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Everything posted by gifti
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Hello again, we use a snom ONE mini (yellow - twenty) and we get about 100 calls inbound and 50 calls outbound a day. Approx. 5-10 calls a day sporadically can not be established, both incoming and outgoing. Incoming: the calls coming in over a ISDN gateway (patton) huntgoup members or a single extension is ringing the error occurs (I receive a syslog email) -> codec_preference size 4, available codecs list is empty the caller gets an announcement "Dienst oder Dienstmerkmal nicht verfügbar" -> http://de.wikipedia.org/wiki/Telefonansage Ansage 8 the huntgoup members are continue to ring ... a ghost ring you can't answer or disconnect the ghost call ... after approx. 30 sec. the ghost disappears Syslog message incoming issue: <131>1 2013-05-23T07:20:23+02:00 snomonemini Call - - - Call port 114: update_codecs for 76f6b3c1febc6cae: codec_preference size 4, available codecs list is empty <133>1 2013-05-23T07:20:23+02:00 snomonemini SIP - - - SIP Tx udp:192.168.0.220:5060: SIP/2.0 415 Unsupported Media Type#015#012Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK96ea7d97115bfd740#015#012From: <sip:xxx@192.168.0.220:5060>;tag=c92113b1a8#015#012To: <sip:xxx@192.168.0.200>;tag=64a885bb70#015#012Call-ID: 76f6b3c1febc6cae#015#012CSeq: 2063 INVITE#015#012Contact: <sip:730730@192.168.0.200:5060;transport=udp>#015#012Supported: 100rel, replaces, norefersub#015#012Allow-Events: refer#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE#015#012Accept: application/sdp#015#012User-Agent: snomONE/5.0.10#015#012Content-Length: 0#015#012#015 Outgoing: whether mailbox-, inside- or outgoing call over a trunk we get the Display Message (SNOM370) "Unsupported Media Typ" next try - everything is working again ... mostly Syslog message outgoing issue: <131>1 2013-05-27T10:49:41+02:00 snomonemini Call - - - Call port 66: update_codecs for 51a33a41aea8-og0aij2eyh65: codec_preference size 4, available codecs list is empty <133>1 2013-05-27T10:49:41+02:00 snomonemini SIP - - - SIP Tx tcp:192.168.0.205:2482: SIP/2.0 415 Unsupported Media Type#015#012Via: SIP/2.0/TCP 192.168.0.205:2482;branch=z9hG4bK-ubcei96kykxh;rport=2482#015#012From: "xxx" <sip:10@pbx.ggizef.lokal>;tag=ktdzuxmjrj#015#012To: <sip:xxx@pbx.ggizef.lokal;user=phone>;tag=997bb35ced#015#012Call-ID: 51a33a41aea8-og0aij2eyh65#015#012CSeq: 2 INVITE#015#012Contact: <sip:10@192.168.0.200:5060;transport=tcp>#015#012Supported: 100rel, replaces, norefersub#015#012Allow-Events: refer#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE#015#012Accept: application/sdp#015#012User-Agent: snomONE/5.0.10#015#012Content-Length: 0#015#012#015 Another outgoing issue (Logfile): [5] 2013/05/27 15:21:42: SIP Rx udp:192.168.0.220:5060: INVITE sip:730730@192.168.0.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bKec73be0131f94e676 Max-Forwards: 70 From: <sip:0049xxx@192.168.0.220:5060>;tag=eff3f8af9c To: <sip:xxx@192.168.0.200> Call-ID: a2918e67b77c0aed CSeq: 18467 INVITE Contact: <sip:00xxx@192.168.0.220:5060> Supported: replaces User-Agent: Patton SN4638 5BIS 00A0BA076E36 R6.2 2012-07-13 H323 SIP BRI M5T SIP Stack/4.0.30.30 Content-Type: application/sdp Content-Length: 214 v=0 o=MxSIP 0 9194 IN IP4 192.168.0.220 s=SIP Call c=IN IP4 192.168.0.220 t=0 0 m=audio 5304 RTP/AVP 0 8 18 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=sendrecv [5] 2013/05/27 15:21:42: SIP Tx udp:192.168.0.220:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bKec73be0131f94e676 From: <sip:00xxx@192.168.0.220:5060>;tag=eff3f8af9c To: <sip:xxx@192.168.0.200>;tag=241c555897 Call-ID: a2918e67b77c0aed CSeq: 18467 INVITE Content-Length: 0 [3] 2013/05/27 15:21:42: Hunt group 0 wants to add 4 