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Jeremy Salmon

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Posts posted by Jeremy Salmon

  2. snom ONE on Linux : High Availability

    in Best Practices

    Posted

    Hi,

     

    I just installed snom ONE on a hight availability cluster.

     

    My configuration

     

     

    A shared IP between 2 box (ex : 192.168.1.83)

     

    Box 1 (192.168.1.81)
    - Linux Debian 6.x
    - snom ONE with /etc/init.d/snomONE updated to make it LSB compliant
    - DRBD for HDD replication to Box 2
    - Corosync + Pacemaker for failover management
    - Dhcpd


    Box 2 (192.168.1.82)
    - Linux Debian 6.x
    - snom ONE with /etc/init.d/snomONE updated to make it LSB compliant
    - DRBD for HDD replication to Box 1
    - Corosync + Pacemaker for failover management
    - Dhcpd

     

    Patton gateway to connect to Telco.

     

    Everything work well with a Grey (free) édition.

    Services toggle between Box 1 and Box 2 in case of fail of network or anything else.

    Phones and Gateway are registered to 192.168.1.83.

     

    But now I want to activate a Blue licence.

     

    Is there a solution for this type of solution or should I install 2 licenses?

     

    Regards,

     

    Jeremy

     

    PS : A tutorial will be published quickly on http://blog.snom.ma.

  3. snom SOHO : lost config

    in Linux-related Topics

    Posted

    Ok.

     

    I plugged an USB driver and :

     

    apt-get update
apt-get install usbmount
reboot

     

    First USB Key plugged is mounted on /mnt/usb/

     

    mv /usr/local/snomONE/recordings /mnt/usb/
ln -snf /mnt/usb/recordings/ /usr/local/snomONE/recordings
mv /usr/local/snomONE/tftp /mnt/usb/
ln -snf /mnt/usb/tftp/ /usr/local/snomONE/tftp

     

    I also

     

    rm -rf /usr/local/snomONE/audio_en/

     

    (I use french language)

     

    Thanks for your help !

  4. snom SOHO : lost config

    in Linux-related Topics

    Posted

    Thanks for your EXTREMELY quick answer !!!

     

    Is there a method to automatically conserve space on the disk ?

     

    Check if the file system is full (you can see that in the status screen of the PBX). If thats is the case, remove some files and make a backup from the web interface. Dont just reboot, as then your config might really get screwed up.

  5. snom SOHO : lost config

    in Linux-related Topics

    Posted

    Hi,

     

    I have a recurring problem on several snom SOHO.

     

    Sometimes I lose the configuration. The weird thing is that I lose a few pieces (eg extensions or license).

     

    It's very embarrassing ... and this happens to me on several (about 6) SOHO.

     

    Is there a BUG ?

     

    (the same thing happening on SheevaPlug installed with snom ONE)

     

    Regards,

     

    Jeremy

  6. Mutiple GSM Gateway : trunk load balancing

    in Trunk Setup

    Posted

    Hi Mr X,

     

    Thanks for your response.

     

    I know the Failover mecanism of trunk. My problem is when I have no credit on a SIM it's not a fault .... No busy tone or anything else ... Just a voice with "you have no credit to make this call ...."

    I use failover to pass on 2nd, 3rd SIM, ... when previous is busy.

     

    I think I have to dev a small plugin :

    - Make a sum of total call duration on trunk_x between 01/mm/yyy and now()

    - Check if SIM duration > total call duration

    - Update xml of dialplan to remove trunk if not

     

    I think I have to put it on a cron ....

     

    or can I launght an app on each hangup ?

     

    Thanks

  7. Mutiple GSM Gateway : trunk load balancing

    in Trunk Setup

    Posted

    Hi,

     

    I have 4 GSM Gateway with 36 hours of communication on each SIM.

     

    GSM gateway are connected to a Patton SN4114 with one trunk for each port (TRUNK_GSM_1, TRUNK_GSM_2, ...)

     

    I search a solution to make "load balancing" between each Trunk because I can't detect when I have no credit on a SIM card ...

