pbx support Posted June 29, 2011 Report Share Posted June 29, 2011 It's been a while since we released the last bug fix version(2011-4.2.0.3981) for snomONE. In the meanwhile, we heard the feedback from you in the form of forum requests, support tickets, etc. We tried to resolve most of the known issues during this time and as a result of this is the latest bug fix release 2011-4.2.1.4025. For details such as release notes and download links please refer the below link http://wiki.snomone.com/index.php?title=Release_notes Thank you all!!! Quote Link to comment Share on other sites More sharing options...
fonny Posted June 29, 2011 Report Share Posted June 29, 2011 Dear, Thanks for the update. After update I can no longer start the MacOSX snomOne Free version. Key is not correct for this version ! But on the wiki it states it is for all versions. Any idea what can be wrong regards Fonny Quote Link to comment Share on other sites More sharing options...
venom Posted June 29, 2011 Report Share Posted June 29, 2011 Thank you for the update. How do we update the SnomOne Plus? There seems to be no link on WIKI. Thank you. Venom Quote Link to comment Share on other sites More sharing options...
fonny Posted June 29, 2011 Report Share Posted June 29, 2011 Vernom, Just follow the link on the first post. Quote Link to comment Share on other sites More sharing options...
pbx support Posted June 30, 2011 Author Report Share Posted June 30, 2011 Dear, Thanks for the update. After update I can no longer start the MacOSX snomOne Free version. Key is not correct for this version ! But on the wiki it states it is for all versions. Any idea what can be wrong regards Fonny There should not be anything special here. We will double check. Quote Link to comment Share on other sites More sharing options...
pbx support Posted June 30, 2011 Author Report Share Posted June 30, 2011 There was mistake during the upload. We uploaded the pbxnsip binary instead of the snomONE binary. Sorry about that. Now we have replaced it with the proper binary. If you download the file now, you will have the proper version. Quote Link to comment Share on other sites More sharing options...
Vodia support Posted June 30, 2011 Report Share Posted June 30, 2011 V, if you want to update the snomONE plus here is how. Upgrading/Updating the snomONE Once the snomONE is installed and running succefully, the upgrade to a newer version is very simple. Download the new version to snomONE directory Make the file executable using chmod a+x <new file> command. Stop the snomONE using service snomONE stop command. Point the link to new binary using ln -sf <new file> snomONE-ctrl command. Start the snomONE using service snomONE start command. This procedure has to be followed anytime you want to upgrade (or downgrade) the snomONE. Quote Link to comment Share on other sites More sharing options...
p800aul Posted June 30, 2011 Report Share Posted June 30, 2011 It's been a while since we released the last bug fix version(2011-4.2.0.3981) for snomONE. In the meanwhile, we heard the feedback from you in the form of forum requests, support tickets, etc. We tried to resolve most of the known issues during this time and as a result of this is the latest bug fix release 2011-4.2.1.4025. For details such as release notes and download links please refer the below link http://wiki.snomone.com/index.php?title=Release_notes Thank you all!!! Any reason why the patton trunk will not dial out using this the sipgate still works. So previous version 2011-4.2.0.3981 worked incoming - outgoing sipgate and worked incoming - outgoing patton 2xfxo Upgrade to 2011-4.2.1.4025, sipgate incoming - outgoing, patton incoming only - no outgoing (engaged tone). Downgrade back to 2011-4.2.0.3981 everything fine again. Regards Paul patton trunk set up: # Trunk 5 in domain localhost Name: Patton Type: register To: sip RegPass: ******** Direction: Disabled: false Global: false Display: RegAccount: RegRegistrar: 192.168.1.200 RegKeep: RegUser: Icid: Require: OutboundProxy: 192.168.1.200 Ani: DialExtension: 72 Prefix: Trusted: false AcceptRedirect: false RfcRtp: false Analog: false SendEmail: UseUuid: false Ring180: false Failover: only_5xx Privacy: false Glob: RequestTimeout: Codecs: CodecLock: true Expires: 3600 FromUser: Tel: true TranscodeDtmf: false AssociatedAddresses: InterOffice: false DialPlan: Colines: DialogPermission: Quote Link to comment Share on other sites More sharing options...
