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yoyo

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  1. Hi, With both the old and the latest version, when the internet is restored after it went down, the SIP trunk does not automatically re-register, instead it stuck at proxy timeout. I have to login and manually click the register button. Would you please tell me what should be done to fix this? I set the "proposed duration" to 60 and keep alive to 30. also, almost every couple of days or so, the trunk would go from ok to proxy timeout or auth failure, then quickly back to ok. thank you!
  2. "There you find the domain admin PIN which you need to go from admin to user mode." The admin PIN is all digits - I assume it is not the admin web login password. I tried entering the PIN when I login to the phone's web interface, it fails authentication. Where do I enter this PIN? btw, I did add a custom xml config, and it worked as well. thank you
  3. Hi, the editor in the HTML template page does not load the entire template, it cuts off after certain number of characters. For example, the snom_820_phone.xml, it cuts off after some of the multicast address fields, even though there are much more after it (by looking at the generated config file). It makes it impossible to overwrite some of the settings. Pls take a look and let me know of any workaround or fix. thank you
  4. Hi, I just upgraded to the latest snomone version 2011-4.2.1.4025. Now I can not access the phone's admin menu -- even though I have logged in using the admin user/password - set in the snomone. What can I do to be able to access it? Thank you!
  5. I just used the windows 2003 active directory server, and added an ou=people for all the contacts, and they work great for incoming calls. There is no delay, and I put all the numbers for a single person in one contact object. using the builtin LDAP server, you are limited to 1 phone # per person. even for snom870, using the phone to look up directory is a real headache... so for now, it is for incoming display only
  6. Hi, the snom821 and snom870 directory key work fine - it is going through that action URL http://xx.xx.xx.xx/snom/adrbook.xml?auth=basic However, when looking up incoming call, it never works. this is using the builtin snomone LDAP server PCAP trace review that the response to the query string contains the whole directory lists (up to 50 entries), instead of the matched cn. (BTW, the PnP file says telephoneNumber as attribute, but I see telephoneNumber1 in the response from PBX). I then used a LDAP browser to see what's wrong. It complains about the "missing schema location in rootDSE, using default schema", and it shows the "cn=firstentry", expanding it shows all other user entries nested under it, and it goes on and on (you can keep clicking it and it will keep expanding showing the same CNs). what did I do wrong? thank you
  7. I understand they are 2 calls. But if the 2nd call maintains the same @address (the external callcentric ip), would the phone still think it missed 2 calls instead of one? The reason I put the extension number in the final destination field, instead of the cellphone# directly, is that I can not set the pbx to listen for a key press to connect the call (otherwise it would go into cellphone voicemail). Also, the ring duration can not be controlled any more. Maybe I am doing it wrong? Any suggestions you have on how to deal with this? thank you
  8. Hi, I have another issue needs help. I have setup hunt group with 2 stage calling. extension 20 is among extensions in the first stage, and it is also the final destination (for cell phone call out). When the call (from external) is not picked up, the phone (snom 821) would display 2 calls missed. one with phone#@xx.xx.xx.xx the external IP of callcentric -- for the initial hunt I believe. one with phone#@mypbx.company.com -- for the final destination so, it seems like the snomone has changed the final stage forking to use the domain, instead of the original header from SIP invite, resulting 2 missed call indications. what should I do (or change setup) to resolve this? thank you
  9. Not only that, if you hang up during the long delay (with noise only, no ringback), after the DNS is renewed, the pbx places the call to the called # (which does not show in the status window), and when picked up, there is no sound (as the caller has hung up), until a few seconds later. I have not been able to test the last free version 3958 I believe, but that version, the cell phone virtual personal assistant does not work, and other issues. why the TTL is so short? and why there is a 25 seconds delay between the NAPTR expiration and add again? please help. thank you!