members [3] 2013/05/27 15:21:42: Call port 193: update_codecs for a2918e67b77c0aed: codec_preference size 4, available codecs list is empty [5] 2013/05/27 15:21:42: SIP Tx udp:192.168.0.220:5060: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bKec73be0131f94e676 From: <sip:00xxx@192.168.0.220:5060>;tag=eff3f8af9c To: <sip:xxx@192.168.0.200>;tag=241c555897 Call-ID: a2918e67b77c0aed CSeq: 18467 INVITE Contact: <sip:xxx@192.168.0.200:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/5.0.10 Content-Length: 0 After a rebooting snomONE the error occurs on rare. We use the snom pbx for a emergency hotline (poison center). Every incoming call could be very important ! System General: Version: 5.0.10 (snom ONE mini) Created on: May 14 2013 15:07:42 License Status: snom ONE twenty (pbx.ggizef.lokal) License Duration: Permanent Additional license information: Domains: 1/1, Calls: 0/10, G729A: 10, Extensions: 20/20, Attendants: 1/4, Callingcards: 0/2, Hunt Groups: 3/4, Paging Groups: 0/0, Service Flags: 1/20, IVR Nodes: 0/20, Agent Groups: 0/1, Conference Rooms: 1/2, CO Lines: 0/20, Adhoc Recording, Barge, Listen, Whisper, Trunk Accounting, Prepaid, Fax2Email Working Directory: /usr/local/snomONE DNS Servers: 192.168.0.154 IP Addresses: 127.0.0.1 192.168.0.200 CDR: Duration(360d): trunk = 15689, extension = 215, ivr = 17643 Calls: Total 1346/107, Active 0/0 Calls SIP packet statistics: Tx: 1459338 Rx: 1461857 Emails: Successful sent: 238 Unsuccessful attempts: 0 Available file system space: 37% Uptime: 2013/5/27 14:54:39 (uptime: 11 days 04:04:26) (126441 137560-0) WAV cache: 3 Number of HTTP sessions: Sessions: 7; Threads: SIP=15, HTTP=5 Domain Statistics: Total Domains: 1, Total Accounts: 27 regards gifti
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hunt-group members continue to ring after the CANCEL message
gifti replied to gifti's topic in Hunt Group Setup
After upate to 5.0.8 the problem is gone ! -
Since 5.0.8 the problem is resolved !
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That sounds good
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I've changed it from <max_mb_duration>0</max_mb_duration> to <max_mb_duration>15</max_mb_duration> . Problem persists. After 5 min and a few seconds I get an error. <133>1 2013-03-25T12:19:10+01:00 snomonemini SIP - - - SIP Tx udp:192.168.0.220:5060: INVITE sip:... ... <132>1 2013-03-25T12:19:17+01:00 snomonemini Dropping - - - Dropping HDLC byte
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It is a "no registration" PSTN Gateway configuration for Patton!? I've figured out, that the issue happens in both directions and it's not only a problem with the trunk configuration. I also sometimes get the message "Unsupported Media Type" when i try to dial my mailbox or an internal/external number. When I dial my own mailbox a few times, I sporadic get an "Unsupported Media Type" displayed on the SNOM370 like this: <133>1 2013-03-19T12:43:41+01:00 snomonemini SIP - - - SIP Rx tls:192.168.0.202:1057: INVITE sip:21@pbx.ggizef.lokal;user=phone SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.0.202:1057;branch=z9hG4bK-mw2zkbllida2;rport#015#012From: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>;tag=aum8svrlcy#015#012To: <sip:21@pbx.ggizef.lokal;user=phone>#015#012Call-ID: 51485d7c70dc-m9xajgf82xni#015#012CSeq: 1 INVITE#015#012Max-Forwards: 70#015#012Contact: <sip:21@192.168.0.202:1057;transport=tls;line=ol5l28iy>;reg-id=1#015#012X-Serialnumber: 0004133A8410#015#012P-Key-Flags: resolution="31x13", keys="4"#015#012User-Agent: snom370/8.7.3.19#015#012Accept: application/sdp#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE#015#012Allow-Events: talk, hold, refer, call-info#015#012Supported: timer, 100rel, replaces, from-change#015#012Session-Expires: 