pbx support Posted July 1, 2011 Author Report Share Posted July 1, 2011 Do you have SIP INVITE that is sent to patton from PBX using .4025 & .3981? There isn't much changed in the outbound behavior. Maybe just 10 or 11 digit (with or without +) issue. Quote Link to comment Share on other sites More sharing options...
p800aul Posted July 1, 2011 Report Share Posted July 1, 2011 There you go Thanks .3981 [6] 2011/07/01 18:09:22: Received bindRequest for user localhost\48 [5] 2011/07/01 18:09:25: SIP Rx tls:192.168.1.8:2778: INVITE sip:819161@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.8:2778;branch=z9hG4bK-h41any7bdh3p;rport From: "Study" <sip:48@localhost>;tag=qzv6db7x6u To: <sip:819161@localhost;user=phone> Call-ID: 3c641d1fe2f7-4xrvrha9is0i CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:48@192.168.1.8:2778;transport=tls;line=mjvwc7ij>;reg-id=1 X-Serialnumber: 00041336B86D P-Key-Flags: keys="3" User-Agent: snom300/8.4.31 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 520 v=0 o=root 391218360 391218360 IN IP4 192.168.1.8 s=call c=IN IP4 192.168.1.8 t=0 0 m=audio 56900 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:AbJQ3lgRvtJ7BbbTxRK15Rg3nXBsgROXGul4+G7J a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [8] 2011/07/01 18:09:25: Packet authenticated by transport layer [9] 2011/07/01 18:09:25: UDP: Opening socket on 0.0.0.0:58970 [9] 2011/07/01 18:09:25: UDP: Opening socket on 0.0.0.0:58971 [8] 2011/07/01 18:09:25: Could not find a trunk (2 trunks) [5] 2011/07/01 18:09:25: SIP Rx tls:192.168.1.8:2778: INVITE sip:819161@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.8:2778;branch=z9hG4bK-h41any7bdh3p;rport From: "Study" <sip:48@localhost>;tag=qzv6db7x6u To: <sip:819161@localhost;user=phone> Call-ID: 3c641d1fe2f7-4xrvrha9is0i CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:48@192.168.1.8:2778;transport=tls;line=mjvwc7ij>;reg-id=1 X-Serialnumber: 00041336B86D P-Key-Flags: keys="3" User-Agent: snom300/8.4.31 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 520 v=0 o=root 391218360 391218360 IN IP4 192.168.1.8 s=call c=IN IP4 192.168.1.8 t=0 0 m=audio 56900 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:AbJQ3lgRvtJ7BbbTxRK15Rg3nXBsgROXGul4+G7J a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [9] 2011/07/01 18:09:25: Using outbound proxy sip:192.168.1.8:2778;transport=tls because of flow-label [9] 2011/07/01 18:09:25: Last message repeated 3 times [6] 2011/07/01 18:09:25: Received bindRequest for user localhost\48 [5] 2011/07/01 18:09:25: SIP Tx tls:192.168.1.8:2778: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.1.8:2778;branch=z9hG4bK-h41any7bdh3p;rport=2778 From: "Study" <sip:48@localhost>;tag=qzv6db7x6u To: <sip:819161@localhost;user=phone>;tag=ebe544b72c Call-ID: 3c641d1fe2f7-4xrvrha9is0i CSeq: 1 INVITE Content-Length: 0 [7] 2011/07/01 18:09:25: Set packet length to 20 [6] 2011/07/01 18:09:25: Sending RTP for 3c641d1fe2f7-4xrvrha9is0i to 192.168.1.8:56900, codec not set yet [8] 2011/07/01 18:09:25: Call from an user 48 [8] 2011/07/01 18:09:25: To is <sip:819161@localhost;user=phone>, user 0, domain 1 [8] 2011/07/01 18:09:25: From user 48 [8] 2011/07/01 18:09:25: Set the To domain based on From user 48@localhost [8] 2011/07/01 18:09:25: Call state for call object 13: idle [7] 2011/07/01 18:09:25: set_codecs: for 3c641d1fe2f7-4xrvrha9is0i codecs "", codec_preference count 6 [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^(999)@.