  10. Hi, I just upgrade to snomone free 4.2 (2/1/2011) load for windows 32 bit. I noticed that many times, the pbx will not dial out on the trunk until about 30 seconds later. I think I nailed it down to the DNS registration expiration - if the dialing happens right after the expiration and before the renewal, the pbx will ack the initial invite from the extension, but will not send invite to the external number through SIP trunk until about 30 seconds later - after the DNS renewal happens. This is really not working with all sorts of issues with timing. Would you please look into ASAP? Anything bad on my end? (I will try to find the old version and roll back and see) P.S.: In addition, I had one case where the snom821 just lost communication with snomone, and would not talk to it after reset powerdown or whatever. I had to reset 821 to factory default and reprovision to work (not sure who fault it was?). Also, the button profiles would stuck at 4 buttons, no matter how I edit the profile (both domain and extension level), only the initial 4 buttons went to the phone. This one is minor. [8] 2011/02/13 09:05:49: DNS: SRV _sip._tcp.callcentric.com expired [8] 2011/02/13 09:05:49: DNS: A alpha5.callcentric.com expired [8] 2011/02/13 09:05:49: DNS: AAAA alpha5.callcentric.com expired [8] 2011/02/13 09:05:49: DNS: AAAA alpha6.callcentric.com expired [8] 2011/02/13 09:05:49: DNS: AAAA alpha7.callcentric.com expired [8] 2011/02/13 09:05:49: DNS: AAAA alpha8.callcentric.com expired [8] 2011/02/13 09:05:49: DNS: AAAA alpha9.callcentric.com expired [8] 2011/02/13 09:05:49: DNS: AAAA alpha1.callcentric.com expired [8] 2011/02/13 09:05:50: DNS: AAAA alpha2.callcentric.com expired [8] 2011/02/13 09:05:50: DNS: AAAA alpha3.callcentric.com expired [8] 2011/02/13 09:05:50: DNS: AAAA alpha4.callcentric.com expired [8] 2011/02/13 09:05:50: DNS: AAAA callcentric.com expired [8] 2011/02/13 09:05:50: DNS: NAPTR callcentric.com expired [8] 2011/02/13 09:05:50: DNS: Add NAPTR callcentric.com (ttl=60) [5] 2011/02/13 09:05:55: SIP Rx udp:192.168.0.193:5060: INVITE sip:xxxxxxxxxx@192.168.0.68 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.193;branch=z9hG4bKg5nzjpt0o8lfcssjv18wzjzgr Max-Forwards: 70 From: "28" <sip:28@192.168.0.68>;tag=j6..k6va To: <sip:xxxxxxxxxx@192.168.0.68> Call-ID: v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 CSeq: 38083 INVITE Contact: <sip:28@192.168.0.193;line=58316> Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Content-Disposition: session Supported: replaces User-Agent: snom-m3-SIP/02.11 (MAC=0004132A2FC5; HW=1) Content-Type: application/sdp Content-Length: 320 v=0 o=28 48052301 48052301 IN IP4 192.168.0.193 s=- c=IN IP4 192.168.0.193 t=0 0 a=sendrecv m=audio 5008 RTP/AVP 0 8 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:5009 [8] 2011/02/13 09:05:55: Could not find a trunk (3 trunks) [5] 2011/02/13 09:05:55: SIP Rx udp:192.168.0.193:5060: INVITE sip:xxxxxxxxxx@192.168.0.68 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.193;branch=z9hG4bKg5nzjpt0o8lfcssjv18wzjzgr Max-Forwards: 70 From: "28" <sip:28@192.168.0.68>;tag=j6..k6va To: <sip:xxxxxxxxxx@192.168.0.68> Call-ID: v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 CSeq: 38083 INVITE Contact: <sip:28@192.168.0.193;line=58316> Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Content-Disposition: session Supported: replaces User-Agent: snom-m3-SIP/02.11 (MAC=0004132A2FC5; HW=1) Content-Type: application/sdp Content-Length: 320 v=0 o=28 48052301 48052301 IN IP4 192.168.0.193 s=- c=IN IP4 192.168.0.193 t=0 0 a=sendrecv m=audio 5008 RTP/AVP 0 8 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:5009 [5] 2011/02/13 09:05:55: SIP Tx udp:192.168.0.193:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.193;branch=z9hG4bKg5nzjpt0o8lfcssjv18wzjzgr From: "28" <sip:28@192.168.0.68>;tag=j6..k6va To: <sip:xxxxxxxxxx@192.168.0.68>;tag=619765c407 Call-ID: v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 CSeq: 38083 INVITE Content-Length: 0 [5] 2011/02/13 09:05:55: SIP Tx udp:192.168.0.193:5060: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.0.193;branch=z9hG4bKg5nzjpt0o8lfcssjv18wzjzgr From: "28" <sip:28@192.168.0.68>;tag=j6..k6va To: <sip:xxxxxxxxxx@192.168.0.68>;tag=619765c407 