eyeless

Yes, but the last: "Then you need to put your phone number into the setting Trunk ANI of the PBX." might not be needed as I received the exact same result with it filled in or not. Thanks!!

I understand. I will try and work this out with the trunk provider, and it should in their interest to be able to help out in such cases, so … . Will post the solution here if I get one.

I see. Not very helpful though as I only want the SnomOne to send out headers just like it always has done up to version 4.5.x … when there has been no problem. Apparently the trunk providers are using Asterix servers themselves and do not understand SnomOne … . So was the reason for changing this to get people to buy the SnomOne 5 instead? One wonders … .

Well, the provider did not understand what was wrong or how to change to a custom header. I tried all the different alternatives in the Number/Call Identification section now (well, almost all, at least all in the drop-down menu). Here is the log if it makes anything more clear (I only have added to the Trunk ANI field the phone number after upgrade - this field was blank before, but adding the number there did not change anything) - the hidden outgoing number is 031109430 (account name is both 30 & 031109430): [5] 2013/02/08 16:11:29: SIP Rx tls:10.0.3.234:3448: INVITE sip:0317018939@10.0.3.10;user=phone SIP/2.0 Via: SIP/2.0/TLS 10.0.3.234:3448;branch=z9hG4bK-hcuffvg1f2s0;rport From: "Eva Levin" <sip:30@10.0.3.10>;tag=s1609stbfc To: <sip:0317018939@10.0.3.10;user=phone> Call-ID: 3c31d1beece8-lhkffaqe75rn CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:30@10.0.3.234:3448;transport=tls;line=omt9jyts>;reg-id=1 X-Serialnumber: 00041331E1F5 P-Key-Flags: keys="3" User-Agent: snom320/8.4.18 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: 518 v=0 o=root 484433817 484433817 IN IP4 10.0.3.234 s=call c=IN IP4 10.0.3.234 t=0 0 m=audio 64710 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:E2SAYAZBeodnsVnNhmM1XR7Y9uOpVl68nShKlNlx 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] 2013/02/08 16:11:29: Packet authenticated by transport layer [9] 2013/02/08 16:11:29: Using outbound proxy sip:10.0.3.234:3448;transport=tls because of flow-label [9] 2013/02/08 16:11:30: Last message repeated 3 times [5] 2013/02/08 16:11:30: SIP Tx tls:10.0.3.234:3448: SIP/2.0 100 Trying Via: SIP/2.0/TLS 10.0.3.234:3448;branch=z9hG4bK-hcuffvg1f2s0;rport=3448 From: "Eva Levin" <sip:30@10.0.3.10>;tag=s1609stbfc To: <sip:0317018939@10.0.3.10;user=phone>;tag=8d078c94b0 Call-ID: 3c31d1beece8-lhkffaqe75rn CSeq: 1 INVITE Content-Length: 0 [8] 2013/02/08 16:11:30: Incoming call: Request URI sip:0317018939@10.0.3.10;user=phone, To is <sip:0317018939@10.0.3.10;user=phone> [8] 2013/02/08 16:11:30: Set the To domain based on From user 30@10.0.3.10 [9] 2013/02/08 16:11:30: Resolve 10167: url sip:sip.voicetech.se [9] 2013/02/08 16:11:30: Resolve 10167: naptr sip.voicetech.se [9] 2013/02/08 16:11:30: Resolve 10167: srv tls _sips._tcp.sip.voicetech.se [9] 2013/02/08 16:11:30: Resolve 10167: srv tcp _sip._tcp.sip.voicetech.se [9] 2013/02/08 16:11:30: Resolve 10167: srv udp _sip._udp.sip.voicetech.se [9] 2013/02/08 16:11:30: Resolve 10167: aaaa udp sip.voicetech.se 5060 [9] 2013/02/08 16:11:30: Resolve 10167: a udp sip.voicetech.se 5060 [9] 2013/02/08 16:11:30: Resolve 10167: udp 212.3.0.180 