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Jeremy Salmon

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  1. But with "virtual machine" there is only ONE machine. Where is the failover if the machine completely crash ?
  2. 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) ├ędition. 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. 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. Thanks for your EXTREMELY quick answer !!! Is there a method to automatically conserve space on the disk ?
  5. 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. 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. 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
  8. Great ! Thanks ... Not the same philosophy with Asterisk ... That's why I have not thought of that Many thanks, Jeremy
  9. 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 ...
  10. Hi All, I have a trunk with multiple DID (ex : 33144736315, 33356764576, ...) I want to select the outgoing DID from Dialplan ... For example when I call a 335xxxxxxxx I want to use the local 33356764576. How can I play with replacement to make this ? Thanks ! JS
  11. 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. 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. 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. 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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