phone edition (300 translated +39081999999300).

     

    first I put on hold from the linksys phone and everything works ok and I can resume the call correctly, after that I put on hold from the office communicator phone edition and I get the error message (hold unsucessfull) and the call suddenly drops.

     

    Can you help ?

     

     

    [7] 2008/01/14 20:42:27: SIP Rx tcp:10.10.10.140:5073:

    INVITE sip:300@localhost SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.140:5073;branch=z9hG4bK-6ba17d2e

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    To: <sip:300@localhost>

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 101 INVITE

    Max-Forwards: 70

    Contact: "Valerio Capodacqua" <sip:3300@10.10.10.140:5073;transport=tcp>

    Expires: 240

    User-Agent: Linksys/SPA962-5.2.2(SCb)

    Content-Length: 397

    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER

    Supported: 100rel, replaces

    Content-Type: application/sdp

     

    v=0

    o=- 1711557 1711557 IN IP4 10.10.10.140

    s=-

    c=IN IP4 10.10.10.140

    t=0 0

    m=audio 16442 RTP/AVP 0 2 4 8 18 96 97 98 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:2 G726-32/8000

    a=rtpmap:4 G723/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:18 G729a/8000

    a=rtpmap:96 G726-40/8000

    a=rtpmap:97 G726-24/8000

    a=rtpmap:98 G726-16/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=ptime:30

    a=sendrecv

     

    [7] 2008/01/14 20:42:27: UDP: Opening socket on port 64384

    [7] 2008/01/14 20:42:27: UDP: Opening socket on port 64385

    [8] 2008/01/14 20:42:27: Could not find a trunk (3 trunks)

    [9] 2008/01/14 20:42:27: Using outbound proxy sip:10.10.10.140:5073;transport=tcp because of flow-label

    [9] 2008/01/14 20:42:27: Resolve 1118: tcp 10.10.10.140 5073

    [7] 2008/01/14 20:42:27: SIP Tx tcp:10.10.10.140:5073:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 10.10.10.140:5073;branch=z9hG4bK-6ba17d2e

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    To: <sip:300@localhost>;tag=8eba04e153

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 101 INVITE

    Content-Length: 0

     

     

    [7] 2008/01/14 20:42:27: Set packet length to 30

    [6] 2008/01/14 20:42:27: Sending RTP for da2170ee-13f3e42e@10.10.10.140#8eba04e153 to 10.10.10.140:16442

    [9] 2008/01/14 20:42:27: Dialplan: Evaluating !^(3[0-9]{2})@.*!sip:\+39081999999\1@\r;user=phone!i against 300@localhost

    [5] 2008/01/14 20:42:27: Dialplan IDC DP: Match 300@localhost to <sip:+39081999999300@srv-dc1.idc.it;user=phone> on trunk OCS Mediation

    [5] 2008/01/14 20:42:27: Charge user 3300 for redirecting calls

    [8] 2008/01/14 20:42:27: Play audio_moh/noise.wav

    [0] 2008/01/14 20:42:27: UDP: Illegal port number

    [7] 2008/01/14 20:42:27: UDP: Opening socket on port 57912

    [7] 2008/01/14 20:42:27: UDP: Opening socket on port 57913

    [9] 2008/01/14 20:42:27: Resolve 1119: url sip:10.10.10.201:5060;transport=tcp

    [9] 2008/01/14 20:42:27: Resolve 1119: a tcp 10.10.10.201 5060

    [9] 2008/01/14 20:42:27: Resolve 1119: tcp 10.10.10.201 5060

    [7] 2008/01/14 20:42:27: SIP Tx tcp:10.10.10.201:5060:

    INVITE sip:+39081999999300@srv-dc1.idc.it;user=phone SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.202:4754;branch=z9hG4bK-7c5ba84ee32d3ab75a66e5a7ee563540;rport

