gifti

I have 5.0.10a installed. The registration of the extensions after restart runs much faster. It would be nice if the "codecs list is empty" error would be finally resolved. Thank you! I'll be watching the error continues

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

After upate to 5.0.8 the problem is gone !

Since 5.0.8 the problem is resolved !

That sounds good

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

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

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

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

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

Are you able to recieve more than 10 pages?

Yeah, it only was the <div /> tag. Now it works but: - no css for the table? - no possibility to hang up ?

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

...

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

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

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

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

Habe einen Topic im englischen Teil des Forums geöffnet. Bitte dort weiter antworten. Danke Gifti

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.

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