*!sip:\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^0800([0-9]*)@.*!sip:0800\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^00([0-9]*)@.*!sip:00\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^07([0-9]*)@.*!sip:07\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^907([0-9]*)@.*!sip:07\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^900([0-9]*)@.*!sip:00\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 819161@localhost [5] 2011/07/01 18:09:25: Dialplan "Standard Dialplan": Match 819161@localhost to <sip:819161@192.168.1.200;user=phone> on trunk Patton [8] 2011/07/01 18:09:25: Play audio_moh/noise.wav [9] 2011/07/01 18:09:25: UDP: Opening socket on 0.0.0.0:59340 [9] 2011/07/01 18:09:25: UDP: Opening socket on 0.0.0.0:59341 [7] 2011/07/01 18:09:25: set_codecs: for 83572110@pbx codecs "", codec_preference count 6 [9] 2011/07/01 18:09:25: update_codecs for 83572110@pbx: adding codec pcmu/8000 to available list [9] 2011/07/01 18:09:25: update_codecs for 83572110@pbx: adding codec pcma/8000 to available list [9] 2011/07/01 18:09:25: update_codecs for 83572110@pbx: adding codec g722/8000 to available list [9] 2011/07/01 18:09:25: update_codecs for 83572110@pbx: adding codec g726-32/8000 to available list [9] 2011/07/01 18:09:25: update_codecs for 83572110@pbx: adding codec gsm/8000 to available list [9] 2011/07/01 18:09:25: update_codecs for 83572110@pbx: codec_preference size 6, available codecs size 6 [9] 2011/07/01 18:09:25: Resolve 12472: url sip:192.168.1.200 [9] 2011/07/01 18:09:25: Resolve 12472: udp 192.168.1.200 5060 [5] 2011/07/01 18:09:25: SIP Tx udp:192.168.1.200:5060: INVITE sip:819161@192.168.1.200;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-72902ae4db65e28d50e1980ed6df68bf;rport From: "Study" <sip:01246819161@localhost;user=phone>;tag=45409 To: <sip:819161@192.168.1.200;user=phone> Call-ID: 83572110@pbx CSeq: 900 INVITE Max-Forwards: 70 Contact: <sip:01246819161@192.168.1.13:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 Content-Type: application/sdp Content-Length: 327 v=0 o=- 41145 41145 IN IP4 192.168.1.13 s=- c=IN IP4 192.168.1.13 t=0 0 m=audio 59340 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 4025 [6] 2011/07/01 18:16:31: Received bindRequest for user localhost\48 [6] 2011/07/01 18:16:33: Last message repeated 2 times [7] 2011/07/01 18:16:33: SIP Rx tls:192.168.1.8:2782: INVITE sip:819161@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.8:2782;branch=z9hG4bK-v0svqzc2wfnk;rport From: "Study" <sip:48@localhost>;tag=xy3mypr1cv To: <sip:819161@localhost;user=phone> Call-ID: 3c641eccef30-an7w0fgu4eg2 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:48@192.168.1.8:2782;transport=tls;line=mjvwc7ij>;reg-id=1 X-Serialnumber: 00041336B86D P-Key-Flags: keys="3" User-Agent: snom300/8.4.31 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 522 v=0 o=root 1034786031 1034786031 IN IP4 192.168.1.8 s=call c=IN IP4 192.168.1.8 t=0 0 m=audio 54922 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XhQqTHLltypTJWC5vDrHpGfZkxH45okk1VH+jdSi a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [8] 2011/07/01 18:16:33: Packet authenticated by transport layer [9] 2011/07/01 18:16:33: UDP: Opening socket on 0.0.0.0:60914 [9] 2011/07/01 18:16:33: UDP: Opening socket on 0.0.0.0:60915 [8] 2011/07/01 18:16:33: Could not find a trunk (2 trunks) [9] 2011/07/01 18:16:33: Using outbound proxy sip:192.168.1.8:2782;transport=tls because of flow-label [9] 2011/07/01 18:16:33: Last message