Call-ID: v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 CSeq: 38083 INVITE User-Agent: snom-PBX/2011-4.2.0.3981 WWW-Authenticate: Digest realm="192.168.0.68",nonce="cca944c85dd9483b3c581915679c927b",domain="sip:xxxxxxxxxx@192.168.0.68",algorithm=MD5 Content-Length: 0 [5] 2011/02/13 09:05:55: SIP Rx udp:192.168.0.193:5060: ACK sip:xxxxxxxxxx@192.168.0.68 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.193;branch=z9hG4bKg5nzjpt0o8lfcssjv18wzjzgr Max-Forwards: 70 From: "28" <sip:28@192.168.0.68>;tag=j6..k6va To: <sip:xxxxxxxxxx@192.168.0.68>;tag=619765c407 Call-ID: v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 CSeq: 38083 ACK User-Agent: snom-m3-SIP/02.11 (MAC=0004132A2FC5; HW=1) Content-Length: 0 [5] 2011/02/13 09:05:55: SIP Rx udp:192.168.0.193:5060: INVITE sip:xxxxxxxxxx@192.168.0.68 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.193;branch=z9hG4bKtfvcbiv0d Max-Forwards: 70 From: "28" <sip:28@192.168.0.68>;tag=j6..k6va To: <sip:xxxxxxxxxx@192.168.0.68> Call-ID: v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 CSeq: 38084 INVITE Contact: <sip:28@192.168.0.193;line=58316> Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Authorization: Digest username="28", realm="192.168.0.68", nonce="cca944c85dd9483b3c581915679c927b", uri="sip:xxxxxxxxxx@192.168.0.68", response="027b37456b4e36002ae4331bbf6960fb", algorithm=MD5 Content-Disposition: session Supported: replaces User-Agent: snom-m3-SIP/02.11 (MAC=0004132A2FC5; HW=1) Content-Type: application/sdp Content-Length: 320 v=0 o=28 48052301 48052301 IN IP4 192.168.0.193 s=- c=IN IP4 192.168.0.193 t=0 0 a=sendrecv m=audio 5008 RTP/AVP 0 8 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:5009 [8] 2011/02/13 09:05:55: Tagging request with existing tag [6] 2011/02/13 09:05:55: Sending RTP for v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 to 192.168.0.193:5008, codec not set yet [5] 2011/02/13 09:05:55: SIP Tx udp:192.168.0.193:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.193;branch=z9hG4bKtfvcbiv0d From: "28" <sip:28@192.168.0.68>;tag=j6..k6va To: <sip:xxxxxxxxxx@192.168.0.68>;tag=619765c407 Call-ID: v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 CSeq: 38084 INVITE Content-Length: 0 [8] 2011/02/13 09:05:55: Call from an user 28 [8] 2011/02/13 09:05:55: To is <sip:xxxxxxxxxx@192.168.0.68>, user 0, domain 1 [8] 2011/02/13 09:05:55: From user 28 [8] 2011/02/13 09:05:55: Set the To domain based on From user 28@pbx.xxxxx-xxxxxx.com [8] 2011/02/13 09:05:55: Call state for call object 20: idle [5] 2011/02/13 09:05:55: Dialplan "callcentric": Match xxxxxxxxxx@192.168.0.68 to <sip:1xxxxxxxxxx@callcentric.com;user=phone> on trunk CallCentric6302-TF [8] 2011/02/13 09:05:55: Play audio_moh/noise.wav [7] 2011/02/13 09:05:55: set_codecs: for v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 codecs "", codec_preference count 6 [5] 2011/02/13 09:05:55: SIP Tx tls:192.168.0.195:3455: MESSAGE sip:20@192.168.0.195:3455;transport=tls;line=0ii3wcy6 SIP/2.0 Via: SIP/2.0/TLS 192.168.0.68:5061;branch=z9hG4bK-3679d3421a391cfb5f0b06afbc88cee9;rport From: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com>;tag=58990 To: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com> Call-ID: 8x6ae3js@pbx CSeq: 11997 MESSAGE Max-Forwards: 70 Contact: <sip:192.168.0.68:5061;transport=tls> Subject: buttons Content-Type: application/x-buttons Content-Length: 21 k=3 x=ext l=Op8 [7] 2011/02/13 09:05:55: set_codecs: for 2ba7bbea@pbx codecs "", codec_preference count 6 [5] 2011/02/13 09:05:55: SIP Tx tls:192.168.0.195:3455: MESSAGE sip:20@192.168.0.195:3455;transport=tls;line=0ii3wcy6 SIP/2.0 Via: SIP/2.0/TLS 192.168.0.68:5061;branch=z9hG4bK-e6437df73375d69462fba1e0f935a9ac;rport From: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com>;tag=32908 To: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com> Call-ID: sq29vatg@pbx CSeq: 43406 MESSAGE Max-Forwards: 70 Contact: <sip:192.168.0.68:5061;transport=tls> Subject: buttons Content-Type: application/x-buttons Content-Length: 27 k=3 c=on x=ext l=Op8 [6] 2011/02/13 09:05:55: Codec pcmu/8000 is chosen for call id v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 [5] 2011/02/13 09:05:55: SIP Tx