5060 [5] 2013/02/08 16:11:30: SIP Tx udp:212.3.0.180:5060: INVITE sip:0317018939@sip.voicetech.se;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.3.10:5060;branch=z9hG4bK-d480b58ea53b1be72eb8bf65a62c4c8a;rport From: "Eva Levin" <sip:031109430@10.0.3.10;user=phone>;tag=879293534 To: <sip:0317018939@10.0.3.10;user=phone> Call-ID: 173c1240@pbx CSeq: 25139 INVITE Max-Forwards: 70 Contact: <sip:031109430@10.0.3.10:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.1.1107 Zeta Perseids P-Asserted-Identity: "Eva Levin" <sip:031109430@sip.voicetech.se> Privacy: id Content-Type: application/sdp Content-Length: 378 v=0 o=- 1818723674 1818723674 IN IP4 10.0.3.10 s=- c=IN IP4 10.0.3.10 t=0 0 m=audio 50058 RTP/AVP 0 8 9 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no 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 [5] 2013/02/08 16:11:30: set codec: codec pcmu/8000 is set to call-leg 139 [5] 2013/02/08 16:11:30: SIP Tx tls:10.0.3.234:3448: SIP/2.0 183 Session Progress Via: SIP/2.0/TLS 10.0.3.234:3448;branch=z9hG4bK-hcuffvg1f2s0;rport=3448 From: "Eva Levin" <sip:30@10.0.3.10>;tag=s1609stbfc To: <sip:0317018939@10.0.3.10;user=phone>;tag=8d078c94b0 Call-ID: 3c31d1beece8-lhkffaqe75rn CSeq: 1 INVITE Contact: <sip:30@10.0.3.10:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.1.1107 Zeta Perseids Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 474 v=0 o=- 1414671445 1414671445 IN IP4 10.0.3.10 s=- c=IN IP4 10.0.3.10 t=0 0 m=audio 58816 RTP/AVP 0 8 9 18 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ssWlPg8UG7cr5hZuHpunzJ9JfCBJFmfdgZAPsbPa a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:2 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 [5] 2013/02/08 16:11:30: SIP Rx udp:212.3.0.180:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.3.10:5060;branch=z9hG4bK-d480b58ea53b1be72eb8bf65a62c4c8a;rport=47104 From: "Eva Levin" <sip:031109430@10.0.3.10:5060;user=phone>;tag=879293534 To: <sip:0317018939@10.0.3.10:5060;user=phone>;tag=7000166881dd2f9a2a8458b004d02617.ff08 Call-ID: 173c1240@pbx CSeq: 25139 INVITE Proxy-Authenticate: Digest realm="sips.teleman.com", nonce="511519265cc9bda9b08f820a70c78daa8fc448af" Server: OpenSer (1.1.0-tls (x86_64/linux)) Content-Length: 0 Warning: 392 212.3.0.180:5060 "Noisy feedback tells: pid=29552 req_src_ip=81.216.208.134 req_src_port=47104 in_uri=sip:0317018939@sip.voicetech.se;user=phone out_uri=sip:0317018939@sip.voicetech.se;user=phone via_cnt==1" [8] 2013/02/08 16:11:30: Answer challenge with username 031109430 [9] 2013/02/08 16:11:30: Resolve 10169: udp 212.3.0.180 5060 udp:1 [5] 2013/02/08 16:11:30: SIP Tx udp:212.3.0.180:5060: ACK sip:0317018939@sip.voicetech.se;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.3.10:5060;branch=z9hG4bK-d480b58ea53b1be72eb8bf65a62c4c8a;rport From: "Eva Levin" <sip:031109430@10.0.3.10:5060;user=phone>;tag=879293534 To: <sip:0317018939@10.0.3.10:5060;user=phone>;tag=7000166881dd2f9a2a8458b004d02617.ff08 Call-ID: 173c1240@pbx CSeq: 25139 ACK Max-Forwards: 70 Content-Length: 0 [9] 2013/02/08 16:11:30: Resolve 10170: udp 212.3.0.180 5060 udp:1 [5] 2013/02/08 16:11:30: SIP Tx udp:212.3.0.180:5060: INVITE sip:0317018939@sip.voicetech.se;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.3.10:5060;branch=z9hG4bK-60034bf93976763369c60cb96a63233e;rport From: "Eva