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Can't call other extensions from one 870 set


xy3
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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..?

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Looks like the phone always send a Alert-Info header?! If you look at the SIP packet than the phone sends out you should see it. I remember there is a setting on the phone that tells the phone that every call must be connected immediately. There are a lot of settings... Is it an option to factory reset the phone and plug and play it with the PBX?

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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

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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

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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??

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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

http://wiki.snomone.com/index.php?title=Upgrades#General_manual_upgrade_guidelines_for_Linux_based_systems

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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.

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You really need to know couple of things in order to manually upgrade the SoHo (of course, if the phone was in 8.4.31 version, we need not have upgraded the SoHo).

 

First, you need to 'ssh' to the SoHo.

Then on a terminal window, you will have to execute those commands.

 

If that is difficult for you, then please wait until we prepare a package with the latest. Then you can just click on the upgrade link and it will take care of the rest.

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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?

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Found and installed an older 'SSH Secure Shell' tool, which PBXNSIP folk appear to have made use of in past, and seems to run fine. Now struggling to establish SSH session with our SoHo.. What is correct port? Username = root? Password = ??? Just a *little* bit of help would get us going. HELP!

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