xy3

Thanks. In 8.4.31 (and prior) we had very erratic LED behavior on associated VISION devices. Can you manage to share a tip or two on how to establish an SSH session? Can you suggest a shareware tool that might help us? What is the working directory in the SoHo where the supplied file should be placed? Can you (please) elaborate/clarify the command referring to a 'symbolic link' noted in the cryptic "instructions" referred to previously?

Desperately need some step-by-step instruction as to how to manually update a snom SoHo unit. The 'admin' panel features only an 'upgrade button', bringing the unit to the current stable release. There are many instances where a manual upgrade to a specific version is needed. Yes, we are aware there is a short, terse, generalized passage at the bottom of this page: http://wiki.snomone.com/index.php?title=Upgrades#General_manual_upgrade_guidelines_for_Linux_based_systems But for the non-Linux techs, or even those unfamiliar with the SoHo build, more detailed instruction would be a great help. PLEASE ADVISE ASAP. Many thanks in advance for your kind attention!

The terse generalized comments at the bottom of that page don't help much. 1). How do we get onto the device to issue the commands? Through the 'advanced' dialog window on the 'admin' page? 2). What is the 'pbx working directory' for the specific SoHo build? 3). How exactly can we 'download' the supplied file to it? 4). This statement needs to be made specific to our case: "Point the symbolic link to new binary using ln -sf <new file> snomONE-ctrl command (where snomONE-ctrl is the default name of the symbolic link)". We're not Linux techs, and need a bit more help (please) than cryptic remarks on that page. Can you *please* tell us in a bit more detail how we can upgrade the SoHo using the file supplied. We are administering the device from Windows PCs. Quickly. Many thanks in advance.

Guys - Thanks for the file. Just need the procedures for manual update of SoHo unit; been all around forum and wiki and cannot locate. Please, give us some step-by-step instructions, ASAP. Appreciate your kind help. - XY3

Can you advise how we should best do this on the snom SoHo unit?

(this is a snom SOHO unit; I presume it's a Linux OS build)

Can you supply a fix? OS is Windows 7. Many thanks for your kind attention. :-)

Excellent. Greatly prefer to keep the 870 at 8.4.33 -- it's working *nicely* with our snom VISION. I am administering the SoHo via WUI from a Win32 machine.

Ah-hah! Interesting! Thanks. However.. appears our SoHo device has only a nice 'update' button on it's 'admin' page to change f/w. Clicking only brings version up to 5016. How can I update SoHo to 5021? Is there a way to force / apply specific f/w update manually??

Thanks. Apols am inexpert at reading these things as you can tell. Here we go: Sent to udp:192.168.0.100:5060 at 2/12/2011 09:17:36:903 (1482 bytes): INVITE sip:301@192.168.0.100;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.5:3072;branch=z9hG4bK-e1k6148uv5c5;rport From: "200" <sip:200@192.168.0.100>;tag=3xqetluuuo To: <sip:301@192.168.0.100;user=phone> Call-ID: 5971263cf2a0-eefnjtdxmyrv CSeq: 2 INVITE Max-Forwards: 70 Contact: <sip:200@192.168.0.5:3072;line=eigyyxaf>;reg-id=1 X-Serialnumber: 000413410439 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom870/8.4.33 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 Call-Info: <sip:192.168.0.100>;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Authorization: Digest username="200",realm="192.168.0.100",nonce="c71dd4daf217fb7454673c1e86d04df0",uri="sip:301@192.168.0.100;user=phone",response="bf751966cb57fb733067dcf7a0604116",algorithm=MD5 Content-Type: application/sdp Content-Length: 473 v=0 o=root 2104588788 2104588788 IN IP4 192.168.0.5 s=call c=IN IP4 192.168.0.5 t=0 0 m=audio 53200 RTP/AVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:3pDXMAPzJanqLHUBQY6/KIXCr28XXoirqIwry/JZ a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 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=sendrecv Received from udp:192.168.0.100:5060 at 2/12/2011 09:17:36:920 (274 bytes): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.5:3072;branch=z9hG4bK-e1k6148uv5c5;rport=3072 From: "200" <sip:200@192.168.0.100>;tag=3xqetluuuo To: <sip:301@192.168.0.100;user=phone>;tag=ebc7afaccb Call-ID: 5971263cf2a0-eefnjtdxmyrv CSeq: 2 INVITE Content-Length: 0