udp:192.168.0.193:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.193;branch=z9hG4bKtfvcbiv0d From: "28" <sip:28@192.168.0.68>;tag=j6..k6va To: <sip:xxxxxxxxxx@192.168.0.68>;tag=619765c407 Call-ID: v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 CSeq: 38084 INVITE Contact: <sip:28@192.168.0.68:5060> 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-Type: application/sdp Content-Length: 249 v=0 o=- 913 913 IN IP4 192.168.0.68 s=- c=IN IP4 192.168.0.68 t=0 0 m=audio 62202 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2011/02/13 09:05:55: SIP Rx tls:192.168.0.195:3455: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.0.68:5061;branch=z9hG4bK-3679d3421a391cfb5f0b06afbc88cee9;rport=5061 From: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com>;tag=58990 To: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com> Call-ID: 8x6ae3js@pbx CSeq: 11997 MESSAGE Content-Length: 0 [5] 2011/02/13 09:05:55: SIP Rx tls:192.168.0.195:3455: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.0.68:5061;branch=z9hG4bK-e6437df73375d69462fba1e0f935a9ac;rport=5061 From: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com>;tag=32908 To: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com> Call-ID: sq29vatg@pbx CSeq: 43406 MESSAGE Content-Length: 0 [8] 2011/02/13 09:06:04: DNS: Add SRV _sips._tcp.callcentric.com (ttl=60) [8] 2011/02/13 09:06:13: Packet authenticated by transport layer [8] 2011/02/13 09:06:14: DNS: A alpha2.callcentric.com expired [8] 2011/02/13 09:06:18: DNS: Add SRV _sip._tcp.callcentric.com (ttl=60) [8] 2011/02/13 09:06:18: DNS: Add A alpha5.callcentric.com 204.11.192.35 (ttl=579) [5] 2011/02/13 09:06:18: SIP Tx udp:204.11.192.35:5080: INVITE sip:1xxxxxxxxxx@callcentric.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-8a0306ecf28e071e58e668223a3a394f;rport From: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 To: <sip:1xxxxxxxxxx@callcentric.com;user=phone> Call-ID: 2ba7bbea@pbx CSeq: 31672 INVITE Max-Forwards: 70 Contact: <sip:1777xxxxxxx@192.168.0.68:5060;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: "28" <sip:1xxxxxxxxxx@192.168.0.68;user=phone>;party=calling;screen=yes Content-Type: application/sdp Content-Length: 325 v=0 o=- 1953 1953 IN IP4 192.168.0.68 s=- c=IN IP4 192.168.0.68 t=0 0 m=audio 50430 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2011/02/13 09:06:18: SIP Rx udp:204.11.192.35:5080: SIP/2.0 407 Proxy Authentication Required v: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-8a0306ecf28e071e58e668223a3a394f;rport=5060;received=99.112.222.198 f: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 t: <sip:1xxxxxxxxxx@callcentric.com;user=phone> i: 2ba7bbea@pbx CSeq: 31672 INVITE Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="2f1430c2b771ea5b4f948e06ca741b63", opaque="", stale=TRUE, algorithm=MD5 l: 0 [8] 2011/02/13 09:06:18: Answer challenge with username 1777xxxxxxx [5] 2011/02/13 09:06:18: SIP Tx udp:204.11.192.35:5080: ACK sip:1xxxxxxxxxx@callcentric.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-8a0306ecf28e071e58e668223a3a394f;rport From: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 To: <sip:1xxxxxxxxxx@callcentric.com;user=phone> Call-ID: 2ba7bbea@pbx CSeq: 31672 ACK Max-Forwards: 70 Content-Length: 0 [5] 2011/02/13 09:06:18: SIP Tx udp:204.11.192.35:5080: INVITE sip:1xxxxxxxxxx@callcentric.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-31f35de730cc0bc590e224811c6243eb;rport From: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 To: <sip:1xxxxxxxxxx@callcentric.com;user=phone> Call-ID: 2ba7bbea@pbx CSeq: 31673 INVITE Max-Forwards: 70 Contact: <sip:1777xxxxxxx@192.168.0.68:5060;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: "28" <sip:1xxxxxxxxxx@192.168.0.68;user=phone>;party=calling;screen=yes Proxy-Authorization: Digest realm="callcentric.com",nonce="2f1430c2b771ea5b4f948e06ca741b63",response="0cc438a3061c4856c950f5780d8f56d6",username="1777xxxxxxx",uri="sip:1xxxxxxxxxx@callcentric.com;user=phone",algorithm=MD5 Content-Type: application/sdp Content-Length: 325 v=0 o=- 1953 1953 IN IP4 192.168.0.68 s=- c=IN IP4 192.168.0.68 t=0 0 m=audio 50430 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2011/02/13 09:06:18: SIP Rx udp:204.11.192.35:5080: SIP/2.0 100 