3600;refresher=uas#015#012Min-SE: 90#015#012Proxy-Require: buttons#015#012Content-Type: application/sdp#015#012Content-Length: 426#015#012#015#012v=0#015#012o=root 579784838 579784838 IN IP4 192.168.0.202#015#012s=call#015#012c=IN IP4 192.168.0.202#015#012t=0 0#015#012m=audio 58122 RTP/AVP 0 8 18 101#015#012a=crypto: <135>1 2013-03-19T12:43:41+01:00 snomonemini Packet - - - Packet authenticated by transport layer <135>1 2013-03-19T12:43:41+01:00 snomonemini Using - - - Using outbound proxy sip:192.168.0.202:1057;transport=tls because of flow-label <135>1 2013-03-19T12:43:41+01:00 snomonemini Last - - - Last message repeated 3 times <133>1 2013-03-19T12:43:41+01:00 snomonemini SIP - - - SIP Tx tls:192.168.0.202:1057: SIP/2.0 100 Trying#015#012Via: SIP/2.0/TLS 192.168.0.202:1057;branch=z9hG4bK-mw2zkbllida2;rport=1057#015#012From: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>;tag=aum8svrlcy#015#012To: <sip:21@pbx.ggizef.lokal;user=phone>;tag=73e7bda774#015#012Call-ID: 51485d7c70dc-m9xajgf82xni#015#012CSeq: 1 INVITE#015#012Content-Length: 0#015#012#015 <135>1 2013-03-19T12:43:41+01:00 snomonemini Incoming - - - Incoming call: Request URI sip:21@pbx.ggizef.lokal;user=phone, To is <sip:21@pbx.ggizef.lokal;user=phone> <135>1 2013-03-19T12:43:41+01:00 snomonemini Set - - - Set the To domain based on From user 21@pbx.ggizef.lokal <134>1 2013-03-19T12:43:41+01:00 snomonemini Call - - - Call port 201: set_codecs for 51485d7c70dc-m9xajgf82xni codecs "", codec_preference count 4 <135>1 2013-03-19T12:43:41+01:00 snomonemini Play - - - Play audio_de/mb_main_menu.wav audio_de/mb_main_menu1.wav audio_de/mb_main_menu2.wav audio_de/mb_main_menu3.wav audio_de/mb_main_menu4.wav audio_de/mb_enter_choice2.wav audio_de/bi_press_5.wav audio_de/mb_main_menu9.wav space50, caching false <135>1 2013-03-19T12:43:41+01:00 snomonemini call - - - call port 201: state code from 0 to 200 <135>1 2013-03-19T12:43:41+01:00 snomonemini Call - - - Call port 201: update_codecs for 51485d7c70dc-m9xajgf82xni: Other side does not support codec PCMU/8000 <135>1 2013-03-19T12:43:41+01:00 snomonemini Call - - - Call port 201: update_codecs for 51485d7c70dc-m9xajgf82xni: Other side does not support codec PCMA/8000 <135>1 2013-03-19T12:43:41+01:00 snomonemini Call - - - Call port 201: update_codecs for 51485d7c70dc-m9xajgf82xni: Other side does not support codec G729/8000 <135>1 2013-03-19T12:43:41+01:00 snomonemini Call - - - Call port 201: update_codecs for 51485d7c70dc-m9xajgf82xni: Other side does not support codec telephone-event/8000 <131>1 2013-03-19T12:43:41+01:00 snomonemini Call - - - Call port 201: update_codecs for 51485d7c70dc-m9xajgf82xni: codec_preference size 4, available codecs list is empty <133>1 2013-03-19T12:43:41+01:00 snomonemini send_connected - - - send_connected: available codec list is empty for 51485d7c70dc-m9xajgf82xni <133>1 2013-03-19T12:43:41+01:00 snomonemini SIP - - - SIP Tx tls:192.168.0.202:1057: SIP/2.0 415 Unsupported Media Type#015#012Via: SIP/2.0/TLS 192.168.0.202:1057;branch=z9hG4bK-mw2zkbllida2;rport=1057#015#012From: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>;tag=aum8svrlcy#015#012To: <sip:21@pbx.ggizef.lokal;user=phone>;tag=73e7bda774#015#012Call-ID: 51485d7c70dc-m9xajgf82xni#015#012CSeq: 1 INVITE#015#012Contact: <sip:21@192.168.0.200:5061;transport=tls>#015#012Supported: 100rel, replaces, norefersub#015#012Allow-Events: refer#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE#015#012Accept: application/sdp#015#012User-Agent: snomONE/5.0.5#015#012Content-Length: 0#015#012#015 <133>1 2013-03-19T12:43:41+01:00 snomonemini The - - - The call port 201 - 30 seconds callback set for force cleanup The same issue happens spordic with incoming calls on the trunk BRI_2_3_4_Bidirectional. <133>1 2013-03-19T11:35:41+01:00 