p800aul
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Posts posted by p800aul
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I don't get the point.
I will assume you mean "the point" in my question rather than the point in replying.
My point is that I asked if it is Snom's last word on the matter of looking after those that used the snomONE free Permanent license version (10 extension) when buying expensive Snom gear. As far as I can see most distributors of Snom equipment sold Snom phones etc based on the fact that SME's with less than 10 extensions could use snomOne free. Also as far as I can see at no time was this called an evaluation copy.
As I've said I understand the shift in strategy, although I think it's misguided considering the reducing costs of voip communications for SME's. All I asked for is that those who already have the snomOne free system should be allowed to maintain it in the most up to date form. No doubt future decisions will be made on the price and suitability of your software against others out in the market.
To be honest the work we and others did on testing the M9 for you should make Snom more than grateful and supply all your software for free, I personally spent days testing the various firmware updates.
Regards
Paul
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This license is not available any more. If you want to evaluate the software, you'll have to do this with 5 extensions now.
Hi
considering my post above is this your last word on the matter?
Regards
Paul
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Sorry but I can't give you a free pass on this (no pun intended)
The 10 license version was marketed as snomONE free and the software was limited to (2) non snom devices without the transfer ability. It was widely understood that snom was subsidizing the software to sell their phones. I understand the new software license, based upon the marketing and license is for evaluation and have no problem with the pricing for new installations, I actually think it is quite fair.Yep
And a difficult argument when they see:
Uptime: 10 21:00:42 (DD HH:MM:SS)
Memory Status (PBX Usage/Total System Memory, Total Used Memory): 51MB/2037MB, 18%
Version: 2011-4.5.0.1030 Beta Corona Austrinids (Win32)
Build date: Feb 9 2012 10:52:44
License status: snom ONE free
License duration: Permanent
on their PBX
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I am guessing that it is because they want you to pay for it. I must say that Version 5 seams very interesting but still...
What bothers me is probably the same issues you have, Basing an installation on a free licence and being force to purchase after...
I understand that they need to sell and make money, changing the licence model and removing what was agreed does not seems to be the right approche to me. This look like the best way of making costumer unhappy. This is the second time that PBXnSIP/Vodia does that...
Your best chance will be to find someone that have the installer and can share it with you.
I don't need the installer just the pbxctrl.exe for the last 4.5.1, if they want to stop new installs that all they have to supply.
regards
Paul
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This license is not available any more. If you want to evaluate the software, you'll have to do this with 5 extensions now.
The thing is I thought as I suspect a few did that I wasn't evaluating software I thought I was using software for the 7 Snom phones I bought to work. My belief was driven by the following email I receive every evening from the PBX thus:
Uptime: 10 21:00:42 (DD HH:MM:SS)
Memory Status (PBX Usage/Total System Memory, Total Used Memory): 51MB/2037MB, 18%
Version: 2011-4.5.0.1030 Beta Corona Austrinids (Win32)
Build date: Feb 9 2012 10:52:44
License status: snom ONE free
License duration: Permanent
I was one of the early adopters of the M9 which to be frank is only now workable (I still have one that randomly stops working) and while I understand you may want to move to a new licencing model I would ask, as this is clearly not a "new" install, why I can not have the latest ver 4 software?
Regards
Paul
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The installers were on the snom web page, and there they have been removed.
New installations should use version 5. I don't see why that should be a problem. AFAIK we did not drop a single feature in version 5.
So how do I get the key for 10 user free in 5.?
or a 10 user key in 4.5.1.?
Regards
Paul
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Same here already have 2011-4.5.0.1030 Beta Corona Austrinids (Win32)would like pbxctrl-v4.5.0.1090.exe
Anyone
Thanks
Paul
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Yea the PBX was designed to support remote SIP registrations as much as possible, including NAT, two-tier NAT, full cone, symmetrical you name it. Many gateways dont support registrations, and the other problem is how to tell the gateway which number to call (Request-URI does not work because that's fixed in the registration). Also for incoming call the PBX will believe that the call comes from a extension, not from a trunk--which might be challenging for avoiding toll fraud. I would say give it a shot; maybe it works relatively easily and then you are all set; if you run into trouble I would consider the VPN method.
In the interest of completeness I thought I would report back.
In testing this unit it does what it says on the tin, i.e. it registers with the snom one from a remote location and works using a pots type telephone on one of the two ports available, for both PSTN and VOIP.
I haven't set up the dial plan for the router yet to route calls either via the remote snom one or PSTN, but it looks a simple affair.
The next task is to see if I can get my snom 300 telephone to work with it for both routings (if anyone has tried that I'd love to know
)Regards
Paul
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As soon as you introduce NAT things get tricky (but not impossible).
