Jump to content

Snom m9


eyeless
 Share

Recommended Posts

Hi,

 

We have problem with (as it seems) only m9 handsets picking up incoming calls without anyone doing anything. Thus the user can only accept incoming calls by hearing a someone shouting "is anyone there" on the other end. A restart of the base station helps, but we cannot do that several times a day.

 

Here is the log on the phone side:

 

2011/03/30 16:09:38 [sIP-Call:5]: SIP Tx tls:10.0.3.10:5061:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.0.3.10:5061;branch=z9hG4bK-96ce8b1a3be25e5b4a1162c64e04f6ec;rport=5061

From: "Anonymous" <sip:anonymous@10.0.3.10;user=phone>;tag=414020219

To: "Noaks Ark" <sip:40@10.0.3.10>;tag=bc65ma

Call-ID: a418de07@pbx

CSeq: 30181 INVITE

Contact: <sip:40@10.0.3.231:4228;transport=tls;line=mfhmb9>

Supported: 100rel, replaces, norefersub

User-Agent: snom-m9/9.2.42-b

Content-Type: application/sdp

Content-Length: 305

 

v=0

o=root 1282523677 1282523678 IN IP4 10.0.3.231

s=-

c=IN IP4 10.0.3.231

t=0 0

m=audio 54836 RTP/AVP 0 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:BFLvj6eg7F0KU1MjKryxLq7Vutxvf+QrdVsD4Kxp|2^31

a=sendrecv

2011/03/30 16:09:38 [DECT:5]: Send Call Status: encrypted

2011/03/30 16:09:38 [DECT:5]: hanset 3 picked up the call

2011/03/30 16:09:37 [sIP-Call:5]: SIP Tx tls:10.0.3.10:5061:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.0.3.10:5061;branch=z9hG4bK-e84a2d62b187fe9e1bf655a18a22e769;rport=5061

From: "Anonymous" <sip:anonymous@10.0.3.10;user=phone>;tag=414020219

To: "Noaks Ark" <sip:40@10.0.3.10>;tag=bc65ma

Call-ID: a418de07@pbx

CSeq: 30182 PRACK

Contact: <sip:40@10.0.3.231:4228;transport=tls;line=mfhmb9>

Supported: 100rel, replaces, norefersub

Content-Length: 0

 

 

And here is the SnomOne server log:

 

[7] 2011/03/30 16:09:35: update_codecs for ZjQ3ZWVkMTQ1YmFmMzgxZjk1NmJjMTgwMjEzZjE5Y2Y.: codec_preference size 7, available codecs size 4

[8] 2011/03/30 16:09:35: Play audio_se/ringback.wav

[6] 2011/03/30 16:09:35: Codec pcmu/8000 is chosen for call id ZjQ3ZWVkMTQ1YmFmMzgxZjk1NmJjMTgwMjEzZjE5Y2Y.

[7] 2011/03/30 16:09:35: Call a418de07@pbx: Clear last request

[7] 2011/03/30 16:09:35: Call c4a9c5be@pbx: Clear last request

[8] 2011/03/30 16:09:36: Packet authenticated by transport layer

[7] 2011/03/30 16:09:37: Call a418de07@pbx: Clear last INVITE

[6] 2011/03/30 16:09:37: Codec pcmu/8000 is chosen for call id a418de07@pbx

[6] 2011/03/30 16:09:37: Sending RTP for a418de07@pbx to 10.0.3.231:54836, codec pcmu/8000

[7] 2011/03/30 16:09:37: Determine pass-through mode after receiving response

[8] 2011/03/30 16:09:37: Call state for call object 556: connected

[7] 2011/03/30 16:09:37: ZjQ3ZWVkMTQ1YmFmMzgxZjk1NmJjMTgwMjEzZjE5Y2Y.: RTP pass-through mode

[7] 2011/03/30 16:09:37: a418de07@pbx: RTP pass-through mode

[7] 2011/03/30 16:09:37: Call c4a9c5be@pbx: Clear last request

[7] 2011/03/30 16:09:37: Call c4a9c5be@pbx: Clear last INVITE

[5] 2011/03/30 16:09:37: INVITE Response 487 Request Terminated: Terminate c4a9c5be@pbx

[7] 2011/03/30 16:09:37: update_codecs for ZjQ3ZWVkMTQ1YmFmMzgxZjk1NmJjMTgwMjEzZjE5Y2Y.: adding codec pcmu/8000 to available list

[7] 2011/03/30 16:09:37: update_codecs for ZjQ3ZWVkMTQ1YmFmMzgxZjk1NmJjMTgwMjEzZjE5Y2Y.: Other side does not support codec pcma/8000

[7] 2011/03/30 16:09:37: update_codecs for ZjQ3ZWVkMTQ1YmFmMzgxZjk1NmJjMTgwMjEzZjE5Y2Y.: Other side does not support codec g722/8000