Levin" <sip:031109430@10.0.3.10;user=phone>;tag=879293534 To: <sip:0317018939@10.0.3.10;user=phone> Call-ID: 173c1240@pbx CSeq: 25140 INVITE Max-Forwards: 70 Contact: <sip:031109430@10.0.3.10:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.1.1107 Zeta Perseids P-Asserted-Identity: "Eva Levin" <sip:031109430@sip.voicetech.se> Privacy: id Proxy-Authorization: Digest realm="sips.teleman.com",nonce="511519265cc9bda9b08f820a70c78daa8fc448af",response="b49cb9031c05e2a49124edc6790d170a",username="031109430",uri="sip:0317018939@sip.voicetech.se;user=phone",algorithm=MD5 Content-Type: application/sdp Content-Length: 378 v=0 o=- 1818723674 1818723674 IN IP4 10.0.3.10 s=- c=IN IP4 10.0.3.10 t=0 0 m=audio 50058 RTP/AVP 0 8 9 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no 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 [9] 2013/02/08 16:11:30: Message repetition, packet dropped [5] 2013/02/08 16:11:30: SIP Rx udp:212.3.0.180:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.0.3.10:5060;branch=z9hG4bK-60034bf93976763369c60cb96a63233e;rport=47104 From: "Eva Levin" <sip:031109430@10.0.3.10:5060;user=phone>;tag=879293534 To: <sip:0317018939@10.0.3.10:5060;user=phone> Call-ID: 173c1240@pbx CSeq: 25140 INVITE Server: OpenSer (1.1.0-tls (x86_64/linux)) Content-Length: 0 Warning: 392 212.3.0.180:5060 "Noisy feedback tells: pid=29553 req_src_ip=81.216.208.134 req_src_port=47104 in_uri=sip:0317018939@sip.voicetech.se;user=phone out_uri=sip:0317018939@sip4.teleman.com;user=phone via_cnt==1" [9] 2013/02/08 16:11:30: Message repetition, packet dropped [5] 2013/02/08 16:11:30: SIP Rx tls:10.0.3.234:3448: PRACK sip:30@10.0.3.10:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.3.234:3448;branch=z9hG4bK-oy0pueeg4a5m;rport From: "Eva Levin" <sip:30@10.0.3.10>;tag=s1609stbfc To: <sip:0317018939@10.0.3.10;user=phone>;tag=8d078c94b0 Call-ID: 3c31d1beece8-lhkffaqe75rn CSeq: 2 PRACK Max-Forwards: 70 Contact: <sip:30@10.0.3.234:3448;transport=tls;line=omt9jyts>;reg-id=1 RAck: 1 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Proxy-Require: buttons Content-Length: 0 [8] 2013/02/08 16:11:30: Packet authenticated by transport layer [5] 2013/02/08 16:11:30: SIP Tx tls:10.0.3.234:3448: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.0.3.234:3448;branch=z9hG4bK-oy0pueeg4a5m;rport=3448 From: "Eva Levin" <sip:30@10.0.3.10>;tag=s1609stbfc To: <sip:0317018939@10.0.3.10;user=phone>;tag=8d078c94b0 Call-ID: 3c31d1beece8-lhkffaqe75rn CSeq: 2 PRACK Contact: <sip:30@10.0.3.10:5061;transport=tls> User-Agent: snomONE/4.5.1.1107 Zeta Perseids Content-Length: 0 [5] 2013/02/08 16:11:30: SIP Rx udp:212.3.0.180:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.3.10:5060;branch=z9hG4bK-60034bf93976763369c60cb96a63233e;rport=47104 Record-Route: <sip:212.3.0.180;lr=on;ftag=879293534> Contact: <sip:0317018939@212.3.0.165:5060;transport=udp> To: <sip:0317018939@10.0.3.10:5060;user=phone>;tag=97be3903 From: "Eva Levin"<sip:031109430@10.0.3.10:5060;user=phone>;tag=879293534 Call-ID: 173c1240@pbx CSeq: 25140 INVITE Content-Type: application/sdp Content-Length: 367 v=0 o=- 19337546 0 IN IP4 88.131.198.35 s=Cisco SDP 0 c=IN IP4 62.80.216.14 t=0 0 m=audio 43684 RTP/AVP 8 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 192-194,200-202 a=X-cap: 2 image udptl t38 [5] 2013/02/08 16:11:30: set codec: codec pcma/8000 is set to call-leg 140