Trying v: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-31f35de730cc0bc590e224811c6243eb;rport=5060;received=99.112.222.198 f: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 t: <sip:1xxxxxxxxxx@callcentric.com;user=phone> i: 2ba7bbea@pbx CSeq: 31673 INVITE l: 0 [8] 2011/02/13 09:06:18: DNS: Add AAAA alpha5.callcentric.com (ttl=60) [8] 2011/02/13 09:06:18: DNS: Add AAAA alpha6.callcentric.com (ttl=60) [5] 2011/02/13 09:06:18: SIP Rx udp:204.11.192.35:5080: SIP/2.0 100 Trying v: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-31f35de730cc0bc590e224811c6243eb;rport=5060;received=99.112.222.198 f: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 t: <sip:1xxxxxxxxxx@callcentric.com;user=phone> i: 2ba7bbea@pbx CSeq: 31673 INVITE l: 0 [8] 2011/02/13 09:06:18: DNS: Add AAAA alpha7.callcentric.com (ttl=60) [8] 2011/02/13 09:06:18: DNS: Add AAAA alpha8.callcentric.com (ttl=60) [8] 2011/02/13 09:06:18: DNS: Add AAAA alpha9.callcentric.com (ttl=60) [8] 2011/02/13 09:06:18: DNS: Add AAAA alpha1.callcentric.com (ttl=60) [8] 2011/02/13 09:06:18: DNS: Add A alpha2.callcentric.com 204.11.192.23 (ttl=541) [8] 2011/02/13 09:06:19: DNS: Add AAAA alpha2.callcentric.com (ttl=60) [8] 2011/02/13 09:06:19: DNS: Add AAAA alpha3.callcentric.com (ttl=60) [8] 2011/02/13 09:06:19: DNS: Add AAAA alpha4.callcentric.com (ttl=60) [8] 2011/02/13 09:06:19: DNS: Add AAAA callcentric.com (ttl=60) [8] 2011/02/13 09:06:19: Trunk 1 (CallCentric6302-TF) is associated with the following addresses: udp:204.11.192.22:5080 udp:204.11.192.23:5080 udp:204.11.192.31:5080 udp:204.11.192.34:5080 udp:204.11.192.35:5080 udp:204.11.192.36:5080 udp:204.11.192.37:5080 udp:204.11.192.38:5080 udp:204.11.192.39:5060 udp:204.11.192.39:5080 [5] 2011/02/13 09:06:20: SIP Rx udp:204.11.192.35:5080: SIP/2.0 183 Session Progress v: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-31f35de730cc0bc590e224811c6243eb;rport=5060;received=99.112.222.198 f: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 t: <sip:1xxxxxxxxxx@callcentric.com;user=phone>;tag=3506598369-868616 i: 2ba7bbea@pbx CSeq: 31673 INVITE m: <sip:b0bfe101b6820dd2cbfd61fe0d310d20@204.11.192.35:5080;transport=udp> c: application/sdp l: 267 v=0 o=NexTone-MSW 27505 11232 IN IP4 204.11.192.35 s=sip call c=IN IP4 204.11.192.35 t=0 0 m=audio 59830 RTP/AVP 0 101 a=ptime:20 a=sendrecv a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=setup:actpass [7] 2011/02/13 09:06:20: Set packet length to 20 [6] 2011/02/13 09:06:20: Codec pcmu/8000 is chosen for call id 2ba7bbea@pbx [6] 2011/02/13 09:06:20: Sending RTP for 2ba7bbea@pbx to 204.11.192.35:59830, codec pcmu/8000 [8] 2011/02/13 09:06:20: Call state for call object 20: alerting [5] 2011/02/13 09:06:20: SIP Tx tls:192.168.0.195:3455: MESSAGE sip:20@192.168.0.195:3455;transport=tls;line=0ii3wcy6 SIP/2.0 Via: SIP/2.0/TLS 192.168.0.68:5061;branch=z9hG4bK-94184c3cd17576ce1149a8d3bbb256eb;rport From: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com>;tag=21985 To: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com> Call-ID: xsq4dk8y@pbx CSeq: 1867 MESSAGE Max-Forwards: 70 Contact: <sip:192.168.0.68:5061;transport=tls> Subject: buttons Content-Type: application/x-buttons Content-Length: 27 k=3 c=on x=ext l=Op8 [7] 2011/02/13 09:06:20: v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68: RTP pass-through mode [7] 2011/02/13 09:06:20: 2ba7bbea@pbx: RTP pass-through mode [5] 2011/02/13 09:06:20: SIP Rx tls:192.168.0.195:3455: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.0.68:5061;branch=z9hG4bK-94184c3cd17576ce1149a8d3bbb256eb;rport=5061 From: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com>;tag=21985 To: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com> Call-ID: xsq4dk8y@pbx CSeq: 1867 MESSAGE Content-Length: 0 [5] 2011/02/13 09:06:20: SIP Rx udp:204.11.192.35:5080: SIP/2.0 183 Session Progress v: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-31f35de730cc0bc590e224811c6243eb;rport=5060;received=99.112.222.198 f: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 t: <sip:1xxxxxxxxxx@callcentric.com;user=phone>;tag=3506598369-868616 i: 2ba7bbea@pbx CSeq: 31673 INVITE m: <sip:b0bfe101b6820dd2cbfd61fe0d310d20@204.11.192.35:5080;transport=udp> c: application/sdp