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=33859

    To: <sip:+39081999999300@srv-dc1.idc.it;user=phone>

    Call-ID: b13069d2@pbx

    CSeq: 21315 INVITE

    Max-Forwards: 70

    Contact: <sip:Anonymous@10.10.10.202:4754;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.5.2357

    P-Asserted-Identity: "Valerio Capodacqua" <sip:3300@localhost>

    Content-Type: application/sdp

    Content-Length: 216

     

    v=0

    o=- 22114 22114 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 57912 RTP/AVP 8 0 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

     

    [7] 2008/01/14 20:42:27: Set packet length to 30

    [9] 2008/01/14 20:42:27: Resolve 1120: tcp 10.10.10.140 5073

    [7] 2008/01/14 20:42:27: SIP Tx tcp:10.10.10.140:5073:

    SIP/2.0 183 Ringing

    Via: SIP/2.0/TCP 10.10.10.140:5073;branch=z9hG4bK-6ba17d2e

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    To: <sip:300@localhost>;tag=8eba04e153

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 101 INVITE

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.5.2357

    Require: 100rel

    RSeq: 1

    Content-Type: application/sdp

    Content-Length: 228

     

    v=0

    o=- 45371 45371 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 64384 RTP/AVP 8 0 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=sendrecv

     

    [7] 2008/01/14 20:42:27: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 100 Trying

    FROM: "Valerio Capodacqua"<sip:3300@localhost>;tag=33859

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>

    CSEQ: 21315 INVITE

    CALL-ID: b13069d2@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:4754;branch=z9hG4bK-7c5ba84ee32d3ab75a66e5a7ee563540;rport

    CONTENT-LENGTH: 0

     

     

    [7] 2008/01/14 20:42:27: SIP Rx tcp:10.10.10.140:5073:

    PRACK sip:3300@10.10.10.202:5160;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.140:5073;branch=z9hG4bK-a6794507

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    To: <sip:300@localhost>;tag=8eba04e153

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 102 PRACK

    Max-Forwards: 70

    Contact: "Valerio Capodacqua" <sip:3300@10.10.10.140:5073;transport=tcp>

    User-Agent: Linksys/SPA962-5.2.2(SCb)

    RAck: 1 101 INVITE

    Content-Length: 0

     

     

    [9] 2008/01/14 20:42:27: Resolve 1121: tcp 10.10.10.140 5073

    [7] 2008/01/14 20:42:27: SIP Tx tcp:10.10.10.140:5073:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 10.10.10.140:5073;branch=z9hG4bK-a6794507

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    To: <sip:300@localhost>;tag=8eba04e153

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 102 PRACK

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    User-Agent: pbxnsip-PBX/2.1.5.2357

    Content-Length: 0

     

     

    [7] 2008/01/14 20:42:27: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 183 Session Progress

    FROM: "Valerio Capodacqua"<sip:3300@localhost>;tag=33859

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=479be4d7db

    CSEQ: 21315 INVITE

    CALL-ID: b13069d2@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:4754;branch=z9hG4bK-7c5ba84ee32d3ab75a66e5a7ee563540;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

     

    [8] 2008/01/14 20:42:27: Play audio_it/ringback.wav

    [7] 2008/01/14 20:42:27: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 180 Ringing

    FROM: "Valerio Capodacqua"<sip:3300@localhost>;tag=33859

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=479be4d7db

    CSEQ: 21315 INVITE

    CALL-ID: b13069d2@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:4754;branch=z9hG4bK-7c5ba84ee32d3ab75a66e5a7ee563540;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0 MediationServer

     

     

    [7] 2008/01/14 20:42:28: Last message repeated 2 times

    [9] 2008/01/14 20:42:28: Message repetition, packet dropped

    [7] 2008/01/14 20:42:29: SIP Rx tcp:10.10.10.201:5060:

    SIP/2.0 200 OK

    FROM: "Valerio Capodacqua"<sip:3300@localhost>;tag=33859

    TO: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=479be4d7db

    CSEQ: 21315 INVITE

    CALL-ID: b13069d2@pbx

    VIA: SIP/2.0/TCP 10.10.10.202:4754;branch=z9hG4bK-7c5ba84ee32d3ab75a66e5a7ee563540;rport

    CONTACT: <sip:srv-dc1.idc.it:5060;transport=Tcp;maddr=10.10.10.201>

    CONTENT-LENGTH: 253

    SUPPORTED: 100rel

    CONTENT-TYPE: application/sdp; charset=utf-8

    ALLOW: UPDATE

    SERVER: RTCC/3.0.0.0 MediationServer

    ALLOW: Ack, Cancel, Bye,Invite

     

    v=0

    o=- 0 0 IN IP4 10.10.10.201

    s=session

    c=IN IP4 10.10.10.201

    b=CT:1000

    t=0 0

    m=audio 63624 RTP/AVP 8 101

    c=IN IP4 10.10.10.201

    a=rtcp:63625

    a=label:Audio

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 2008/01/14 20:42:29: Call b13069d2@pbx#33859: Clear last INVITE

    [7] 2008/01/14 20:42:29: Set packet length to 20

    [6] 2008/01/14 20:42:29: Sending RTP for b13069d2@pbx#33859 to 10.10.10.201:63624

    [9] 2008/01/14 20:42:29: Resolve 1122: aaaa tcp 10.10.10.201 5060

    [9] 2008/01/14 20:42:29: Resolve 1122: a tcp 10.10.10.201 5060

    [9] 2008/01/14 20:42:29: Resolve 1122: tcp 10.10.10.201 5060

    [7] 2008/01/14 20:42:29: SIP Tx tcp:10.10.10.201:5060:

    ACK sip:srv-dc1.idc.it:5060;transport=Tcp;maddr=10.10.10.201 SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.202:4754;branch=z9hG4bK-89d4494676fb2984a533053bedbb3f89;rport

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=33859

    To: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;tag=479be4d7db

    Call-ID: b13069d2@pbx

    CSeq: 21315 ACK

    Max-Forwards: 70

    Contact: <sip:Anonymous@10.10.10.202:4754;transport=tcp>

    P-Asserted-Identity: "Valerio Capodacqua" <sip:3300@localhost>

    Content-Length: 0

     

     

    [7] 2008/01/14 20:42:29: Determine pass-through mode after receiving response

    [9] 2008/01/14 20:42:29: Resolve 1123: tcp 10.10.10.140 5073

    [7] 2008/01/14 20:42:29: SIP Tx tcp:10.10.10.140:5073:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 10.10.10.140:5073;branch=z9hG4bK-6ba17d2e

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    To: <sip:300@localhost>;tag=8eba04e153

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 101 INVITE

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.5.2357

    Content-Type: application/sdp

    Content-Length: 228

     

    v=0

    o=- 45371 45371 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 64384 RTP/AVP 8 0 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=sendrecv

     

    [7] 2008/01/14 20:42:29: b13069d2@pbx#33859: RTP pass-through mode

    [7] 2008/01/14 20:42:29: da2170ee-13f3e42e@10.10.10.140#8eba04e153: RTP pass-through mode

    [7] 2008/01/14 20:42:29: Different packet size (20 and 30), falling back to transcoding

    [7] 2008/01/14 20:42:29: SIP Rx tcp:10.10.10.140:5073:

    ACK sip:3300@10.10.10.202:5160;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.140:5073;branch=z9hG4bK-511ff7d9

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    To: <sip:300@localhost>;tag=8eba04e153

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 101 ACK

    Max-Forwards: 70

    Contact: "Valerio Capodacqua" <sip:3300@10.10.10.140:5073;transport=tcp>

    User-Agent: Linksys/SPA962-5.2.2(SCb)

    Content-Length: 0

     

     

    [7] 2008/01/14 20:42:29: Different packet size (30 and 20), falling back to transcoding