[7] 2011/03/30 16:09:37: update_codecs for ZjQ3ZWVkMTQ1YmFmMzgxZjk1NmJjMTgwMjEzZjE5Y2Y.: Other side does not support codec g729/8000

[7] 2011/03/30 16:09:37: update_codecs for ZjQ3ZWVkMTQ1YmFmMzgxZjk1NmJjMTgwMjEzZjE5Y2Y.: Other side does not support codec g726-32/8000

[7] 2011/03/30 16:09:37: update_codecs for ZjQ3ZWVkMTQ1YmFmMzgxZjk1NmJjMTgwMjEzZjE5Y2Y.: Other side does not support codec gsm/8000

[7] 2011/03/30 16:09:37: update_codecs for ZjQ3ZWVkMTQ1YmFmMzgxZjk1NmJjMTgwMjEzZjE5Y2Y.: codec_preference size 7, available codecs size 2

[6] 2011/03/30 16:09:37: Codec pcmu/8000 is chosen for call id ZjQ3ZWVkMTQ1YmFmMzgxZjk1NmJjMTgwMjEzZjE5Y2Y.

[6] 2011/03/30 16:09:37: Call hold from trunk

[8] 2011/03/30 16:09:37: Packet authenticated by transport layer

[8] 2011/03/30 16:09:37: SRTP MAC mismatch: a5fa72d9 != df10d397

[7] 2011/03/30 16:09:37: Discard SRTCP packet from 10.0.3.231:54837 with wrong MAC

[8] 2011/03/30 16:09:38: Packet authenticated by transport layer

[8] 2011/03/30 16:09:39: Last message repeated 3 times

[8] 2011/03/30 16:09:39: SRTP MAC mismatch: e815da8b != 5e45a75f

[7] 2011/03/30 16:09:39: Discard SRTCP packet from 10.0.3.231:54837 with wrong MAC

[8] 2011/03/30 16:09:40: Packet authenticated by transport layer

 

(not sure if I cut that right).

 

Any idea? It could well be that the people using the phones have hit a button by mistake, but it is not the reply button when one is calling as they never hears any signals in the first place. (But they have managed to activate DND by mistake a number of times.) I cannot figure out what is going on though ... .

 

Thanks, Jerry

Link to comment
Share on other sites

currently the m9 has lots of issues---but none you mention! a new FW is on way this week. (these expectations have been set before so we'll see)

 

 

-accidentally pressed DND?!! on mine i currently have to press in a PIN to changed DND! ;-)

-there is a setting in the m9 to automatically answers calls as intercom (i think that is the languague) turn it off.

 

matt

Link to comment
Share on other sites

currently the m9 has lots of issues---but none you mention! a new FW is on way this week. (these expectations have been set before so we'll see)

 

 

-accidentally pressed DND?!! on mine i currently have to press in a PIN to changed DND! ;-)

-there is a setting in the m9 to automatically answers calls as intercom (i think that is the languague) turn it off.

 

matt

 

Hmm -- yes, one problem we realised after getting the m9s was that the customer in question really need long range connections and they are no good in that respect and apparently there are no "repeaters" available either. Another problem is probably the users -- nursery employees, where they have to play with kids outside and still need their phones with them .... .

 

DND - ah, I forgot how to change that on the m9s -- so then I know for sure it is on their Snom 320 (on the same extension) where they have fiddled with the button ... maybe I could disable it ... .

 

But the main problem here is calls going through without anyone picking up the handsets and that creates a security problem which is very important (crucial) here.

 

/Jerry

Link to comment
Share on other sites

i'm not sure how that is happening. It isn't on my unless you have the phone set to intercom answering mode. (not sure that is the right languague)

 

no repeaters available is true.

 

I will also try and inspect the handset (maybe it is only one, not quite sure yet) -- might be some mechanical problem ... .

Link to comment
Share on other sites

No, Intercom is not on. I get one signal on my side and then I get through and can hear everything on the other side, but they never hear a signal on their side and only hears me if I say something load enough. The behaviour was back now -- I was inspecting the phones and restarting them earlier today and then all worked fine (of course). The log from the m9:

 

2011/03/31 20:30:27 [sIP-Reg:5]: SIP Tx tls:10.0.3.10:5061:

SUBSCRIBE sip:41@pbx.company.com SIP/2.0

Via: SIP/2.0/TLS 10.0.3.231:4228;branch=z9hG4bK-gw2esf;rport

From: "Noaks Ark 29" <sip:41@pbx.company.com>;tag=5uy9iw

To: "Noaks Ark 29" <sip:41@pbx.company.com>

Call-ID: 5n6e66sb@snom

CSeq: 15722 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:41@10.0.3.231:4228;transport=tls;line=48asmd>;reg-id=1;+sip.instance="<urn:uuid:d48a0e1e-bba2-4c33-a19a-5804d0708584>"