snomonemini SIP - - - SIP Rx udp:192.168.0.220:5060: INVITE sip:730730@192.168.0.200 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK5f3fe8836cb72b10c#015#012Max-Forwards: 70#015#012From: <sip:00493617315293@192.168.0.220:5060>;tag=91f426b026#015#012To: <sip:730730@192.168.0.200>#015#012Call-ID: 6b7f622466197438#015#012CSeq: 21336 INVITE#015#012Contact: <sip:00493617315293@192.168.0.220:5060>#015#012Supported: replaces#015#012User-Agent: Patton SN4638 5BIS 00A0BA076E36 R6.2 2012-07-13 H323 SIP BRI M5T SIP Stack/4.0.30.30#015#012Content-Type: application/sdp#015#012Content-Length: 269#015#012#015#012v=0#015#012o=MxSIP 0 163 IN IP4 192.168.0.220#015#012s=SIP Call#015#012c=IN IP4 192.168.0.220#015#012t=0 0#015#012m=audio 5038 RTP/AVP 0 8 18 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:18 annexb=no#015#012a=fmtp:101 0-16#015#012a=sendrecv#015 <133>1 2013-03-19T11:35:41+01:00 snomonemini SIP - - - SIP Tx udp:192.168.0.220:5060: SIP/2.0 100 Trying#015#012Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK5f3fe8836cb72b10c#015#012From: <sip:00493617315293@192.168.0.220:5060>;tag=91f426b026#015#012To: <sip:730730@192.168.0.200>;tag=927ca7a609#015#012Call-ID: 6b7f622466197438#015#012CSeq: 21336 INVITE#015#012Content-Length: 0#015#012#015 <134>1 2013-03-19T11:35:41+01:00 snomonemini Call - - - Call port 155: set_codecs for 6b7f622466197438 codecs "", codec_preference count 4 <135>1 2013-03-19T11:35:41+01:00 snomonemini Play - - - Play audio_moh/noise.wav, caching false <135>1 2013-03-19T11:35:41+01:00 snomonemini call - - - call port 155: state code from 0 to 180 <135>1 2013-03-19T11:35:41+01:00 snomonemini Call - - - Call port 155: update_codecs for 6b7f622466197438: Other side does not support codec PCMU/8000 <135>1 2013-03-19T11:35:41+01:00 snomonemini Call - - - Call port 155: update_codecs for 6b7f622466197438: Other side does not support codec PCMA/8000 <135>1 2013-03-19T11:35:41+01:00 snomonemini Call - - - Call port 155: update_codecs for 6b7f622466197438: Other side does not support codec G729/8000 <135>1 2013-03-19T11:35:41+01:00 snomonemini Call - - - Call port 155: update_codecs for 6b7f622466197438: Other side does not support codec telephone-event/8000 <131>1 2013-03-19T11:35:41+01:00 snomonemini Call - - - Call port 155: update_codecs for 6b7f622466197438: codec_preference size 4, available codecs list is empty <133>1 2013-03-19T11:35:41+01:00 snomonemini Available - - - Available codec list is empty for 6b7f622466197438 <133>1 2013-03-19T11:35:41+01:00 snomonemini SIP - - - SIP Tx udp:192.168.0.220:5060: SIP/2.0 415 Unsupported Media Type#015#012Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK5f3fe8836cb72b10c#015#012From: <sip:00493617315293@192.168.0.220:5060>;tag=91f426b026#015#012To: <sip:730730@192.168.0.200>;tag=927ca7a609#015#012Call-ID: 6b7f622466197438#015#012CSeq: 21336 INVITE#015#012Contact: <sip:730730@192.168.0.200:5060;transport=udp>#015#012Supported: 100rel, replaces, norefersub#015#012Allow-Events: refer#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE#015#012Accept: application/sdp#015#012User-Agent: snomONE/5.0.5#015#012Content-Length: 0#015#012#015 Any ideas? I've changed the ProxyAddress into sip:patton.ggizef.lokal:5060 but the problem still persists. Here is one of the 95% bug-free calls. <133>1 2013-03-19T13:56:46+01:00 snomonemini SIP - - - SIP Rx udp:192.168.0.220:5060: INVITE sip:7307321@192.168.0.200 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK5e5b883ef5b2aa7e6#015#012Max-Forwards: 70#015#012From: <sip:00493617315293@192.168.0.220:5060>;tag=d03e8bd9ed#015#012To: <sip:7307321@192.168.0.200>#015#012Call-ID: e8e5ad9640944b34#015#012CSeq: 9171 INVITE#015#012Contact: <sip:00493617315293@192.168.0.220:5060>#015#012Supported: replaces#015#012User-Agent: Patton SN4638 