repeated 3 times [7] 2011/07/01 18:16:33: SIP Tx tls:192.168.1.8:2782: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.1.8:2782;branch=z9hG4bK-v0svqzc2wfnk;rport=2782 From: "Study" <sip:48@localhost>;tag=xy3mypr1cv To: <sip:819161@localhost;user=phone>;tag=1abb0eb12d Call-ID: 3c641eccef30-an7w0fgu4eg2 CSeq: 1 INVITE Content-Length: 0 [7] 2011/07/01 18:16:33: Set packet length to 20 [6] 2011/07/01 18:16:33: Sending RTP for 3c641eccef30-an7w0fgu4eg2 to 192.168.1.8:54922, codec not set yet [8] 2011/07/01 18:16:33: Incoming call: Request URI sip:819161@localhost;user=phone, To is <sip:819161@localhost;user=phone> [8] 2011/07/01 18:16:33: Call from an user 48 [8] 2011/07/01 18:16:33: To is <sip:819161@localhost;user=phone>, user 0, domain 1 [8] 2011/07/01 18:16:33: From user 48 [8] 2011/07/01 18:16:33: Set the To domain based on From user 48@localhost [8] 2011/07/01 18:16:33: Call state for call object 1: idle [7] 2011/07/01 18:16:33: set_codecs: for 3c641eccef30-an7w0fgu4eg2 codecs "", codec_preference count 6 [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^(999)@.*!sip:\1@\r;user=phone!i against 819161@localhost [6] 2011/07/01 18:16:33: The registration type trunk Patton is not registered. Skipping it... [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^0800([0-9]*)@.*!sip:0800\1@\r;user=phone!i against 819161@localhost [6] 2011/07/01 18:16:33: The registration type trunk Patton is not registered. Skipping it... [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^00([0-9]*)@.*!sip:00\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^07([0-9]*)@.*!sip:07\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^907([0-9]*)@.*!sip:07\1@\r;user=phone!i against 819161@localhost [6] 2011/07/01 18:16:33: The registration type trunk Patton is not registered. Skipping it... [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^900([0-9]*)@.*!sip:00\1@\r;user=phone!i against 819161@localhost [6] 2011/07/01 18:16:33: The registration type trunk Patton is not registered. Skipping it... [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 819161@localhost [6] 2011/07/01 18:16:33: The registration type trunk Patton is not registered. Skipping it... [8] 2011/07/01 18:16:33: call port 0: state code from 0 to 404 [7] 2011/07/01 18:16:33: Set packet length to 20 [7] 2011/07/01 18:16:33: SIP Tx tls:192.168.1.8:2782: SIP/2.0 404 Not Found Via: SIP/2.0/TLS 192.168.1.8:2782;branch=z9hG4bK-v0svqzc2wfnk;rport=2782 From: "Study" <sip:48@localhost>;tag=xy3mypr1cv To: <sip:819161@localhost;user=phone>;tag=1abb0eb12d Call-ID: 3c641eccef30-an7w0fgu4eg2 CSeq: 1 INVITE Contact: <sip:48@192.168.1.13:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.1.4025 Content-Length: 0 [6] 2011/07/01 18:16:33: Received searchRequest, equalityMatch (description=telephoneNumber, value=819161) [7] 2011/07/01 18:16:33: SIP Rx tls:192.168.1.8:2782: ACK sip:819161@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.8:2782;branch=z9hG4bK-v0svqzc2wfnk;rport From: "Study" <sip:48@localhost>;tag=xy3mypr1cv To: <sip:819161@localhost;user=phone>;tag=1abb0eb12d Call-ID: 3c641eccef30-an7w0fgu4eg2 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:48@192.168.1.8:2782;transport=tls;line=mjvwc7ij>;reg-id=1 Proxy-Require: buttons Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
venom Posted July 1, 2011 Report Share Posted July 1, 2011 Thank you for the SnomOne Plus Update Info Venom Quote Link to comment Share on other sites More sharing options...