     

    Idea 1 : make call cycling (call 1 on trunk 1, call 2 on trunk 2, etc ...) from snom ONE (not from the Patton)

    Idea 2 : use an "outbound credit" on trunk not extension ... Is it possible ?

     

    Other idea or solution ?

     

    Thanks in advance,

     

    Regards,

     

    Jeremy

  9. One Trunk / Multiple DID / Choose from Dialplan

    in Trunk Setup

    Posted

    hmmm ... Thanks for this idea ...

     

    But not working for me.

     

    I have 9 DID for 9 geographic area ...

     

    In asterisk I wrote Set(CALLERID(num)=33144675643 in my dialplan to rewrite the outgoing callerid ...

     

    In this post you talked about dialplan replacement parameters : http://forum.snomone.com/index.php?/topic/3412-dialplan-replacement/

     

    What is parameter 1,2,3 ???

     

     

    The last solution is to subscribe 9 trunk to my provider ... but it's expensive .... 2 euros for a DID, 12 euros for a trunk ... :(

  11. OVH Trunk - no sound

    in Trunk Setup

    Posted

    For info firmware version of the DIR300 is ME_2.01.

     

    Under "Advanced Setup", "NAT" I have ALG. By default it's activated.

     

    I turned off "SIP Enabled". It seem to work fine.

     

    Thanks for your patience !!!

     

    Just a last question : why SIP Provider IPPI.fr worked and OVH no ? For info OVH have a CIRPACK and Ippi an Asterisk.

     

    Regards,

     

    JS

  12. OVH Trunk - no sound

    in Trunk Setup

    Posted

    It's a DLINK Dir300. But I tried with two others. :(

     

     

     

    Here my Trunk settings:

     

    # Trunk 3 in domain societe4.topsystem.be

    Name: OVH_TRUNK

    Type: register

    To: sip

    RegPass: ********

    Direction:

    Disabled:

    Global: false

    Display:

    RegAccount: 0033184190197

    RegRegistrar: sip.ovh.net

    RegKeep:

    RegUser: 0033184190197

    Icid:

    Require:

    OutboundProxy: sip.ovh.net

    Ani:

    DialExtension:

    Prefix:

    Trusted: false

    AcceptRedirect: false

    RfcRtp: false

    Analog: false

    SendEmail:

    UseUuid: false

    Ring180: false

    Failover: never

    Privacy: false

    Glob:

    RequestTimeout:

    Codecs:

    CodecLock: true

    Expires: 3600

    FromUser:

    Tel: true

    TranscodeDtmf: false

    AssociatedAddresses:

    InterOffice: false

    DialPlan:

    Colines:

    DialogPermission:

  13. OVH Trunk - no sound

    in Trunk Setup

    Posted

    Thanks for your response

     

    192.168.1.1 is my gateway IP.

     

    I have 4 snom 320 in my LAN connected to my snom ONE (192.168.1.13). I can make internal call without problem.

     

    I use an another SIP Provider (ippi.fr) on this server without problem.

     

    I have just sound problem with this OVH provider.

  14. OVH Trunk - no sound

    in Trunk Setup

    Posted

    Hi All,

     

    I have a trunk to OVH SIP Provider (ovh.fr).

     

    Trunk is registered but when I place a call, I ear just the beginning of the first ring and after nothing else ...

     

    I played with trunk option, sip replacement list, ... without success

     

    snom ONE IP : 192.168.1.13

    snom320 IP : 192.168.1.12

    OVH sip server : 91.121.129.17

    Called number : 0972101112

     

    Here my SIP log :

     

    =====

    [5] 2011/12/28 10:28:32: Last message repeated 6 times

    [5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081:

    INVITE sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport

    From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

    To: <sip:0972101112@societe4.topsystem.be;user=phone>

    Call-ID: 3c267cc8ba57-s7m7z0rilz9z

    CSeq: 1 INVITE

    Max-Forwards: 70

    Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1

    X-Serialnumber: 000413318D7F

    P-Key-Flags: keys="3"