5BIS 00A0BA076E36 R6.2 2012-07-13 H323 SIP BRI M5T SIP Stack/4.0.30.30#015#012Content-Type: application/sdp#015#012Content-Length: 269#015#012#015#012v=0#015#012o=MxSIP 0 193 IN IP4 192.168.0.220#015#012s=SIP Call#015#012c=IN IP4 192.168.0.220#015#012t=0 0#015#012m=audio 5072 RTP/AVP 0 8 18 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:18 annexb=no#015#012a=fmtp:101 0-16#015#012a=sendrecv#015 <135>1 2013-03-19T13:56:46+01:00 snomonemini Resolve - - - Resolve 792009: aaaa udp 192.168.0.220 5060 <135>1 2013-03-19T13:56:46+01:00 snomonemini Resolve - - - Resolve 792009: a udp 192.168.0.220 5060 <135>1 2013-03-19T13:56:46+01:00 snomonemini Resolve - - - Resolve 792009: udp 192.168.0.220 5060 <133>1 2013-03-19T13:56:46+01:00 snomonemini SIP - - - SIP Tx udp:192.168.0.220:5060: SIP/2.0 100 Trying#015#012Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK5e5b883ef5b2aa7e6#015#012From: <sip:00493617315293@192.168.0.220:5060>;tag=d03e8bd9ed#015#012To: <sip:7307321@192.168.0.200>;tag=32eb02a52a#015#012Call-ID: e8e5ad9640944b34#015#012CSeq: 9171 INVITE#015#012Content-Length: 0#015#012#015 <133>1 2013-03-19T13:56:46+01:00 snomonemini Call-leg - - - Call-leg 223: Sending RTP for e8e5ad9640944b34 to 192.168.0.220:5072, codec not set yet <135>1 2013-03-19T13:56:46+01:00 snomonemini Incoming - - - Incoming call: Request URI sip:7307321@192.168.0.200, To is <sip:7307321@192.168.0.200> <135>1 2013-03-19T13:56:46+01:00 snomonemini Set - - - Set the To domain based on To user 21@pbx.ggizef.lokal <134>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 223: set_codecs for e8e5ad9640944b34 codecs "0 8 18", codec_preference count 4 <134>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 224: set_codecs for 875a3b0b@pbx codecs "", codec_preference count 4 <135>1 2013-03-19T13:56:46+01:00 snomonemini Using - - - Using outbound proxy sip:192.168.0.202:1057;transport=tls because of flow-label <135>1 2013-03-19T13:56:46+01:00 snomonemini call - - - call port 224: state code from 0 to 100 <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 224: update_codecs for 875a3b0b@pbx: adding codec PCMU/8000 to available list <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 224: update_codecs for 875a3b0b@pbx: adding codec PCMA/8000 to available list <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 224: update_codecs for 875a3b0b@pbx: adding codec G729/8000 to available list <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 224: update_codecs for 875a3b0b@pbx: codec_preference size 4, available codecs size 4 <133>1 2013-03-19T13:56:46+01:00 snomonemini SIP - - - SIP Tx tls:192.168.0.202:1057: INVITE sip:21@192.168.0.202:1057;transport=tls;line=ol5l28iy SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.0.200:5061;branch=z9hG4bK-bae6ffc35cbb42fcd5ff49ca0ca62d4e;rport#015#012From: <sip:7315293@pbx.ggizef.lokal:5060;user=phone>;tag=232636715#015#012To: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>#015#012Call-ID: 875a3b0b@pbx#015#012CSeq: 2695 INVITE#015#012Max-Forwards: 70#015#012Contact: <sip:21@192.168.0.200:5061;transport=tls>#015#012Supported: 100rel, replaces, norefersub#015#012Allow-Events: refer#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE#015#012Accept: application/sdp#015#012User-Agent: snomONE/5.0.5#015#012Alert-Info: <http://127.0.0.1/Bellcore-dr3>#015#012Content-Type: application/sdp#015#012Content-Length: 406#015#012#015#012v=0#015#012o=- 861259913 861259913 IN IP4 192.168.0.200#015#012s=-#015#012c=IN IP4 192.168.0.200#015#012t=0 0#015#012m=audio 59610 RTP/AVP 0 8 18 101#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:RokvG5jvSh3iKEgiYMZ+WA71XRjICZl7NCAeGHnD#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:18 G729/8000#015#012a=fmtp:18 annexb=no#015 <135>1 2013-03-19T13:56:46+01:00 