l: 267 v=0 o=NexTone-MSW 27505 11232 IN IP4 204.11.192.35 s=sip call c=IN IP4 204.11.192.35 t=0 0 m=audio 59830 RTP/AVP 0 101 a=ptime:20 a=sendrecv a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=setup:actpass [7] 2011/02/13 09:06:20: Set packet length to 20 [5] 2011/02/13 09:06:21: SIP Rx udp:204.11.192.35:5080: SIP/2.0 183 Session Progress v: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-31f35de730cc0bc590e224811c6243eb;rport=5060;received=99.112.222.198 f: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 t: <sip:1xxxxxxxxxx@callcentric.com;user=phone>;tag=3506598369-868616 i: 2ba7bbea@pbx CSeq: 31673 INVITE m: <sip:b0bfe101b6820dd2cbfd61fe0d310d20@204.11.192.35:5080;transport=udp> c: application/sdp l: 267 v=0 o=NexTone-MSW 27505 11232 IN IP4 204.11.192.35 s=sip call c=IN IP4 204.11.192.35 t=0 0 m=audio 59830 RTP/AVP 0 101 a=ptime:20 a=sendrecv a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=setup:actpass [7] 2011/02/13 09:06:21: Set packet length to 20 [5] 2011/02/13 09:06:23: SIP Rx udp:204.11.192.35:5080: SIP/2.0 183 Session Progress v: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-31f35de730cc0bc590e224811c6243eb;rport=5060;received=99.112.222.198 f: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 t: <sip:1xxxxxxxxxx@callcentric.com;user=phone>;tag=3506598369-868616 i: 2ba7bbea@pbx CSeq: 31673 INVITE m: <sip:b0bfe101b6820dd2cbfd61fe0d310d20@204.11.192.35:5080;transport=udp> c: application/sdp l: 267 v=0 o=NexTone-MSW 27505 11232 IN IP4 204.11.192.35 s=sip call c=IN IP4 204.11.192.35 t=0 0 m=audio 59830 RTP/AVP 0 101 a=ptime:20 a=sendrecv a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=setup:actpass [7] 2011/02/13 09:06:23: Set packet length to 20 [5] 2011/02/13 09:06:24: SIP Rx tls:192.168.0.195:3455: SUBSCRIBE sip:192.168.0.68:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.0.195:3455;branch=z9hG4bK-jeymxplsl7h4;rport From: <sip:20@pbx.xxxxx-xxxxxx.com>;tag=geina9ukdy To: <sip:20@pbx.xxxxx-xxxxxx.com;user=phone>;tag=18519a5474 Call-ID: e720293c26e0-1y6gba6gyxij CSeq: 1448 SUBSCRIBE Max-Forwards: 70 Contact: <sip:20@192.168.0.195:3455;transport=tls;line=0ii3wcy6>;reg-id=1 Event: message-summary Accept: application/simple-message-summary User-Agent: snom821/8.4.18 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2011/02/13 09:06:24: Packet authenticated by transport layer [5] 2011/02/13 09:06:24: SIP Tx tls:192.168.0.195:3455: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.0.195:3455;branch=z9hG4bK-jeymxplsl7h4;rport=3455 From: <sip:20@pbx.xxxxx-xxxxxx.com>;tag=geina9ukdy To: <sip:20@pbx.xxxxx-xxxxxx.com;user=phone>;tag=18519a5474 Call-ID: e720293c26e0-1y6gba6gyxij CSeq: 1448 SUBSCRIBE Contact: <sip:192.168.0.68:5061;transport=tls> Expires: 179 Content-Length: 0 [5] 2011/02/13 09:06:24: SIP Rx udp:192.168.0.193:5060: CANCEL sip:xxxxxxxxxx@192.168.0.68 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.193;branch=z9hG4bKtfvcbiv0d Max-Forwards: 70 From: "28" <sip:28@192.168.0.68>;tag=j6..k6va To: <sip:xxxxxxxxxx@192.168.0.68> Call-ID: v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 CSeq: 38084 CANCEL Contact: <sip:28@192.168.0.193;line=58316> Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Authorization: Digest username="28", realm="192.168.0.68", nonce="cca944c85dd9483b3c581915679c927b", uri="sip:28@192.168.0.68", response="36e1bfcebf410abc9026a26d84291362", algorithm=MD5 User-Agent: snom-m3-SIP/02.11 (MAC=0004132A2FC5; HW=1) Content-Length: 0 [5] 2011/02/13 09:06:24: SIP Tx udp:192.168.0.193:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.0.193;branch=z9hG4bKtfvcbiv0d From: "28" <sip:28@192.168.0.68>;tag=j6..k6va To: <sip:xxxxxxxxxx@192.168.0.68>;tag=619765c407 Call-ID: v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 CSeq: 38084 CANCEL Contact: <sip:28@192.168.0.68:5060> User-Agent: snom-PBX/2011-4.2.0.3981 Content-Length: 0 [5] 2011/02/13 09:06:24: SIP Tx udp:192.168.0.193:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.0.193;branch=z9hG4bKtfvcbiv0d From: "28" <sip:28@192.168.0.68>;tag=j6..k6va To: <sip:xxxxxxxxxx@192.168.0.68>;tag=619765c407 Call-ID: v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 CSeq: 38084 INVITE Contact: <sip:28@192.168.0.68:5060> 