McFone Posted May 9, 2013 Report Share Posted May 9, 2013 Hi, if we set DND on the phone this extension is immediately marked as busy. The DND deactivation status on the phone takes about 120 sec before this update is marked on the phone and also on the other extensions. Why the deactivation status is not immediately on the phones we use snom 720 ? thanks Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 9, 2013 Report Share Posted May 9, 2013 Did you PnP the phone? There are trouble tickets pending with snom in this area. For example, click to dial does not work when the phone is on DND. Other phone vendors work fine with the synchronization of the DND and redirection status; the PBX side seems to be okay. Quote Link to comment Share on other sites More sharing options...
McFone Posted May 9, 2013 Author Report Share Posted May 9, 2013 We use PNP why you mean the PBX seems OK ? This updated for the DND comes from the pbx or not. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 9, 2013 Report Share Posted May 9, 2013 There is a special subscription that the phone needs to send to the PBX to receives updates when the DND state changes. For example, if the user changes the DND state from the web interface, the user can see that also on the phone. It is a little complicated to manually set that up, that's why PnP is the way to go here. Quote Link to comment Share on other sites More sharing options...
McFone Posted May 9, 2013 Author Report Share Posted May 9, 2013 Hi, we test the DND function with a snom 870 the same result the snom 870 have also a PNP config. Activate is direct and deactivate takes a long time to refresh, is this a snomone problem or a snom firmware and the must open direct by snom a ticket. The PBX send this information via email The call between sip:*79@pbx.awo.de;user=phone and sip:41@pbx.awo.de has been disconnected because media session was not established (source=xxxxxxxxxx:3326) The call between sip:*78@pbx.awo.de;user=phone and sip:41@pbx.awo.de has been disconnected because media session was not established (source=xxxxxxxxxx:3326) 2013/5/9 20:58:27 Tx: tls:xxxxxxxxxxx:3326 (307 bytes) SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.178.24:3326;branch=z9hG4bK-x32h21wzvwx4;rport=3326;received=xxxxxxxxxx From: "AWO Konferenzraum" <sip:41@pbx.awo.de>;tag=2ax4aofm16 To: <sip:*78@pbx.awo.de;user=phone>;tag=40e1a2c850 Call-ID: d3f18b51114b-huu01m607kxr CSeq: 1 INVITE Content-Length: 0 2013/5/9 20:58:27 Tx: tls:xxxxxxxxxxx:3326 (894 bytes) SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.178.24:3326;branch=z9hG4bK-x32h21wzvwx4;rport=3326;received=xxxxxxxxxx From: "AWO Konferenzraum" <sip:41@pbx.awo.de>;tag=2ax4aofm16 To: <sip:*78@pbx.awo.de;user=phone>;tag=40e1a2c850 Call-ID: d3f18b51114b-huu01m607kxr CSeq: 1 INVITE Contact: <sip:41@xxxxxxxxx:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/5.0.8 Content-Type: application/sdp Content-Length: 333 v=0 o=- 1975364849 1975364849 IN IP4 xxxxxxxx s=- c=IN IP4 xxxxxxxxxx t=0 0 m=audio 57078 RTP/AVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:zo1r7tsVBKAONiXFzvWwm2S7T31WvD2IC1f8YTq4 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=recvonly 2013/5/9 20:58:27 Rx: tls:xxxxxxxxxx:3326 (433 bytes) ACK sip:41@xxxxxxxxxxxx:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.178.24:3326;branch=z9hG4bK-fd9n70gy2vqv;rport From: "AWO Konferenzraum" <sip:41@pbx.awo.de>;tag=2ax4aofm16 To: <sip:*78@pbx.awo.de;user=phone>;tag=40e1a2c850 Call-ID: d3f18b51114b-huu01m607kxr CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:41@192.168.178.24:3326;transport=tls;line=tjdf0bu5>;reg-id=1 Proxy-Require: buttons-snom870 Content-Length: 0 2013/5/9 20:58:12 Tx: tls:xxxxxxxxxxxxx:3326 (307 bytes) SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.178.24:3326;branch=z9hG4bK-l06jawifprit;rport=3326;received=xxxxxxxxxxxxx From: "AWO Konferenzraum" <sip:41@pbx.awo.de>;tag=yaxlta276t To: <sip:*79@pbx.awo.de;user=phone>;tag=fa7272e543 Call-ID: c4f18b5197a4-xqivq2gebv5x CSeq: 1 INVITE Content-Length: 0 2013/5/9 20:58:12 Tx: tls:xxxxxxxxxxxxx:3326 (894 bytes) SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.178.24:3326;branch=z9hG4bK-l06jawifprit;rport=3326;received=xxxxxxxxxxxxx From: "AWO Konferenzraum" <sip:41@pbx.awo.de>;tag=yaxlta276t To: <sip:*79@pbx.awo.de;user=phone>;tag=fa7272e543 