    User-Agent: snom320/8.4.18

    Accept: application/sdp

    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

    Session-Expires: 3600;refresher=uas

    Min-SE: 90

    Proxy-Require: buttons

    Content-Type: application/sdp

    Content-Length: 522

     

    v=0

    o=root 469490837 469490837 IN IP4 192.168.1.12

    s=call

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 62652 RTP/AVP 9 0 8 2 3 18 4 101

    a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:fpsUzc8Bo1L9ecyoUesYEHcSpeeJL9NY265vLFVc

    a=rtpmap:9 g722/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:4 g723/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt

    a=sendrecv

    [5] 2011/12/28 10:28:32: Last message repeated 2 times

    [5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport=2081

    From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

    To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

    Call-ID: 3c267cc8ba57-s7m7z0rilz9z

    CSeq: 1 INVITE

    Content-Length: 0

     

    [5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:

    SIP/2.0 401 Authentication Required

    Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport=2081

    From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

    To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

    Call-ID: 3c267cc8ba57-s7m7z0rilz9z

    CSeq: 1 INVITE

    User-Agent: snom-PBX/2011-4.2.0.3981

    WWW-Authenticate: Digest realm="societe4.topsystem.be",nonce="1a6d0303a5486a12117e695eaaaf8663",domain="sip:0972101112@societe4.topsystem.be;user=phone",algorithm=MD5

    Content-Length: 0

     

    [5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081:

    ACK sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport

    From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

    To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

    Call-ID: 3c267cc8ba57-s7m7z0rilz9z

    CSeq: 1 ACK

    Max-Forwards: 70

    Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1

    Proxy-Require: buttons

    Content-Length: 0

     

    [5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081:

    INVITE sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport

    From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

    To: <sip:0972101112@societe4.topsystem.be;user=phone>

    Call-ID: 3c267cc8ba57-s7m7z0rilz9z

    CSeq: 2 INVITE

    Max-Forwards: 70

    Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1

    X-Serialnumber: 000413318D7F

    P-Key-Flags: keys="3"

    User-Agent: snom320/8.4.18

    Accept: application/sdp

    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

    Session-Expires: 3600;refresher=uas

    Min-SE: 90

    Authorization: Digest username="400",realm="societe4.topsystem.be",nonce="1a6d0303a5486a12117e695eaaaf8663",uri="sip:0972101112@societe4.topsystem.be;user=phone",response="1f49712e3eb260cf6b2c006d031e9218",algorithm=MD5

    Proxy-Require: buttons

    Content-Type: application/sdp

    Content-Length: 522

     

    v=0

    o=root 469490837 469490837 IN IP4 192.168.1.12

    s=call

    c=IN IP4 192.168.1.12

    t=0 0

    m=audio 62652 RTP/AVP 9 0 8 2 3 18 4 101

    a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:fpsUzc8Bo1L9ecyoUesYEHcSpeeJL9NY265vLFVc

    a=rtpmap:9 g722/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:4 g723/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt

    a=sendrecv

    [5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081

    From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

    To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

    Call-ID: 3c267cc8ba57-s7m7z0rilz9z

    CSeq: 2 INVITE

    Content-Length: 0

     

    [5] 2011/12/28 10:28:32: SIP Tx udp:91.121.129.17:5060:

    INVITE sip:0972101112@91.121.129.17;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8e16178196f7d0955897860527ca4960;rport

    From: <sip:0033184190197@91.121.129.17>;tag=1119664557

    To: <sip:0972101112@91.121.129.17;user=phone>

    Call-ID: 0f4287ed@pbx

    CSeq: 12053 INVITE

    Max-Forwards: 70

    Contact: <sip:0033184190197@192.168.1.13:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.0.3981

    Remote-Party-ID: "Standardiste" <sip:400@societe4.topsystem.be;user=phone>;party=calling;screen=yes

    Content-Type: application/sdp

    Content-Length: 323

     

    v=0

    o=- 801070725 801070725 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 60346 RTP/AVP 0 8 3 2 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    [5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081