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] 2011/02/13 09:06:24: SIP Rx udp:192.168.0.193:5060: ACK sip:xxxxxxxxxx@192.168.0.68 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.193;branch=z9hG4bKtfvcbiv0d Max-Forwards: 70 From: "28" <sip:28@192.168.0.68>;tag=j6..k6va To: <sip:xxxxxxxxxx@192.168.0.68>;tag=619765c407 Call-ID: v7aadg0nzynnjej1zeeh9kpsmlx0@192.168.0.68 CSeq: 38084 ACK User-Agent: snom-m3-SIP/02.11 (MAC=0004132A2FC5; HW=1) Content-Length: 0 [7] 2011/02/13 09:06:24: 2ba7bbea@pbx: Media-aware pass-through mode [8] 2011/02/13 09:06:24: Hangup: Call 70 not found [5] 2011/02/13 09:06:24: SIP Tx udp:204.11.192.35:5080: CANCEL sip:1xxxxxxxxxx@callcentric.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-31f35de730cc0bc590e224811c6243eb;rport From: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 To: <sip:1xxxxxxxxxx@callcentric.com;user=phone> Call-ID: 2ba7bbea@pbx CSeq: 31673 CANCEL Max-Forwards: 70 Remote-Party-ID: "28" <sip:1xxxxxxxxxx@192.168.0.68;user=phone>;party=calling;screen=yes Content-Length: 0 [5] 2011/02/13 09:06:24: SIP Tx tls:192.168.0.195:3455: MESSAGE sip:20@192.168.0.195:3455;transport=tls;line=0ii3wcy6 SIP/2.0 Via: SIP/2.0/TLS 192.168.0.68:5061;branch=z9hG4bK-6c473e7a406bdbf245aae65bb91c99e4;rport From: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com>;tag=702 To: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com> Call-ID: o472vzqk@pbx CSeq: 2589 MESSAGE Max-Forwards: 70 Contact: <sip:192.168.0.68:5061;transport=tls> Subject: buttons Content-Type: application/x-buttons Content-Length: 21 k=3 x=ext l=Op8 [8] 2011/02/13 09:06:24: Hangup: Call 70 not found [5] 2011/02/13 09:06:24: SIP Rx tls:192.168.0.195:3455: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.0.68:5061;branch=z9hG4bK-6c473e7a406bdbf245aae65bb91c99e4;rport=5061 From: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com>;tag=702 To: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com> Call-ID: o472vzqk@pbx CSeq: 2589 MESSAGE Content-Length: 0 [5] 2011/02/13 09:06:24: SIP Rx udp:204.11.192.35:5080: SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-31f35de730cc0bc590e224811c6243eb;rport=5060;received=99.112.222.198 f: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 t: <sip:1xxxxxxxxxx@callcentric.com;user=phone> i: 2ba7bbea@pbx CSeq: 31673 CANCEL l: 0 [7] 2011/02/13 09:06:24: Call 2ba7bbea@pbx: Clear last request [5] 2011/02/13 09:06:24: SIP Rx udp:204.11.192.35:5080: SIP/2.0 487 Request Terminated v: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-31f35de730cc0bc590e224811c6243eb;rport=5060;received=99.112.222.198 f: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 t: <sip:1xxxxxxxxxx@callcentric.com;user=phone>;tag=3506598369-868616 i: 2ba7bbea@pbx CSeq: 31673 INVITE m: <sip:b0bfe101b6820dd2cbfd61fe0d310d20@204.11.192.35:5080;transport=udp> l: 0 [7] 2011/02/13 09:06:24: Call 2ba7bbea@pbx: Clear last INVITE [5] 2011/02/13 09:06:24: SIP Tx udp:204.11.192.35:5080: ACK sip:b0bfe101b6820dd2cbfd61fe0d310d20@204.11.192.35:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.68:5060;branch=z9hG4bK-31f35de730cc0bc590e224811c6243eb;rport From: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;tag=34600 To: <sip:1xxxxxxxxxx@callcentric.com;user=phone>;tag=3506598369-868616 Call-ID: 2ba7bbea@pbx CSeq: 31673 ACK Max-Forwards: 70 Contact: <sip:1777xxxxxxx@192.168.0.68:5060;transport=udp> Remote-Party-ID: "NOT WORKING" <sip:1777xxxxxxx@callcentric.com>;party=calling;screen=yes Content-Length: 0 [5] 2011/02/13 09:06:24: INVITE Response 487 Request Terminated: Terminate 2ba7bbea@pbx [5] 2011/02/13 09:06:24: SIP Tx tls:192.168.0.195:3455: MESSAGE sip:20@192.168.0.195:3455;transport=tls;line=0ii3wcy6 SIP/2.0 Via: SIP/2.0/TLS 192.168.0.68:5061;branch=z9hG4bK-d56a11de4294f68ce78a63070c4e7899;rport From: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com>;tag=34693 To: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com> Call-ID: irdhbbb2@pbx CSeq: 32636 MESSAGE Max-Forwards: 70 Contact: <sip:192.168.0.68:5061;transport=tls> Subject: buttons Content-Type: application/x-buttons Content-Length: 21 k=3 x=ext l=Op8 [5] 2011/02/13 09:06:24: SIP Rx tls:192.168.0.195:3455: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.0.68:5061;branch=z9hG4bK-d56a11de4294f68ce78a63070c4e7899;rport=5061 From: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com>;tag=34693 To: "FrontDesk FDC" <sip:20@pbx.xxxxx-xxxxxx.com> Call-ID: irdhbbb2@pbx CSeq: 32636 MESSAGE Content-Length: 0