tommymtl Posted July 4, 2011 Report Share Posted July 4, 2011 A simple question for PBXnSIP user ... 4.2.0.3981(win32) was 9,20 MB (9 649 152 bytes) and new version on PBXnSIP 4.2.1.4025 (Win32) is now 7,22 MB (7 572 480 bytes)... Over 2 mb smaller... what was removed? what are we missing? Are we loosing something or does the update should work seamless? with all complex elements, we are getting nervous ... thanks Quote Link to comment Share on other sites More sharing options...
pbx support Posted July 5, 2011 Author Report Share Posted July 5, 2011 No, the last version was built with some project info(which wasn't needed). The new one does not. So, nothing is removed or missing from the functionality point of view. Quote Link to comment Share on other sites More sharing options...
mattlandis Posted July 5, 2011 Report Share Posted July 5, 2011 even smaller? I already thot snomONE was incredibly small! Way to go! (after a 3day exchange/uc & lync implementation ....it looks amazing.) But in all fairness lync feature set and scaling is very impressive. Quote Link to comment Share on other sites More sharing options...
p800aul Posted July 5, 2011 Report Share Posted July 5, 2011 Hi Any thoughts on my issues above pbx support? thanks Paul Quote Link to comment Share on other sites More sharing options...
pbx support Posted July 5, 2011 Author Report Share Posted July 5, 2011 Oops, missed that totally. The issue is that on the PBX, "Patton" trunk type is set to "SIP Registration", but the trunk is not registered. In .4025, we skip the unregistered trunks for outbound calls. You can do either - make sure that the trunk is registered OR change the trunk type to "SIP Gateway" to avoid the issue. The registration type trunk Patton is not registered. Skipping it... Quote Link to comment Share on other sites More sharing options...
p800aul Posted July 5, 2011 Report Share Posted July 5, 2011 Oops, missed that totally. The issue is that on the PBX, "Patton" trunk type is set to "SIP Registration", but the trunk is not registered. In .4025, we skip the unregistered trunks for outbound calls. You can do either - make sure that the trunk is registered OR change the trunk type to "SIP Gateway" to avoid the issue. The registration type trunk Patton is not registered. Skipping it... Thanks That seems to have fixed it Regards Paul Quote Link to comment Share on other sites More sharing options...
hmcap Posted July 6, 2011 Report Share Posted July 6, 2011 I'll post my question here and also in 'general', but I pretty desperate and don't know where to start. It happens after upgrading to 4.2.1.4025. First I had some problems with the caller-ID and now all my inbound trunks are not working. I've a DID via DIDWW, which is mapped to a voipcheap-trunk with a failover to a voipbuster-trunk. Since two days, it is not possible anymore to call our office. A long silence is heard. I than down=graded to x.x.x.3981 and it looks like the snom one was working again. Just for a few minutes and the office was not reacheable again. How to solve this? I don't know where to start looking, but it has to work. Regards, Harry Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted July 31, 2011 Report Share Posted July 31, 2011 If or when will SnomOne tightly integrate with Snom Phones. By this I mean control the many settings in the phones from a WEB interface on the PBX. We are fine with PNP, and buttons, but being able to set global Snom Settings in the PBX and control on a EXT by EXT basis the many other settings that you commonly will need to adjust. Might this be on the Road Map? Quote Link to comment Share on other sites More sharing options...
marsbewohner Posted August 1, 2011 Report Share Posted August 1, 2011 I second that, hopefully the V5 brings some improvements with it when it comes to the device specific options as well as having configuration templates and groups which can be applied to the devices on the webinterface of SnomOne. Quote Link to comment Share on other sites More sharing options...