Call-ID: c4f18b5197a4-xqivq2gebv5x CSeq: 1 INVITE Contact: <sip:41@xxxxxxxxxxxxx:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/5.0.8 Content-Type: application/sdp Content-Length: 333 v=0 o=- 1284771858 1284771858 IN IP4 xxxxxxxxxxxxx s=- c=IN IP4 xxxxxxxxxxxxx t=0 0 m=audio 63032 RTP/AVP 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:vya3Cx1y/JkuKznovqkIfCmaSTQo31ZQir61D23B a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=recvonly 2013/5/9 20:58:13 Rx: tls:xxxxxxxxxxxxx:3326 (433 bytes) ACK sip:41@xxxxxxxxxxxxx:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.178.24:3326;branch=z9hG4bK-0n1zfqsdgi5y;rport From: "AWO Konferenzraum" <sip:41@pbx.awo.de>;tag=yaxlta276t To: <sip:*79@pbx.awo.de;user=phone>;tag=fa7272e543 Call-ID: c4f18b5197a4-xqivq2gebv5x CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:41@192.168.178.24:3326;transport=tls;line=tjdf0bu5>;reg-id=1 Proxy-Require: buttons-snom870 Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 9, 2013 Report Share Posted May 9, 2013 The call between sip:*79@pbx.awo.de;user=phone and sip:41@pbx.awo.de has been disconnected because media session was not established (source=xxxxxxxxxx:3326) The call between sip:*78@pbx.awo.de;user=phone and sip:41@pbx.awo.de has been disconnected because media session was not established (source=xxxxxxxxxx:3326) Oh that! Yea there is a problem with the phones that dial the number at teh same time when using the bespoken feature synchronization. Newer firmware versions have that problem fixed, however are not ready for release yet. The calls eventually time out, and then it could be that the status is cahnged some time later. Quote Link to comment Share on other sites More sharing options...
McFone Posted May 9, 2013 Author Report Share Posted May 9, 2013 [/size] Oh that! Yea there is a problem with the phones that dial the number at teh same time when using the bespoken feature synchronization. Newer firmware versions have that problem fixed, however are not ready for release yet. The calls eventually time out, and then it could be that the status is cahnged some time later. all the phones have 8.7.3.19 ! the only thing we can do is to wait for the new firmware is this right :-( thanks Quote Link to comment Share on other sites More sharing options...
hosted Posted May 11, 2013 Report Share Posted May 11, 2013 waiting is agonizing as there are lots of "button" issues. Quote Link to comment Share on other sites More sharing options...
metwest Posted May 14, 2013 Report Share Posted May 14, 2013 I am currently waiting for this dnd issue and a CID issue that shows address book entries instead of e correct CID to be fixed for a large amount of 870s. Also using shared line buttons on 870s is useless as monitoring light hangs in some cases. I played around with the new beta but it seems to bring just as many bugs as it does new features. (Park key monitor light not working being a big one) As much as I love some of the new features Snom are throwing in I really think there should have been a better stable release after 8.7.3.19 before adding so much new stuff. At present my only option at the moment might be downgrading my 870s to 8.4.35 and hoping no other features break in the downgrade... Quote Link to comment Share on other sites More sharing options...
hosted Posted May 14, 2013 Report Share Posted May 14, 2013 agreed! the coolest features with snom don't work. I get complaints all the time to the point i have slowed down our sales efforts. The buttons are critical! and if they dont turn off.. well Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 15, 2013 Report Share Posted May 15, 2013 Well the latest beta firmware had a lot of cool features but lacked the stability. Hopefully the next general firmware release will combine the stability of the good old versions with the latest features. Quote Link to comment Share on other sites More sharing options...
hosted Posted June 5, 2013 Report Share Posted June 5, 2013 stable yet? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted June 6, 2013 Report Share Posted June 6, 2013 I hear good things about the new firmware, but we were not able to try it out yet. Quote Link to comment Share on other sites More sharing options...
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