  11. Just want to share my understanding of the flags after desperately trying to understand how they work (as the manual does not explain fully). Hopefully this will help newbies like me Pros, if any of the following is incorrect, please educate me! thank you IVR Greeting/NightService order: If no service flag is set, use the default (must use the *98xx first then overwrite with a .wav file, setting in the default welcome message has no effect) If there is only one service flag is set, that one is announced If more than one service flags are set, the first one listed in the IVR greeting page is announced If Night Service fields are filled in, but no flag (in this field) is set, greeting is played by above rules. If Night Service fields are filled in, and any one of the flag is set, there is NO greeting, forward to the corresponding Night Service Number directly If Night Service fields are filled in, and more than one of the flags are set, there is NO greeting, forward to the Night Service Number corresponding to the first service flag in the SET state directly The service flag listing in the IVR greeting page and the Night Service fields (intended for immediate escape) are TOTALLY INDEPENDENT!
  12. Thank you for the quick explanation. The manual was saying totally different things I am not picking, it just seems that the manual is missing some important stuff, and can not quite keep up with the development (kind of a good thing in a sense:)) btw, for the call log, when logged in as the administrator, the call log only shows calls going through the trunk, while when logged in as the user, it shows all the calls (including hunt group and IVR). Is it intentional? also, sometimes you post the binary that are several versions ahead than what's in the snomone free. Can we use those binary(beta version I guess)? Or we should stick with the official version for snomone free? thank you again!
  13. Hi, I am looking at the files under those CDR directories, but they are not in the correct xml format? they are all separated with weird characters. Please let me know what went wrong? thank you TLVB c 1295477461.93 cid 0rd0wcqjpwwocfu5@192.168.0.8 ct d d 1 e 1295477470.296 f ""40" <sip:40@xxxx.com> i 0rd0wcqjpwwocfu5@192.168.0.8 o I p udp:192.168.0.183:5060 r <sip:72@192.168.0.8> s 1295477450.437 t <sip:72@192.168.0.8> u 1 vq TVQSessionReport: CallTerm LocalMetrics: Timestamps:START=2011-01-19T22:51:01Z STOP=2011-01-19T22:51:10Z CallID:0rd0wcqjpwwocfu5@192.168.0.8 FromID:"40" <sip:40@192.168.0.8>;tag=h39uzz ToID:<sip:72@192.168.0.8>;tag=43127b5b7e SessionDesc:PT=0 PD=pcmu SR=8000 FD=20 FO=160 FPP=1 PPS=50 PLC=3 LocalAddr:IP=192.168.0.8 PORT=57746 SSRC=0x94d1769a RemoteAddr:IP=192.168.0.183 PORT=5022 SSRC=0xe197ebba x-UserAgent:snom-PBX/4.2.0.3950 x-SIPterm:SDC=OK SDR=AN PacketLoss:NLR=0.0 JDR=0.0 BurstGapLoss:BLD=0.0 BD=0 GLD=0.0 GD=0 GMIN=16 Delay:RTD=0 ESD=0 IAJ=0 QualityEst:MOSLQ=4.1 MOSCQ=4.1
  14. I just changed the LDAP port away from 389, and now the service stays up and I can finally log in. Question though: is the SnomOne running a LDAP server? If this is for client only, shouldn't we set the correct port 389? I searched the manual and did not see any info on this. Please help. thank you!
  15. Hi, I just downloaded the SnomOne free version for windows 32 bit. Installed it on windows 2003 service pack 2 standard version. No other services using ports 80 and 443. The service will start then stop in 5 seconds, the event log showing the application has failed. " Faulting application pbxctrl.exe, version 0.0.0.0, faulting module pbxctrl.exe, version 0.0.0.0, fault address 0x00186541. For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp. " btw, I installed the dot net 4 as well. Same error. The server has DNS server running and is the primary domain controller. Any idea on what could be the problem? (I can attached the windows error files if needed) Thank you very much!
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