I see that said a lot but the snom300 at the remote site is working through NAT just fine. I suppose I'm asking if the Snom300 works through my current router (netgear domestic type)on NAT. Should (all things being equal) i be able to get the Vigor to connect using similar if not the same sip settings? See Here
Thanks for your input
Paul
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Hi
This is a just before I buy question really.
I have a snomone setup which works well in the main office location and one external extension (snom300) at a remote site which also works well.
At the remote site I also have pots line on which i have a regular telephone. i.e. two phones one attached to the snomone and one PSTN.
I would like to attach both to one telephone and wondered if anyone had done this with Vigor 2710Vn which has Twin VoIP Phone Ports with POTS passthrough.
Apparently with this I "can choose to make calls using VoIP via the Internet or switch over to your POTS line (your normal phone line) and dial out via the PSTN (the conventional phone network). By setting up the LCR (Least Cost Routing) facility, you can automate this process so that the router automatically sets your preferred route for calls, according to your own rules. Then, depending on the destination dialled, the router will use either your POTS line or one of up to 4 VoIP/SIP providers/gateways (for example DrayTEL)".
Which seems ideal
I would guess I can attach to the snomone on the VOIP side (as a SIP provider) and use the pots on the other?
Anyone tried/done this or agree that it should work?
Thanks
Paul
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Did you see http://wiki.snomone.com/index.php?title=Server_Behind_NAT? I think that pages describes pretty much your situation.
Once we all have IPv6, life will be a lot easier
So looking at this I should move the metric to 1 on the routable and 2 on the internal?
Or
Do I use bill's example and place 192.168.xxx.xxx (internal ip of the PBX)/81.176.xxx.xxx (routable ip of the PBX)in the indicated position of the PBX
or
Both
:-)
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Not the biggest expert here, but you can set the metrics of the interface to make sure that the public IP is used when sending traffic out to the Internet. Use route -print to show the route.
Hi
Here is what I've got I would be grateful if anyone could check if I have it right. As I said before I had this working before and I've changed something that's stopped it working. The external ext is in Spain and the PBX is in the UK. I've changed the PBX software version as an when, I have also changed the PBX PC for one that was identical to the original (maybe I've missed something!)
I have routable IP (123.123.123.123) serving a 192.168 internal network router from which the the PBX gets a DHCP address set at the router by reservation on one nic. A further nic gets a static routable IP (123.123.123.124)for the PBX.
I set both of them up as is, without any additional settings. this worked fine on the local network but on testing from the external ext it registered with the PBX but no sound in or out, even on auto attendant voice mail etc. As the poster Snom One suggested it appeared that there was a routing issue. Rather than messing with the metric I made sure that the default gateways matched, in other words both nic's pointing to the router, the one handling the routable IP's.This fixed the issue for about a day, then today the external ext is still attached the PBX but no sound again (I've changed nothing.
While looking at what could be wrong I did a tracert from the PBX to look at the route it was taking and it used the routable router to go out rather than the internal, low and behold the external ext worked again.
Clearly I do not understand what I am doing and would love help in understanding what's is going on and how to fix it permanently
Sorry in advance if I'm being an idiot:-)
Regards
Paul
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If the problem is between the phone and the PBX, I would first try something local, e.g. calling the auto attendant. Then it is easier to nail the problem. From the above, the phone is able to send and receive SIP packets to and from the PBX, which is pretty good. However it tells the phone to send the media to an unroutable (private) IP address, which is not so good. Your PBX has a NAT problem :-( maybe it is just the routing table on the PBX that needs to be fixed, if you have already a public IP address.
Thanks
I've put the local ip card to DHCP (set the ip via an address reservation)
I've tried both the auto attendant and voice mail still nothing, how do you suggest I fix the routing table i've released and renewed via ipconfig.
regards
Paul
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Hi
I have a simple set up snom one 2011-4.5.0.1030 Beta Corona Austrinids (Win32)6 ext including one in a remote location. I've had the remote ext working previously without a problem. As the remote ext (snom 300) is only used periodically (it has no internet connection when not in use)as it's in Spain with the pbx in the UK. This time the ext has registered on the system but I can not hear any calls and the caller can not hear me.
The only thing that's changed is the pbx version I've tried the one previous to 2011-4.5.0.1030 Beta Corona Austrinids to no effect.
Can anyone offer any clues as to where the problem my lie?