    [7] 2008/01/14 20:42:33: SIP Rx tcp:10.10.10.140:5073:

    INVITE sip:3300@10.10.10.202:5160;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.140:5073;branch=z9hG4bK-891b94ce

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    To: <sip:300@localhost>;tag=8eba04e153

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 103 INVITE

    Max-Forwards: 70

    Contact: "Valerio Capodacqua" <sip:3300@10.10.10.140:5073;transport=tcp>

    Expires: 30

    User-Agent: Linksys/SPA962-5.2.2(SCb)

    Content-Length: 227

    Content-Type: application/sdp

     

    v=0

    o=- 1711557 1711558 IN IP4 10.10.10.140

    s=-

    c=IN IP4 0.0.0.0

    t=0 0

    m=audio 16442 RTP/AVP 0 8 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=ptime:30

    a=sendonly

     

    [7] 2008/01/14 20:42:33: Set packet length to 30

    [9] 2008/01/14 20:42:33: Resolve 1124: tcp 10.10.10.140 5073

    [7] 2008/01/14 20:42:33: SIP Tx tcp:10.10.10.140:5073:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 10.10.10.140:5073;branch=z9hG4bK-891b94ce

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    To: <sip:300@localhost>;tag=8eba04e153

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 103 INVITE

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.5.2357

    Content-Type: application/sdp

    Content-Length: 228

     

    v=0

    o=- 45371 45371 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 64384 RTP/AVP 8 0 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=recvonly

     

    [7] 2008/01/14 20:42:33: SIP Rx tcp:10.10.10.140:5073:

    ACK sip:3300@10.10.10.202:5160;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.140:5073;branch=z9hG4bK-4d2d1462

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    To: <sip:300@localhost>;tag=8eba04e153

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 103 ACK

    Max-Forwards: 70

    Contact: "Valerio Capodacqua" <sip:3300@10.10.10.140:5073;transport=tcp>

    User-Agent: Linksys/SPA962-5.2.2(SCb)

    Content-Length: 0

     

     

    [7] 2008/01/14 20:42:38: SIP Rx tcp:10.10.10.140:5073:

    INVITE sip:3300@10.10.10.202:5160;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.140:5073;branch=z9hG4bK-afe57b9c

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    To: <sip:300@localhost>;tag=8eba04e153

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 104 INVITE

    Max-Forwards: 70

    Contact: "Valerio Capodacqua" <sip:3300@10.10.10.140:5073;transport=tcp>

    Expires: 30

    User-Agent: Linksys/SPA962-5.2.2(SCb)

    Content-Length: 232

    Content-Type: application/sdp

     

    v=0

    o=- 1711557 1711559 IN IP4 10.10.10.140

    s=-

    c=IN IP4 10.10.10.140

    t=0 0

    m=audio 16442 RTP/AVP 0 8 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=ptime:30

    a=sendrecv

     

    [7] 2008/01/14 20:42:38: Set packet length to 30

    [9] 2008/01/14 20:42:38: Resolve 1125: tcp 10.10.10.140 5073

    [7] 2008/01/14 20:42:38: SIP Tx tcp:10.10.10.140:5073:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 10.10.10.140:5073;branch=z9hG4bK-afe57b9c

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    To: <sip:300@localhost>;tag=8eba04e153

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 104 INVITE

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.5.2357

    Content-Type: application/sdp

    Content-Length: 228

     

    v=0

    o=- 45371 45371 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 64384 RTP/AVP 8 0 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=sendrecv

     

    [7] 2008/01/14 20:42:38: da2170ee-13f3e42e@10.10.10.140#8eba04e153: RTP pass-through mode

    [7] 2008/01/14 20:42:38: Different packet size (30 and 20), falling back to transcoding

    [7] 2008/01/14 20:42:38: b13069d2@pbx#33859: RTP pass-through mode

    [7] 2008/01/14 20:42:38: da2170ee-13f3e42e@10.10.10.140#8eba04e153: RTP pass-through mode