snomonemini Play - - - Play audio_moh/noise.wav, caching true <135>1 2013-03-19T13:56:46+01:00 snomonemini call - - - call port 223: state code from 0 to 100 <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 223: update_codecs for e8e5ad9640944b34: adding codec PCMU/8000 to available list <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 223: update_codecs for e8e5ad9640944b34: adding codec PCMA/8000 to available list <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 223: update_codecs for e8e5ad9640944b34: adding codec G729/8000 to available list <135>1 2013-03-19T13:56:46+01:00 snomonemini Call - - - Call port 223: update_codecs for e8e5ad9640944b34: codec_preference size 4, available codecs size 4 <133>1 2013-03-19T13:56:46+01:00 snomonemini SIP - - - SIP Rx tls:192.168.0.202:1057: SIP/2.0 100 Trying#015#012Via: SIP/2.0/TLS 192.168.0.200:5061;branch=z9hG4bK-bae6ffc35cbb42fcd5ff49ca0ca62d4e;rport=5061#015#012From: <sip:7315293@pbx.ggizef.lokal:5060;user=phone>;tag=232636715#015#012To: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>;tag=oirh8epwhw#015#012Call-ID: 875a3b0b@pbx#015#012CSeq: 2695 INVITE#015#012Contact: <sip:21@192.168.0.202:1057;transport=tls;line=ol5l28iy>;reg-id=1#015#012Content-Length: 0#015#012#015 <135>1 2013-03-19T13:56:46+01:00 snomonemini Message - - - Message repetition, packet dropped <133>1 2013-03-19T13:56:47+01:00 snomonemini SIP - - - SIP Rx tls:192.168.0.202:1057: SIP/2.0 180 Ringing#015#012Via: SIP/2.0/TLS 192.168.0.200:5061;branch=z9hG4bK-bae6ffc35cbb42fcd5ff49ca0ca62d4e;rport=5061#015#012From: <sip:7315293@pbx.ggizef.lokal:5060;user=phone>;tag=232636715#015#012To: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>;tag=oirh8epwhw#015#012Call-ID: 875a3b0b@pbx#015#012CSeq: 2695 INVITE#015#012Contact: <sip:21@192.168.0.202:1057;transport=tls;line=ol5l28iy>;reg-id=1#015#012Require: 100rel#015#012RSeq: 1#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE#015#012Allow-Events: talk, hold, refer, call-info#015#012Content-Length: 0#015#012#015 <133>1 2013-03-19T13:56:47+01:00 snomonemini SIP - - - SIP Tx tls:192.168.0.202:1057: PRACK sip:21@192.168.0.202:1057;transport=tls;line=ol5l28iy SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.0.200:5061;branch=z9hG4bK-68fd95afa82a61df3ae3c6dd654816cc;rport#015#012From: <sip:7315293@pbx.ggizef.lokal:5060;user=phone>;tag=232636715#015#012To: "Wolfgang Ackermann" <sip:21@pbx.ggizef.lokal>;tag=oirh8epwhw#015#012Call-ID: 875a3b0b@pbx#015#012CSeq: 2696 PRACK#015#012Max-Forwards: 70#015#012Contact: <sip:21@192.168.0.200:5061;transport=tls>#015#012RAck: 1 2695 INVITE#015#012Content-Length: 0#015#012#015 <135>1 2013-03-19T13:56:47+01:00 snomonemini Play - - - Play audio_de/ringback.wav, caching true <135>1 2013-03-19T13:56:47+01:00 snomonemini call - - - call port 223: state code from 100 to 180 <135>1 2013-03-19T13:56:47+01:00 snomonemini Resolve - - - Resolve 792012: aaaa udp 192.168.0.220 5060 <135>1 2013-03-19T13:56:47+01:00 snomonemini Resolve - - - Resolve 792012: a udp 192.168.0.220 5060 <135>1 2013-03-19T13:56:47+01:00 snomonemini Resolve - - - Resolve 792012: udp 192.168.0.220 5060 <133>1 2013-03-19T13:56:47+01:00 snomonemini SIP - - - SIP Tx udp:192.168.0.220:5060: SIP/2.0 180 Ringing#015#012Via: SIP/2.0/UDP 192.168.0.220:5060;branch=z9hG4bK5e5b883ef5b2aa7e6#015#012From: <sip:00493617315293@192.168.0.220:5060>;tag=d03e8bd9ed#015#012To: <sip:7307321@192.168.0.200>;tag=32eb02a52a#015#012Call-ID: e8e5ad9640944b34#015#012CSeq: 9171 INVITE#015#012Contact: <sip:7307321@192.168.0.200:5060;transport=udp>#015#012Supported: 100rel, replaces, norefersub#015#012Allow-Events: refer#015#012Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE#015#012Accept: application/sdp#015#012User-Agent: snomONE/5.0.5#015#012Content-Length: 0#015#012#015