Yes, I saw that field, but was not sure if it would be meaningful to change it … will see what I can do to get it to work …. .

Hi, We upgraded to the latest version of SnomOne 4: Zeta Perseids (4.5.1.1107) (Mac OS X) Apart from having to upgrade the Snom M9 telephones in order to get two-way voice, we only have one problem left and that is that all outgoing calls are now displayed as hidden to the recipient and I cannot simply find a way to make the recipient see what number the calls are coming from any longer. Anyone who has a hint of what might have changed from version 4.2-4.3 to the 4.5 version that could affect this? All the best, Jerry

But hopefully it will get better ... . Will try and analyze what is causing this ... .

Sounds like something to follow up on (yes, now that you point it out it is strange with the two IPs - will look into that). However, soon after I somewhat frustrated posted this, I got learn about a problem at the telephony station affecting our DSL connection (strange organisational arrangements at the location), which may be the reason why this has started to happen recently (as far as I can make out). I get back here if I find out something more of relevance as the problems might have different parts ... .

Hi, Tried to post in the Snom phones forum section, but could not find out any way in which to do so as I could not login there no matter what ... (but not sure this problem is about the Snom phones after all). Since some weeks back (before we upgraded to the latest version of the SnomOne server) and continuing now after upgrade, phone calls are dropped every now and then (seems like 1 in 5 calls during the day) on all types of Snom phones. It is not exactly acceptable ... . This has only started to happen recently (as far as I can make out). Since the latest version of SnomOne started to make e-mail functionality working, I today also received a message when a call was dropped, but when I later today got a call from one of the users which also got dropped (I could not hear her, but she could hear me - voice out got cut), I received no message about this. So we have all sorts of problems going on here every hour. Here's the full log from the e-mail reporting the dropped call - (what phone and firmware that is used seems to be of no relevance as some are updated and some are not and all have problems now). The call between sip:031109430@opensips.teleman.com;user=phone and sip:0708442407@sip.teleman.com;user=phone has been disconnected because of media timeout (120 seconds), 390/7393 packets have been received/sent 2011/6/20 08:30:06 Rx: udp:213.131.156.66:0 (1055 bytes) INVITE sip:031109430@10.0.3.10:5060;transport=udp;line=c81e728d SIP/2.0 Record-Route: <sip:213.131.156.66;lr=on;ftag=252a9b6a> Via: SIP/2.0/UDP 213.131.156.66;branch=z9hG4bKd1eb.5655e011.0 Via: SIP/2.0/UDP 212.3.0.165:5060;branch=z9hG4bK-d8754z-acf0121eefbbdd47-1---d8754z-;rport=5060 Max-Forwards: 69 Contact: <sip:0708442407@212.3.0.165:5060;transport=udp> To: "031109430"<sip:031109430@opensips.teleman.com;user=phone> From: "0708442407"<sip:0708442407@sip.teleman.com;user=phone>;tag=252a9b6a Call-ID: Y2YyOGYwYWQwYjcxNmZjODE4MmI5MzA1ZGM0OTY5MjU. CSeq: 1 INVITE Allow: INVITE, ACK, BYE, CANCEL Content-Type: application/sdp User-Agent: LEICA-1.8.31.4 X-Ecan: On Content-Length: 352 v=0 o=- 691105141 0 IN IP4 213.50.90.4 s=- c=IN IP4 88.131.158.234 t=0 0 m=audio 40312 RTP/AVP 8 0 18 101 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar:T38FaxUdpEC:t38UDPRedundancy a=cpar:T38FaxVersion:0 a=cpar:T38MaxBitRate:14400 a=sendrecv 2011/6/20 08:30:06 Tx: udp:213.131.156.66:5060 (485 bytes) SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.131.156.66;branch=z9hG4bKd1eb.5655e011.0 Via: SIP/2.0/UDP 212.3.0.165:5060;branch=z9hG4bK-d8754z-acf0121eefbbdd47-1---d8754z-;rport=5060 Record-Route: <sip:213.131.156.66;lr=on;ftag=252a9b6a> From: "0708442407" <sip:0708442407@sip.teleman.com;user=phone>;tag=252a9b6a To: "031109430" <sip:031109430@opensips.teleman.com;user=phone>;tag=5576c1abba Call-ID: Y2YyOGYwYWQwYjcxNmZjODE4MmI5MzA1ZGM0OTY5MjU. CSeq: 1 INVITE Content-Length: 0 2011/6/20 08:30:06 Tx: udp:213.131.156.66:5060 (1071 bytes) SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 213.131.156.66;branch=z9hG4bKd1eb.5655e011.0 Via: SIP/2.0/UDP 212.3.0.165:5060;branch=z9hG4bK-d8754z-acf0121eefbbdd47-1---d8754z-;rport=5060 