    From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

    To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

    Call-ID: 3c267cc8ba57-s7m7z0rilz9z

    CSeq: 2 INVITE

    Contact: <sip:400@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.0.3981

    Require: 100rel

    RSeq: 1

    Content-Type: application/sdp

    Content-Length: 298

     

    v=0

    o=- 1825812257 1825812257 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 57150 RTP/AVP 0 8 3 101

    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=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    [5] 2011/12/28 10:28:32: SIP Rx udp:91.121.129.17:5060:

    SIP/2.0 407 authentication required

    Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8e16178196f7d0955897860527ca4960

    Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

    From: <sip:0033184190197@91.121.129.17>;tag=1119664557

    To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a364c-25053b736

    Call-ID: 0f4287ed@pbx

    CSeq: 12053 INVITE

    Contact: <sip:0972101112@41.141.84.105:5060;user=phone>

    Proxy-Authenticate: Digest realm="sip.ovh.net", nonce="001a360d6b5d399f4797a37b635627ac", opaque="0010abd225acd41", stale=false, algorithm=MD5

    server: Cirpack/v4.42j (gw_sip)

    Allow: UPDATE, REFER, INFO

    Content-Length: 0

     

    [5] 2011/12/28 10:28:32: SIP Tx udp:91.121.129.17:5060:

    ACK sip:0972101112@91.121.129.17;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8e16178196f7d0955897860527ca4960;rport

    From: <sip:0033184190197@91.121.129.17>;tag=1119664557

    To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a364c-25053b736

    Call-ID: 0f4287ed@pbx

    CSeq: 12053 ACK

    Max-Forwards: 70

    Content-Length: 0

     

    [5] 2011/12/28 10:28:32: SIP Tx udp:91.121.129.17:5060:

    INVITE sip:0972101112@91.121.129.17;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport

    From: <sip:0033184190197@91.121.129.17>;tag=1119664557

    To: <sip:0972101112@91.121.129.17;user=phone>

    Call-ID: 0f4287ed@pbx

    CSeq: 12054 INVITE

    Max-Forwards: 70

    Contact: <sip:0033184190197@192.168.1.13:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.0.3981

    Remote-Party-ID: "Standardiste" <sip:400@societe4.topsystem.be;user=phone>;party=calling;screen=yes

    Proxy-Authorization: Digest realm="sip.ovh.net",nonce="001a360d6b5d399f4797a37b635627ac",response="16890c1b4a1beaf825a0f2e3fd1234fc",username="0033184190197",uri="sip:0972101112@91.121.129.17;user=phone",opaque="0010abd225acd41",algorithm=MD5

    Content-Type: application/sdp

    Content-Length: 323

     

    v=0

    o=- 801070725 801070725 IN IP4 192.168.1.13

    s=-

    c=IN IP4 192.168.1.13

    t=0 0

    m=audio 60346 RTP/AVP 0 8 3 2 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    [5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081:

    PRACK sip:400@192.168.1.13:5060;transport=tcp SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-ckw8xvotqm55;rport

    From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

    To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

    Call-ID: 3c267cc8ba57-s7m7z0rilz9z

    CSeq: 3 PRACK

    Max-Forwards: 70

    Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1

    RAck: 1 2 INVITE

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

    Allow-Events: talk, hold, refer, call-info

    Proxy-Require: buttons

    Content-Length: 0

     

    [5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-ckw8xvotqm55;rport=2081

    From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

    To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

    Call-ID: 3c267cc8ba57-s7m7z0rilz9z

    CSeq: 3 PRACK

    Contact: <sip:400@192.168.1.13:5060;transport=tcp>

    User-Agent: snom-PBX/2011-4.2.0.3981

    Content-Length: 0

     

    [5] 2011/12/28 10:28:32: SIP Rx udp:91.121.129.17:5060:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39

    Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

    From: <sip:0033184190197@91.121.129.17>;tag=1119664557

    To: <sip:0972101112@91.121.129.17;user=phone>

    Call-ID: 0f4287ed@pbx

    CSeq: 12054 INVITE

    Contact: <sip:41.141.84.105:5060>

    server: Cirpack/v4.42j (gw_sip)