Supported: outbound, gruu

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom-m9/9.2.42-b

Expires: 120

Content-Length: 0

 

2011/03/31 20:30:26 [sIP-Reg:5]: SIP Rx tls:10.0.3.10:5061:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.0.3.231:4228;branch=z9hG4bK-r3vgqe;rport=4228

From: "Noaks Ark" <sip:40@pbx.company.com>;tag=toa3ao

To: "Noaks Ark" <sip:40@pbx.company.com>;tag=67f953fa50

Call-ID: 3ze5h53f@snom

CSeq: 7067 SUBSCRIBE

Contact: <sip:10.0.3.10:5061;transport=tls>

Require: outbound

Expires: 120

Content-Length: 0

 

2011/03/31 20:30:26 [sIP-Reg:5]: SIP Tx tls:10.0.3.10:5061:

SUBSCRIBE sip:40@pbx.company.com SIP/2.0

Via: SIP/2.0/TLS 10.0.3.231:4228;branch=z9hG4bK-r3vgqe;rport

From: "Noaks Ark" <sip:40@pbx.company.com>;tag=toa3ao

To: "Noaks Ark" <sip:40@pbx.company.com>

Call-ID: 3ze5h53f@snom

CSeq: 7067 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@10.0.3.231:4228;transport=tls;line=54j08s>;reg-id=1;+sip.instance="<urn:uuid:475442e6-c2c3-4b7f-a6bd-2be1849f0a0b>"

Supported: outbound, gruu

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom-m9/9.2.42-b

Expires: 120

Content-Length: 0

 

2011/03/31 20:30:14 [sIP-Call:5]: SIP Rx tls:10.0.3.10:5061:

ACK sip:40@10.0.3.231:4228;transport=tls;line=6tb41i SIP/2.0

Via: SIP/2.0/TLS 10.0.3.10:5061;branch=z9hG4bK-7dac033c448ca302ec96fad6439d558e;rport

From: "Anonymous" <sip:anonymous@10.0.3.10;user=phone>;tag=1081400685

To: "Noaks Ark" <sip:40@10.0.3.10>;tag=pdkdqy

Call-ID: e519427a@pbx

CSeq: 16426 ACK

Max-Forwards: 70

Contact: <sip:40@10.0.3.10:5061;transport=tls>

Content-Length: 0

 

2011/03/31 20:30:14 [sIP-Call:5]: SIP Tx tls:10.0.3.10:5061:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.0.3.10:5061;branch=z9hG4bK-b95f12d24d36faf9b7bb6131074719e0;rport=5061

From: "Anonymous" <sip:anonymous@10.0.3.10;user=phone>;tag=1081400685

To: "Noaks Ark" <sip:40@10.0.3.10>;tag=pdkdqy

Call-ID: e519427a@pbx

CSeq: 16426 INVITE

Contact: <sip:40@10.0.3.231:4228;transport=tls;line=6tb41i>

Supported: 100rel, replaces, norefersub

User-Agent: snom-m9/9.2.42-b

Content-Type: application/sdp

Content-Length: 303

 

v=0

o=root 181735547 181735548 IN IP4 10.0.3.231

s=-

c=IN IP4 10.0.3.231

t=0 0

m=audio 61490 RTP/AVP 0 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:5U9Whv7bgZbRLzPV8JiAKB7bWpJyc7hYNvmtNgN7|2^31

a=sendrecv

2011/03/31 20:30:14 [DECT:5]: Send Call Status: encrypted

2011/03/31 20:30:14 [DECT:5]: hanset 3 picked up the call

2011/03/31 20:30:13 [sIP-Call:5]: SIP Tx tls:10.0.3.10:5061:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 10.0.3.10:5061;branch=z9hG4bK-e720444c43d23cdf46ca8876f3bd7cab;rport=5061

From: "Anonymous" <sip:anonymous@10.0.3.10;user=phone>;tag=1081400685

To: "Noaks Ark" <sip:40@10.0.3.10>;tag=pdkdqy

Call-ID: e519427a@pbx

CSeq: 16427 PRACK

Contact: <sip:40@10.0.3.231:4228;transport=tls;line=6tb41i>

Supported: 100rel, replaces, norefersub

Content-Length: 0

_________

 

"hanset 3" seems to be good on picking up calls and I guess we will have to replace it .... .

Link to comment
Share on other sites

Hanged up after completing the previous post and then called again and then get many tones, so it seems like it has something to do with the ending/closing of the calls after they have been picked up, but what they do remains a mystery ... (or if this may just be random behaviour).

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

Loading...
 Share

×
×
  • Create New...