Record-Route: <sip:213.131.156.66;lr=on;ftag=252a9b6a> From: "0708442407" <sip:0708442407@sip.teleman.com;user=phone>;tag=252a9b6a To: "031109430" <sip:031109430@opensips.teleman.com;user=phone>;tag=5576c1abba Call-ID: Y2YyOGYwYWQwYjcxNmZjODE4MmI5MzA1ZGM0OTY5MjU. CSeq: 1 INVITE Contact: <sip:031109430@10.0.3.10:5060;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: 302 v=0 o=- 653521262 653521262 IN IP4 10.0.3.10 s=- c=IN IP4 10.0.3.10 t=0 0 m=audio 57840 RTP/AVP 0 8 18 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv 2011/6/20 08:30:26 Tx: udp:213.131.156.66:5060 (1057 bytes) SIP/2.0 200 Ok Via: SIP/2.0/UDP 213.131.156.66;branch=z9hG4bKd1eb.5655e011.0 Via: SIP/2.0/UDP 212.3.0.165:5060;branch=z9hG4bK-d8754z-acf0121eefbbdd47-1---d8754z-;rport=5060 Record-Route: <sip:213.131.156.66;lr=on;ftag=252a9b6a> From: "0708442407" <sip:0708442407@sip.teleman.com;user=phone>;tag=252a9b6a To: "031109430" <sip:031109430@opensips.teleman.com;user=phone>;tag=5576c1abba Call-ID: Y2YyOGYwYWQwYjcxNmZjODE4MmI5MzA1ZGM0OTY5MjU. CSeq: 1 INVITE Contact: <sip:031109430@10.0.3.10:5060;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: 302 v=0 o=- 653521262 653521262 IN IP4 10.0.3.10 s=- c=IN IP4 10.0.3.10 t=0 0 m=audio 57840 RTP/AVP 0 8 18 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv 2011/6/20 08:30:26 Rx: udp:213.131.156.66:0 (613 bytes) ACK sip:031109430@10.0.3.10:5060;transport=udp SIP/2.0 Record-Route: <sip:213.131.156.66;lr=on;ftag=252a9b6a> Via: SIP/2.0/UDP 213.131.156.66;branch=z9hG4bKd1eb.5655e011.2 Via: SIP/2.0/UDP 212.3.0.165:5060;branch=z9hG4bK-d8754z-23e25602506f0c2b-1---d8754z-;rport=5060 Max-Forwards: 69 Contact: <sip:0708442407@212.3.0.165:5060;transport=udp> To: "031109430"<sip:031109430@opensips.teleman.com;user=phone>;tag=5576c1abba From: "0708442407"<sip:0708442407@sip.teleman.com;user=phone>;tag=252a9b6a Call-ID: Y2YyOGYwYWQwYjcxNmZjODE4MmI5MzA1ZGM0OTY5MjU. CSeq: 1 ACK Content-Length: 0 P-hint: rr-enforced 2011/6/20 08:30:26 Rx: udp:213.131.156.66:0 (1050 bytes) INVITE sip:031109430@10.0.3.10:5060;transport=udp SIP/2.0 Record-Route: <sip:213.131.156.66;lr=on;ftag=252a9b6a> Via: SIP/2.0/UDP 213.131.156.66;branch=z9hG4bKa1eb.9d1e5d45.0 Via: SIP/2.0/UDP 212.3.0.165:5060;branch=z9hG4bK-d8754z-4e5cd45fb2852121-1---d8754z-;rport=5060 Max-Forwards: 69 Contact: <sip:0708442407@212.3.0.165:5060;transport=udp> To: "031109430"<sip:031109430@opensips.teleman.com;user=phone>;tag=5576c1abba From: "0708442407"<sip:0708442407@sip.teleman.com;user=phone>;tag=252a9b6a Call-ID: Y2YyOGYwYWQwYjcxNmZjODE4MmI5MzA1ZGM0OTY5MjU. CSeq: 2 INVITE Allow: INVITE, ACK, BYE, CANCEL Content-Type: application/sdp User-Agent: LEICA-1.8.31.4 X-Ecan: On Content-Length: 325 P-hint: rr-enforced v=0 o=- 691105141 1 IN IP4 213.50.90.4 s=- c=IN IP4 88.131.158.234 t=0 0 m=audio 40312 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 0 a=cdsc: 2 image udptl t38 a=cpar:T38FaxUdpEC:t38UDPRedundancy a=cpar:T38FaxVersion:0 a=cpar:T38MaxBitRate:14400 a=sendrecv 2011/6/20 08:30:26 Tx: udp:213.131.156.66:5060 (986 bytes) SIP/2.0 200 Ok Via: SIP/2.0/UDP 213.131.156.66;branch=z9hG4bKa1eb.9d1e5d45.0 Via: SIP/2.0/UDP 212.3.0.165:5060;branch=z9hG4bK-d8754z-4e5cd45fb2852121-1---d8754z-;rport=5060 Record-Route: <sip:213.131.156.66;lr=on;ftag=252a9b6a> From: "0708442407" <sip:0708442407@sip.teleman.com;user=phone>;tag=252a9b6a To: "031109430" <sip:031109430@opensips.teleman.com;user=phone>;tag=5576c1abba Call-ID: Y2YyOGYwYWQwYjcxNmZjODE4MmI5MzA1ZGM0OTY5MjU. CSeq: 2 INVITE Contact: <sip:031109430@10.0.3.10:5060;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: 231 v=0 o=- 653521262 653521262 IN IP4 10.0.3.10 s=- c=IN IP4 10.0.3.10 t=0 0 m=audio 57840 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv 2011/6/20 08:30:26 Rx: udp:213.131.156.66:0 (613 bytes) ACK sip:031109430@10.0.3.10:5060;transport=udp SIP/2.0 Record-Route: <sip:213.131.156.66;lr=on;ftag=252a9b6a> Via: SIP/2.0/UDP 213.131.156.66;branch=z9hG4bKa1eb.9d1e5d45.2 Via: SIP/2.0/UDP 212.3.0.165:5060;branch=z9hG4bK-d8754z-ee1c8d3d713a886d-1---d8754z-;rport=5060 Max-Forwards: 69 Contact: <sip:0708442407@212.3.0.165:5060;transport=udp> To: "031109430"<sip:031109430@opensips.teleman.com;user=phone>;tag=5576c1abba From: "0708442407"<sip:0708442407@sip.teleman.com;user=phone>;tag=252a9b6a Call-ID: Y2YyOGYwYWQwYjcxNmZjODE4MmI5MzA1ZGM0OTY5MjU. CSeq: 2 ACK Content-Length: 0 P-hint: rr-enforced