    Allow: UPDATE, REFER, INFO

    Content-Length: 0

     

    [5] 2011/12/28 10:28:33: SIP Rx udp:91.121.129.17:5060:

    SIP/2.0 180 Ringing

    Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39

    Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

    From: <sip:0033184190197@91.121.129.17>;tag=1119664557

    To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090

    Call-ID: 0f4287ed@pbx

    CSeq: 12054 INVITE

    Contact: <sip:41.141.84.105:5060>

    server: Cirpack/v4.42j (gw_sip)

    Allow: UPDATE, REFER, INFO

    Content-Length: 0

     

    [5] 2011/12/28 10:28:33: SIP Rx udp:91.121.129.17:5060:

    SIP/2.0 183 Media change

    Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39

    Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

    From: <sip:0033184190197@91.121.129.17>;tag=1119664557

    To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090

    Call-ID: 0f4287ed@pbx

    CSeq: 12054 INVITE

    Contact: <sip:41.141.84.105:5060>

    server: Cirpack/v4.42j (gw_sip)

    Allow: UPDATE, REFER, INFO

    Content-Type: application/sdp

    Content-Length: 262

     

    v=0

    o=cp10 132506811328 132506811330 IN IP4 192.168.1.1

    s=SIP Call

    c=IN IP4 192.168.1.1

    t=0 0

    m=audio 7072 RTP/AVP 0 8 101

    b=AS:82

    a=rtpmap:0 PCMU/8000/1

    a=rtpmap:8 PCMA/8000/1

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=ptime:20

    a=sendrecv

    [5] 2011/12/28 10:28:33: SIP Rx udp:91.121.129.17:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-05b06979574af555d61e176134a2f476

    Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

    From: <sip:0033184190197@91.121.129.17>;tag=1452189312

    To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-07824-001a1068-6632914c5

    Call-ID: ad9d1f30@pbx

    CSeq: 18918 INVITE

    Contact: <sip:41.141.84.105:5060>

    p-asserted-identity: <sip:0972101112@91.121.129.17;user=phone>

    server: Cirpack/v4.42j (gw_sip)

    Allow: UPDATE, REFER, INFO

    Content-Type: application/sdp

    Content-Length: 262

     

    v=0

    o=cp10 132506808498 132506808500 IN IP4 192.168.1.1

    s=SIP Call

    c=IN IP4 192.168.1.1

    t=0 0

    m=audio 7070 RTP/AVP 0 8 101

    b=AS:82

    a=rtpmap:0 PCMU/8000/1

    a=rtpmap:8 PCMA/8000/1

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=ptime:20

    a=sendrecv

    [5] 2011/12/28 10:28:34: SIP Rx tcp:192.168.1.12:2081:

    CANCEL sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport

    From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

    To: <sip:0972101112@societe4.topsystem.be;user=phone>

    Call-ID: 3c267cc8ba57-s7m7z0rilz9z

    CSeq: 2 CANCEL

    Max-Forwards: 70

    Reason: SIP;cause=487;text="Request terminated by user"

    Proxy-Require: buttons

    Content-Length: 0

     

    [5] 2011/12/28 10:28:34: SIP Tx tcp:192.168.1.12:2081:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081

    From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

    To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

    Call-ID: 3c267cc8ba57-s7m7z0rilz9z

    CSeq: 2 CANCEL

    Contact: <sip:400@192.168.1.13:5060;transport=tcp>

    User-Agent: snom-PBX/2011-4.2.0.3981

    Content-Length: 0

     

    [5] 2011/12/28 10:28:34: SIP Tx tcp:192.168.1.12:2081:

    SIP/2.0 487 Request Terminated

    Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081

    From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

    To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

    Call-ID: 3c267cc8ba57-s7m7z0rilz9z

    CSeq: 2 INVITE

    Contact: <sip:400@192.168.1.13:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.0.3981

    Content-Length: 0

     

    [5] 2011/12/28 10:28:34: SIP Tx udp:91.121.129.17:5060:

    CANCEL sip:0972101112@91.121.129.17;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport

    From: <sip:0033184190197@91.121.129.17>;tag=1119664557

    To: <sip:0972101112@91.121.129.17;user=phone>

    Call-ID: 0f4287ed@pbx

    CSeq: 12054 CANCEL

    Max-Forwards: 70

    Remote-Party-ID: "Standardiste" <sip:400@societe4.topsystem.be;user=phone>;party=calling;screen=yes

    Content-Length: 0

     

    [5] 2011/12/28 10:28:34: SIP Rx tcp:192.168.1.12:2081:

    ACK sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport

    From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

    To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

    Call-ID: 3c267cc8ba57-s7m7z0rilz9z

    CSeq: 2 ACK

    Max-Forwards: 70

    Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1

    Proxy-Require: buttons

    Content-Length: 0

     

    [5] 2011/12/28 10:28:34: SIP Rx udp:91.121.129.17:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39

    Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

    From: <sip:0033184190197@91.121.129.17>;tag=1119664557

    To: <sip:0972101112@91.121.129.17;user=phone>

    Call-ID: 0f4287ed@pbx

    CSeq: 12054 CANCEL

    server: Cirpack/v4.42j (gw_sip)

    Content-Length: 0

     

    [5] 2011/12/28 10:28:34: SIP Rx udp:91.121.129.17:5060:

    SIP/2.0 487 Session canceled

    Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39

    Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

    From: <sip:0033184190197@91.121.129.17>;tag=1119664557

    To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090

    Call-ID: 0f4287ed@pbx

    CSeq: 12054 INVITE

    Contact: <sip:41.141.84.105:5060>

    server: Cirpack/v4.42j (gw_sip)

    Allow: UPDATE, REFER, INFO

    Content-Length: 0

     

    [5] 2011/12/28 10:28:34: SIP Tx udp:192.168.1.1:5060:

    ACK sip:41.141.84.105:5060 SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport

    Route: <sip:siproxd@192.168.1.1:5060;lr>

    From: <sip:0033184190197@91.121.129.17>;tag=1119664557

    To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090

    Call-ID: 0f4287ed@pbx

    CSeq: 12054 ACK

    Max-Forwards: 70

    Contact: <sip:0033184190197@192.168.1.13:5060;transport=udp>

    Remote-Party-ID: <sip:0033184190197@91.121.129.17>;party=calling;screen=yes

    Content-Length: 0

     

    [5] 2011/12/28 10:28:34: INVITE Response 487 Session canceled: Terminate 0f4287ed@pbx

    [5] 2011/12/28 10:28:34: SIP Rx udp:91.121.129.17:5060:

    ACK sip:192.168.1.13:5060 SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKb094afeb726b070e8ab8f4e7222edbb0

    Via: SIP/2.0/UDP 41.141.84.105:5060;branch=z9hG4bKc0e36b0d24e61351467780d2975f9369

    Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport

    Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

    From: <sip:0033184190197@91.121.129.17>;tag=1119664557

    To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090

    Call-ID: 0f4287ed@pbx

    CSeq: 12054 ACK

    Contact: <sip:0033184190197@41.141.84.105>

    max-forwards: 68

    remote-party-id: <sip:0033184190197@91.121.129.17>;party=calling;screen=yes

    Content-Length: 0

     

    [5] 2011/12/28 10:28:35: SIP Rx udp:91.121.129.17:5060:

    SIP/2.0 487 Session canceled

    Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39

    Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

    From: <sip:0033184190197@91.121.129.17>;tag=1119664557

    To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090

    Call-ID: 0f4287ed@pbx

    CSeq: 12054 INVITE

    Contact: <sip:41.141.84.105:5060>

    server: Cirpack/v4.42j (gw_sip)

    Allow: UPDATE, REFER, INFO

    Content-Length: 0

    =====

     

     

     

    Any ideas ?

     

    I thing sound disappear just after the "183 Media Change" ....

     

    Thanks in advance,

     

    Jeremy

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