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The \1 part of the DialExtension pattern was escaped after copy & paste . It really looks like this: !73073([0-9]{1,10}$)!\1!t!0
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Hello, I've got the following configuration: 3 x ISDN PMP <> Patton 4638 <> snomONE mini 5.0.5 Up to 5% of incoming calls won't be connectet to the incoming SIP-Interface to snomONE (IF_SIP_730730_PHONE). I get an error message in the patton syslog: <195>1 2013-03-18T14:30:53+01:00 192.168.0.220 SIP - - - SIP: [EP IF_SIP_730730_PHONE-00ad5190 SES 0x106c4e8] REMOVED DYNAMIC REGISTRAR FAILED After that, the call uses the 2nd destination (IF_ISDN_00_DEFAULT). It is an Default-ISDN-Phon which works even when the power fail. Config of the incoming HG on the Patton Gateway: service hunt-group HG_SIP_ISDN_730730_IN_PHONE timeout 2 allows-push-back drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable drop-cause unallocated-number unavailable drop transparent route call 1 dest-interface IF_SIP_730730_PHONE route call 2 dest-interface IF_ISDN_00_DEFAULT Why is the first destination-interface skipped sporadically ? Trunk Config snomONE: # Trunk 10 in domain pbx.ggizef.lokal Name: BRI_2_3_4_Bidirectional Type: gateway To: sip RegPass: ******** Direction: Disabled: false Global: false Display: RegAccount: RegRegistrar: RegKeep: RegUser: Icid: Require: OutboundProxy: 192.168.0.220:5060 Ani: DialExtension: !73073([0-9]{1,10}$)!!t!0 Trusted: false AcceptRedirect: false RfcRtp: true Analog: true RtpBegin: RtpEnd: Prack: false SendEmail: UseUuid: false Ring180: true Failover: never HeaderRequestUri: {request-uri} HeaderFrom: {from} HeaderTo: {to} HeaderPai: {trunk} HeaderPpi: HeaderRpi: HeaderPrivacy: HeaderRpiCharging: BlockCidPrefix: Glob: RequestTimeout: Codecs: CodecLock: true DtmfMode: Expires: 3600 Fraction: 128 Minimum: 10 FromUser: Tel: true TranscodeDtmf: false AssociatedAddresses: InterOffice: false DialPlan: UseEpid: false CidUpdate: Ignore18xSDP: true UserHdr: Diversion: {rfc} CoBusy: 500 Line Unavailable Colines: DialogPermission: And the SIP-Interface on the Patton Gateway: interface sip IF_SIP_730730_PHONE bind context sip-gateway GW_SIP_730730_PHONE route call dest-table RT_TO_ISDN_CLIP remote 192.168.0.200 early-connect no call-transfer pull-in call-reroute accept call-reroute emit privacy address-translation outgoing-call diversion-header host-part call regards gift
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You have to customize the dom_calls.htm. You can do this on the website. Domain View > Customize > Type = Webpages > dom_calls.htm <?xml version="1.0" encoding="UTF-8" ?> <!DOCTYPE html PUBLIC "-//W3C//DTD Xhtml 1.0 Transitional//EN" "http://www.w3.org/tr/xhtml1/DTD/xhtml1-transitional.dtd"> <html xmlns="http://www.w3.org/1999/xhtml"> <!-- {ssi permission domain}{ssi load domains domain name} --> <head> <title>{lng add} [{ssi varh name}]</title> <meta http-equiv="Content-Type" content="text/html;charset=utf-8" /> {ssi if ui_style new}<link href="style_v5.css" type="text/css" rel="stylesheet" />{ssi fi ui_style new}{ssi ifn ui_style new}<link href="style.css" type="text/css" rel="stylesheet" />{ssi fin ui_style new} <script type="text/javascript" src="call_list_scripts.js"></script> </head> <body> {ssi set menu status}{ssi set submenu calls}{ssi file dom_header.htm} <table width="100%" align="center" cellpadding="1"> <tr><td class="headerText" valign="middle" height="70"> {lng 2}: {ssi help DSTcal1 img/help2.gif help} </td></tr> <tr><td><div id="active_calls"></div></td></tr> <tr><td><div id="output"></div></td></tr> </table> {ssi file dom_footer.htm} </body> </html> regards gifti
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Are you able to recieve more than 10 pages?