Reset settings and rebooted set; amazingly, we still get same result. "Sorry, but you are not allowed to call that number". Here's a SIP trace: -------- Received from udp:192.168.0.100:5060 at 2/12/2011 09:17:36:960 (881 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.0.5:3072;branch=z9hG4bK-e1k6148uv5c5;rport=3072 From: "200" <sip:200@192.168.0.100>;tag=3xqetluuuo To: <sip:301@192.168.0.100;user=phone>;tag=ebc7afaccb Call-ID: 5971263cf2a0-eefnjtdxmyrv CSeq: 2 INVITE Contact: <sip:200@192.168.0.100:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.3.0.5016 Content-Type: application/sdp Content-Length: 353 v=0 o=- 1241655547 1241655547 IN IP4 192.168.0.100 s=- c=IN IP4 192.168.0.100 t=0 0 m=audio 62534 RTP/AVP 0 8 9 99 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv Sent to udp:192.168.0.100:5060 at 2/12/2011 09:17:36:977 (364 bytes): ACK sip:200@192.168.0.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.5:3072;branch=z9hG4bK-utzk87an19lt;rport From: "200" <sip:200@192.168.0.100>;tag=3xqetluuuo To: <sip:301@192.168.0.100;user=phone>;tag=ebc7afaccb Call-ID: 5971263cf2a0-eefnjtdxmyrv CSeq: 2 ACK Max-Forwards: 70 Contact: <sip:200@192.168.0.5:3072;line=eigyyxaf>;reg-id=1 Content-Length: 0 Sent to udp:192.168.0.100:5060 at 2/12/2011 09:17:38:211 (528 bytes): BYE sip:200@192.168.0.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.5:3072;branch=z9hG4bK-1zik3qsdxd73;rport From: "200" <sip:200@192.168.0.100>;tag=3xqetluuuo To: <sip:301@192.168.0.100;user=phone>;tag=ebc7afaccb Call-ID: 5971263cf2a0-eefnjtdxmyrv CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:200@192.168.0.5:3072;line=eigyyxaf>;reg-id=1 User-Agent: snom870/8.4.33 RTP-RxStat: Total_Rx_Pkts=49,Rx_Pkts=45,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=51,Tx_Pkts=46,Remote_Tx_Pkts=2 Content-Length: 0 Received from udp:192.168.0.100:5060 at 2/12/2011 09:17:38:220 (344 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.0.5:3072;branch=z9hG4bK-1zik3qsdxd73;rport=3072 From: "200" <sip:200@192.168.0.100>;tag=3xqetluuuo To: <sip:301@192.168.0.100;user=phone>;tag=ebc7afaccb Call-ID: 5971263cf2a0-eefnjtdxmyrv CSeq: 3 BYE Contact: <sip:200@192.168.0.100:5060> User-Agent: snom-PBX/2011-4.3.0.5016 Content-Length: 0

Just setting up a new snomONE SoHo device (running 2011-4.3.0.5016) and suspect we've done something dumb.. but can't see what. One station, an 870 running 8.4.33 (for improved BLF performance on associated VISION device), gets an "We're sorry but you are not allowed to place this call" recording dialing the other extensions. This happens even when calling itself (!). Other extensions can call the 870 (identity = ext 200) without any problem. Interestingly, when we look at this "anti-social" 870's call log, it shows the dialed extension number was treated as an 'intercom call': [5] 2011/12/02 06:28:11: No permission to intercom to "DECT 300" <sip:300@pbx.company.com> [5] 2011/12/02 06:28:14: BYE Response: Terminate 1072263cf633-0ftov9jn1a1t [5] 2011/12/02 06:28:18: No permission to intercom to "DECT 301" <sip:301@pbx.company.com> [5] 2011/12/02 06:28:21: BYE Response: Terminate 1772263cb9f3-0e6wreqht90e [5] 2011/12/02 06:28:23: No permission to intercom to "Desk phone" <sip:200@pbx.company.com> [5] 2011/12/02 06:28:26: BYE Response: Terminate 1d72263c109b-e2q9dr7vs44p We've tried changing intercom policy on sets to no avail (although they did then ring as voice announce/intercom when called from other extensions). Calls from 200 to other extensions still fail. Suggestions..?