Think I did not give a good answer. First, I actually am totally clueless about what Provisioning is and how it works and if I use it or not. I think it pertains to how devices can get some some settings from somewhere else, but do not understand how this is supposed to work and do not know where to activate or deactivate it in case that should be relevant. Maybe this could explain why I have some problems at times with registering the mobile devices, but not sure. Maybe it could also change settings in inconsistent ways, but that seems unlikely as has not happened before. (Only account info and server address and phone specific settings are made on the Snom 320 and base stations, all general settings are made on the server. Only ringtones settings have been made directly on the mobile devices. Sometimes language settings has to be made at various places.) The m3 was somewhat easy to set up and get registered in the first place and has worked fine for ca. 2 months and before that for two years at another customer with firmware 1.0 (it was not possible to upgrade and I cannot figure out how to do it - the manual surely does not help anyway). Before I changed the group setting on the phone, the call was registered in the log on the base station, but was not visible on the phone itself. I can find no place to on either the server or the base station to set this setting ... (the base station for the m3 almost have no settings at all anyway (unlike for the m9)). Well, the outgoing voice is heard on the other side whether or not the speaker phone mode is activated, but in order to hear something on my side I have to press the speaker phone button. And, yes, to hear the other side when making an outgoing call one also has to press the speaker button. It is just strange that this happened precisely after I upgraded the server ... .