    [7] 2008/01/14 20:42:38: Different packet size (30 and 20), falling back to transcoding

    [7] 2008/01/14 20:42:38: SIP Rx tcp:10.10.10.140:5073:

    ACK sip:3300@10.10.10.202:5160;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.140:5073;branch=z9hG4bK-cb90adb0

    From: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    To: <sip:300@localhost>;tag=8eba04e153

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 104 ACK

    Max-Forwards: 70

    Contact: "Valerio Capodacqua" <sip:3300@10.10.10.140:5073;transport=tcp>

    User-Agent: Linksys/SPA962-5.2.2(SCb)

    Content-Length: 0

     

     

    [7] 2008/01/14 20:42:38: Different packet size (20 and 30), falling back to transcoding

    [7] 2008/01/14 20:42:42: SIP Rx tcp:10.10.10.201:5060:

    INVITE sip:Anonymous@10.10.10.202:4754;transport=tcp SIP/2.0

    FROM: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=479be4d7db

    TO: <sip:3300@localhost>;tag=33859

    CSEQ: 1 INVITE

    CALL-ID: b13069d2@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK79d88272

    CONTACT: <sip:srv-dc1.idc.it:5060;transport=Tcp;maddr=10.10.10.201;ms-opaque=077dec9431d90ce3>

    CONTENT-LENGTH: 265

    SUPPORTED: 100rel

    USER-AGENT: RTCC/3.0.0.0 MediationServer

    CONTENT-TYPE: application/sdp; charset=utf-8

     

    v=0

    o=- 0 0 IN IP4 10.10.10.201

    s=session

    c=IN IP4 10.10.10.201

    b=CT:1000

    t=0 0

    m=audio 63624 RTP/AVP 8 101

    c=IN IP4 10.10.10.201

    a=rtcp:63625

    a=inactive

    a=label:Audio

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 2008/01/14 20:42:42: Set packet length to 20

    [9] 2008/01/14 20:42:42: Resolve 1126: tcp 10.10.10.201 5060

    [7] 2008/01/14 20:42:42: SIP Tx tcp:10.10.10.201:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK79d88272

    From: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=479be4d7db

    To: <sip:3300@localhost>;tag=33859

    Call-ID: b13069d2@pbx

    CSeq: 1 INVITE

    Contact: <sip:Anonymous@10.10.10.202:4754;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.5.2357

    Content-Type: application/sdp

    Content-Length: 204

     

    v=0

    o=- 22114 22114 IN IP4 10.10.10.202

    s=-

    c=IN IP4 10.10.10.202

    t=0 0

    m=audio 57912 RTP/AVP 8 101

    a=rtpmap:8 pcma/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=sendrecv

     

    [6] 2008/01/14 20:42:42: Call hold from trunk

    [7] 2008/01/14 20:42:42: SIP Rx tcp:10.10.10.201:5060:

    ACK sip:Anonymous@10.10.10.202:4754;transport=tcp SIP/2.0

    FROM: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=479be4d7db

    TO: <sip:3300@localhost>;tag=33859

    CSEQ: 1 ACK

    CALL-ID: b13069d2@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK8b14f5e8

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0 MediationServer

     

     

    [7] 2008/01/14 20:42:42: SIP Rx tcp:10.10.10.201:5060:

    BYE sip:Anonymous@10.10.10.202:4754;transport=tcp SIP/2.0

    FROM: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=479be4d7db

    TO: <sip:3300@localhost>;tag=33859

    CSEQ: 2 BYE

    CALL-ID: b13069d2@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK5a982dc3

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0 MediationServer

     

     

    [9] 2008/01/14 20:42:42: Resolve 1127: tcp 10.10.10.201 5060

    [7] 2008/01/14 20:42:42: SIP Tx tcp:10.10.10.201:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 10.10.10.201:5060;branch=z9hG4bK5a982dc3