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Yeah, it only was the <div /> tag. Now it works but: - no css for the table? - no possibility to hang up ?
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Hello, i've tested with version 5.0.6. The active calls table isn't working. I discoverd one bug in html code. Old: ... <tr><td><div id="active_calls" /></td></tr> ... New: ... <tr><td><div id="active_calls"></div></td></tr> ... Found here But when I changed the template I'don't get a table or a DISCONNECTED message, too. Seems to be an error in in the scipt call_list_scripts.js or on server side. Mozilla 19.0.2 / IE 9.0.8 / Chrome 25.0.1364.152 regards gifti
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...
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Hello all, we use Fax2Email and we are very happy with that. But if someone sends us a bigger fax with more than about 5 or 6 pages, the mailbox interrupts the call exactly after 5 minutes. Maximum voicemail duration is set to 0. I tested with 15 pages from a G3 fax. After every fifth minute the connection interrupts. Then the fax starts with the remaining pages. So the whole fax was divided into 3 parts. The last part (3 pages) has been sended per mail (notify with attachment from snomONE). The other two parts are only visible in the mailbox. When i download the message.wav files and change the extension into .pdf, i could only open the last part. The other two parts are available but the pdf file is defect. regards gifti
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hunt-group members continue to ring after the CANCEL message
gifti replied to gifti's topic in Hunt Group Setup
Upgrading 8.4.35? I use the latest release 8.7.3.19. I've change the transport layer to TCP and then back to TLS. Problem persists. No, it is the evil pbx . If you take a look at the attachment trace_ghost_ringing_com.pdf 1.144269 => INVITE 5608 was sended before the request was terminated 1.425376 => snomONE sends a CANCEL ... the member of the huntgroup stops ringing but the other HG members 1.424504 => INVITE 10376 was sended after the request was terminated 1.424885 => INVITE 30970 was sended after the request was terminated ... => snomONE sends no CANCEL ... phone rings and rings I would suggest two Solutions: snomONE stops sending INVITEs to huntgroup members when a 487 Request Terminated is coming in (if the INVITE process is not yet complete) snomONE recognizes all INVITE messages after the "487 Request Terminated" and subsequently sends the CANCEL messages -
hunt-group members continue to ring after the CANCEL message
gifti replied to gifti's topic in Hunt Group Setup
I've deactivated TLS to create a readable SIP Logfile. In my opinion, the transport layer doesn't matter ... -
Tried to create CO-Lines, but the link dom_acclist.htm seems to be wrong ... snomONE 5.0.5 regards gifti
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I tried this: - opened my webbrowser - typed in http://admin:password@192.168.0.202/command.htm?key_dtmf=4711%23 - pick up the SNOM phone 192.168.0.202 (SNOM 370) - dialed a foreign mailbox (PIN is necessary) - hit enter in the webbrowser (http request is working) - ... sounds like 5 dtmf tones in the earphone - works fine regards gifti
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Habe einen Topic im englischen Teil des Forums geöffnet. Bitte dort weiter antworten. Danke Gifti
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hunt-group members continue to ring after the CANCEL message
gifti replied to gifti's topic in Hunt Group Setup
I generated two test scenarios without TLS connection between pbx and the phones. 192.168.0.220 = Patton 192.168.0.200 = snomONE 5.0.5 192.168.0.206,207 and 214 = snom370-SIP 8.7.3.19 in huntgoup stage one normal cancel message after 5 seconds ringing(between line 9 and 45): trace_normal.pdf cancel_normal_cut.txt cancel after 20 milliseconds ringing (between line 9 and 10) with ghost ringing on all stage one extensions : trace_ghost_ringing.pdf cancel_ghost_ringing_cut.txt -
Try to replace the hash with %23.
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hunt-group members continue to ring after the CANCEL message
gifti replied to gifti's topic in Hunt Group Setup
Thank you for your quick reponse ! - latest build 5.0.5 is installed ... problem still exists. - the point is, that only when the call ist hung up immediately (milliseconds) after dialing, the issue happens - I dial from outside, wait half a second and put the phone down - the snomONE get's the ringing message a bit later and the the HG stage one is ringing - there is no matter how much stages you use - the first stage continues ringing, whether it's one phone or more phones in the stage - there is no active call on the patton, it's a ghost ringing ... - when I wait a second or more and put the phone down ... everything works fine - the HG is ringing and stops ringing immediately after hang up - sip trace will follow regards gifti -
Ja, zwischen PBX und Telefonen läuft alles über Port 5061 (TLS). Die snomONE ist übrigens auch auf dem aktuellen Stand 5.0.5. Grüße Gifti