I think it was provisoned ... (said something about it in the log - sorry for not being better on this, but only deals with these phones now and then). I am not 100% sure I was not inadvertently changing something when I played around with the settings both on the phone, on the web interface and then maybe something happened when resetting the base station and re-registering etc.) - anyway found that setting. I suspect the problem really came down to a damaged internal speaker, since I have to press the speaker button for every phone call now in order to hear anything, but given that I put on the external speaker, the phone now works otherwise well. They still have problem with calls that are cut on some phones, calls that are not forwarded and instead directed to the Voice mail, bad quality of voice, bad operating distance for the esp. the m9 phones, calls where one side stops hearing, etc. Many of these problems could likely be handled if the phones were in the hands of computer savvy people, but they are not and thus complaints are common and difficult to get a good solution for. The stationary phones (Snom 320 works much more reliably). There might have been some new network problems recently as I also experienced a strange "locking" when remote controlling the server, but could see it ping an external server quite normally while my session was "held up" - will need to know first if the SnomOne update helped or not. For some reason things always works better when I am present ... ;-).

Should also note that the e-mail messaging now magically started to work again after applying the latest upgrade to the SnomOne.

Well, somehow it had unregistered itself from the call group in that setting on the phone. Still I have to press the speaker phone on each call in order to receive any incoming voice - can't find a solution to that, but maybe the internal speaker has died or something ... .

The m3 registers incoming calls, but does not ring or show that someone is trying to call it ... maybe that what one has to live with in order to get incoming voice ... hmm.

I think the problem was simply that they had pushed the speaker phone button on the side, which had muted all incoming sound ... why and how it can do so I do not know, but apparently this is how it works here ... . However, now that incoming voice can be heard on the handset, it can no longer receive incoming calls ... (Will try and restart the SnomOne server, because that is about the only thing I have not tried.) Managing to register a handset (either m3 & m9) always involves magic and rites ... but usually works after some hours ... .