    From: <sip:+39081999999300@srv-dc1.idc.it;user=phone>;epid=9D7B665346;tag=479be4d7db

    To: <sip:3300@localhost>;tag=33859

    Call-ID: b13069d2@pbx

    CSeq: 2 BYE

    Contact: <sip:Anonymous@10.10.10.202:4754;transport=tcp>

    User-Agent: pbxnsip-PBX/2.1.5.2357

    RTP-RxStat: Dur=15,Pkt=621,Oct=106812,Underun=3

    RTP-TxStat: Dur=13,Pkt=628,Oct=108016

    Content-Length: 0

     

     

    [7] 2008/01/14 20:42:42: Other Ports: 1

    [7] 2008/01/14 20:42:42: Call Port: da2170ee-13f3e42e@10.10.10.140#8eba04e153

    [9] 2008/01/14 20:42:42: Resolve 1128: url sip:10.10.10.140:5073;transport=tcp

    [9] 2008/01/14 20:42:42: Resolve 1128: a tcp 10.10.10.140 5073

    [9] 2008/01/14 20:42:42: Resolve 1128: tcp 10.10.10.140 5073

    [7] 2008/01/14 20:42:42: SIP Tx tcp:10.10.10.140:5073:

    BYE sip:3300@10.10.10.140:5073;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 10.10.10.202:5160;branch=z9hG4bK-1e5e71e917d4a0a70fc3c55c2d1dfdcf;rport

    From: <sip:300@localhost>;tag=8eba04e153

    To: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 3140 BYE

    Max-Forwards: 70

    Contact: <sip:3300@10.10.10.202:5160;transport=tcp>

    RTP-RxStat: Dur=15,Pkt=351,Oct=88452,Underun=15

    RTP-TxStat: Dur=13,Pkt=346,Oct=86944

    Content-Length: 0

     

     

    [7] 2008/01/14 20:42:42: SIP Rx tcp:10.10.10.140:5073:

    SIP/2.0 200 OK

    To: "Valerio Capodacqua" <sip:3300@localhost>;tag=d47209ee6bb41f2eo0

    From: <sip:300@localhost>;tag=8eba04e153

    Call-ID: da2170ee-13f3e42e@10.10.10.140

    CSeq: 3140 BYE

    Via: SIP/2.0/TCP 10.10.10.202:5160;branch=z9hG4bK-1e5e71e917d4a0a70fc3c55c2d1dfdcf

    Server: Linksys/SPA962-5.2.2(SCb)

    Content-Length: 0

     

     

    [7] 2008/01/14 20:42:42: Call da2170ee-13f3e42e@10.10.10.140#8eba04e153: Clear last request

    [5] 2008/01/14 20:42:42: BYE Response: Terminate da2170ee-13f3e42e@10.10.10.140

  14. Static Registration

    in Extension Setup

    Posted

    ok.....if there are some changes to be made.....why not implement a checkbox next to the registrations allowing the admin to select the ones he wants to clear ? Doing so might help when is necessary to reset a registration for a phone but leave the static ones in place.

     

    thanks,

    valerio

  15. Static Registration

    in Extension Setup

    Posted

    How can I achive a truly static registration for an external extension ?

     

    I have tried with the "add contact" command in registration tab but on reboot of the pbx the registration goes away, is there a way to make the registration permanent even after reboot ?

  17. OCS Integration - Presence

    in Microsoft OCS

    Posted

    hi,

    working on integration with OCS 2007 I managed to have the Office Communicator client query for presence status of an extension but pnxnsip does not give any info back.

     

    In the veriosn 2.1 docs you say that there is a presence agent, can you plase share more technical details on how it owrks and if it can be queryied from other programs like OC client ?