Ganesh Posted August 18, 2011 Report Share Posted August 18, 2011 Hello, We are using Cent OS and upgraded the pbxnsip to version 4.2.1.4025. After upgrade while I try to start the service I see the error as "[root@sip3 ~]# service pbxnsip start Starting PBX:/etc/init.d/pbxnsip: line 19: 4471 Segmentation fault $PBX --dir $INSTALLDIR" The start up script file works fine with the older version and not with version 4025. Can you please tell me what changes we need to make in start up file? I have checked the installation directory. It is as per the path in the start up file. Regards Ganesh Quote Link to comment Share on other sites More sharing options...
Ganesh Posted August 18, 2011 Report Share Posted August 18, 2011 Hello, We are using Cent OS and upgraded the pbxnsip to version 4.2.1.4025. After upgrade while I try to start the service I see the error as "[root@sip3 ~]# service pbxnsip start Starting PBX:/etc/init.d/pbxnsip: line 19: 4471 Segmentation fault $PBX --dir $INSTALLDIR" The start up script file works fine with the older version and not with version 4025. Can you please tell me what changes we need to make in start up file? I have checked the installation directory. It is as per the path in the start up file. Regards Ganesh Quote Link to comment Share on other sites More sharing options...
Ganesh Posted August 18, 2011 Report Share Posted August 18, 2011 This is the startup file that we are using for CentOS and this is working with older versions and not working with version 4025. The error displayed while starting the service is "[root@sip3 ~]# service pbxnsip start Starting PBX:/etc/init.d/pbxnsip: line 19: 4471 Segmentation fault $PBX --dir $INSTALLDIR" Let me know the changes needed in the file to run version 4025. #!/bin/bash # # Init file for pbxnsip PBX # # Copyright © 2006 pbxnsip Inc., USA # # chkconfig: 2345 20 80 # description: SIP-based PBX # # processname: pbxctrl # pidfile: /var/run/pbxctrl.pid # source function library . /etc/rc.d/init.d/functions RETVAL=0 # Installation location INSTALLDIR=/usr/local/pbxnsip PBX=$INSTALLDIR/pbxctrl start() { echo -n "Starting PBX:" $PBX --dir $INSTALLDIR echo RETVAL=1 } stop() { echo -n "Stopping PBX:" killproc $PBX -TERM echo RETVAL=1 } case "$1" in start) start ;; stop) stop ;; restart) stop start ;; status) status $PBX RETVAL=$? ;; *) echo $"Usage: $0 {start|stop|restart|status}" RETVAL=1 esac exit $RETVAL Quote Link to comment Share on other sites More sharing options...
pbx support Posted August 18, 2011 Author Report Share Posted August 18, 2011 There are no changes required for the new version. All you have to do is to download the new version from pbxnsip download page and follow http://wiki.snomone.com/index.php?title=Upgrades#General_manual_upgrade_guidelines_for_Linux_based_systems. This one is written with snomONE in mind. But you can substitute pbxctrl instead and have it working. Quote Link to comment Share on other sites More sharing options...
rob_acs Posted August 24, 2011 Report Share Posted August 24, 2011 Is this update compatible with a snom One Hosted license?? We have a hosting license and have been on 4.0 for quite a while and want to upgrade but havent heard anything on the hosted side of what used to be pbxnsip since snom took over.... Quote Link to comment Share on other sites More sharing options...
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