     

    follows the trace of the presence query for extension 300@pbx.idc.it that I can call from OC without problems

     

    thanks,

    valerio

     

    [5] 2007/10/31 11:17:41: SIP port accept from 10.10.10.205:2871

    [2] 2007/10/31 11:17:41: SIP Rx tcp:10.10.10.205:2871:

    SUBSCRIBE sip:300@pbx.idc.it SIP/2.0

    ms-user-data: ms-publiccloud=true;ms-federation=true

    Via: SIP/2.0/TCP 10.10.10.205:2871;branch=z9hG4bK54C2451C.6173D6DA;branched=FALSE

    Max-Forwards: 69

    From: "Valerio Capodacqua"<sip:valerio_capodacqua@idc.it>;tag=043fff55fb;epid=690fd4933f

    Via: SIP/2.0/TLS 10.10.10.109:2237;ms-received-port=2237;ms-received-cid=3400

    To: <sip:300@pbx.idc.it>

    Call-ID: 69f59f18865c47d89d50c9f4737bf323

    CSeq: 1 SUBSCRIBE

    Contact: <sip:valerio_capodacqua@idc.it;opaque=user:epid:spDQRFEuEVyR_Fg9qEaqjgAA;gruu>

    User-Agent: UCCP/2.0.6362.13 OC/2.0.6362.13 (Microsoft Office Communicator)

    Event: presence

    Accept: application/msrtc-event-categories+xml, application/xpidf+xml, text/xml+msrtc.pidf, application/pidf+xml, application/rlmi+xml, multipart/related

    Supported: com.microsoft.autoextend

    Supported: ms-piggyback-first-notify

    Expires: 0

    Require: adhoclist, categoryList

    Supported: eventlist

    Content-Type: application/msrtc-adrl-categorylist+xml

    Content-Length: 462

     

    <batchSub xmlns="http://schemas.microsoft.com/2006/01/sip/batch-subscribe" uri="sip:valerio_capodacqua@idc.it" name=""><action name="subscribe" id="1171032"><adhocList><resource uri="sip:300@pbx.idc.it"/></adhocList><categoryList xmlns="http://schemas.microsoft.com/2006/09/sip/categorylist"><category name="calendarData"/><category name="contactCard"/><category name="note"/><category name="services"/><category name="state"/></categoryList></action></batchSub>

     

    [2] 2007/10/31 11:17:41: SIP Tx tcp:10.10.10.205:2871:

    SIP/2.0 404 Not Found

    Via: SIP/2.0/TCP 10.10.10.205:2871;branch=z9hG4bK54C2451C.6173D6DA;branched=FALSE

    Via: SIP/2.0/TLS 10.10.10.109:2237;ms-received-port=2237;ms-received-cid=3400

    From: "Valerio Capodacqua" <sip:valerio_capodacqua@idc.it>;epid=690fd4933f;tag=043fff55fb

    To: <sip:300@pbx.idc.it>;tag=ea6a803a49

    Call-ID: 69f59f18865c47d89d50c9f4737bf323

    CSeq: 1 SUBSCRIBE

    Content-Length: 0

  19. OCS 2007 Integration

    in Microsoft OCS

    Posted

    if you assign to the pbxnsip extension the same number as in ocs they will ring simultaneously.

     

    I have the following config:

     

    PSTN <--> PBXNSIP <---> Mediation <---> OCS <--> Exchange UM

     

    it works if you have DID and also if you dont (tried both)......if you dont have DID just let a pbxnsip AA answer and the route calls based on extesnions in ocs create a fake E164 numbering scheme and create dialplan accordingly

     

    if your ocs number is +14255454300 then you ext is 300 so use 300 as extension number in pbxnsip and put as alias your E164 number plus another fake ext. like 400.

     

    if you play with the dialplans in ocs and pbxnsip you will get simultaneous ring on incoming calls from pstn and will be able to call the pbxnsip extensions by using the fake ext #.

     

    the only problem I have is when calling from a OC client to another OC client I cannot get simultaneous ring of the corresponding pbxnsip ext.

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