Below is the log from a call from the remote ext to a mobile using sipgate using a pots line has the same effect
Thanks
Paul
log
[5] 2012/04/03 17:13:23: SIP Rx tls:193.239.14.1:2111:
INVITE sip:0797xxxxxx@localhost;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.11.25:2111;branch=z9hG4bK-jtq2wxotsyjt;rport
From: "Jardines" <sip:47@localhost>;tag=mczlnvi840
To: <sip:0797xxxxxx@localhost;user=phone>
Call-ID: 3c27b10c0f1d-a7j5f14za2ri
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:47@192.168.11.25:2111;transport=tls;line=yngjfnmi>;reg-id=1
X-Serialnumber: 00041336AD2C
P-Key-Flags: keys="3"
User-Agent: snom300/8.4.32
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: 526
v=0
o=root 1514403803 1514403803 IN IP4 192.168.11.25
s=call
c=IN IP4 192.168.11.25
t=0 0
m=audio 63506 RTP/AVP 9 0 8 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:rqDjcnBB/UDGIzWZM/Mz+lTu1ZoRtNcgmTx/Sz4/
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
[5] 2012/04/03 17:13:23: SIP Tx tls:193.239.14.1:2111:
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.11.25:2111;branch=z9hG4bK-jtq2wxotsyjt;rport=2111;received=193.239.14.1
From: "Jardines" <sip:47@localhost>;tag=mczlnvi840
To: <sip:0797xxxxxx@localhost;user=phone>;tag=b45176d1ce
Call-ID: 3c27b10c0f1d-a7j5f14za2ri
CSeq: 1 INVITE
Content-Length: 0
[5] 2012/04/03 17:13:23: Dialplan "Standard Dialplan": Match 0797xxxxxx@localhost to sip:0797xxxxxx@sipgate.co.uk;user=phone on trunk SipGate
[5] 2012/04/03 17:13:23: SIP Tx udp:217.10.79.23:5060:
INVITE sip:0797xxxxxx@sipgate.co.uk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-e9a32275d904e27e1865141a12367e11;rport
From: "SipGate" <sip:1175451@sipgate.co.uk>;tag=42546
To: <sip:0797xxxxxx@sipgate.co.uk>
Call-ID: 4e16a894@pbx
CSeq: 12949 INVITE
Max-Forwards: 70
Contact: <sip:1175451@192.168.1.13:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids
Remote-Party-ID: "Jardines" <sip:01246xxxxxx@localhost;user=phone>
Privacy: id
P-Charging-Vector: icid-value=;icid-generated-at=192.168.1.13;orig-ioi=localhost
Content-Type: application/sdp
Content-Length: 327
v=0
o=- 24482 24482 IN IP4 192.168.1.13
s=-
c=IN IP4 192.168.1.13
t=0 0
m=audio 49730 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
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] 2012/04/03 17:13:23: set codec: codec pcmu/8000 is set to call-leg 0
[5] 2012/04/03 17:13:23: SIP Tx tls:193.239.14.1:2111:
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 192.168.11.25:2111;branch=z9hG4bK-jtq2wxotsyjt;rport=2111;received=193.239.14.1
From: "Jardines" <sip:47@localhost>;tag=mczlnvi840
To: <sip:0797xxxxxx@localhost;user=phone>;tag=b45176d1ce
Call-ID: 3c27b10c0f1d-a7j5f14za2ri
CSeq: 1 INVITE
Contact: <sip:47@81.143.XXX.XXX: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/2011-4.5.0.1030 Beta Corona Austrinids
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 423
v=0
o=- 36384 36384 IN IP4 192.168.1.13
s=-
c=IN IP4 192.168.1.13
t=0 0
m=audio 58276 RTP/AVP 0 8 9 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:9o9j1C2RM6dhofrUMPEcjEYaw9aw8rqScDDZVHp2
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
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] 2012/04/03 17:13:23: SIP Rx udp:217.10.79.23:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.13:5060;received=81.143.137.173;branch=z9hG4bK-e9a32275d904e27e1865141a12367e11;rport=5060
From: "SipGate" <sip:1175451@sipgate.co.uk>;tag=42546
To: <sip:0797xxxxxx@sipgate.co.uk>;tag=6d6e7f8f352adddb20da2b196524dfa8.e775
Call-ID: 4e16a894@pbx
CSeq: 12949 INVITE
Proxy-Authenticate: Digest realm="sipgate.co.uk", nonce="4f7b15e16e978f46f73d28b5e7f176df57b71688"
Content-Length: 0
[5] 2012/04/03 17:13:23: SIP Tx udp:217.10.79.23:5060:
INVITE sip:0797xxxxxx@sipgate.co.uk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-cf0472c374f4ecaa67c158f920221bfb;rport
From: "SipGate" <sip:1175451@sipgate.co.uk>;tag=42546
To: <sip:0797xxxxxx@sipgate.co.uk>
Call-ID: 4e16a894@pbx
CSeq: 12950 INVITE
Max-Forwards: 70
Contact: <sip:1175451@192.168.1.13:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids
Remote-Party-ID: "Jardines" <sip:01246xxxxxx@localhost;user=phone>
Privacy: id
P-Charging-Vector: icid-value=;icid-generated-at=192.168.1.13;orig-ioi=localhost
Proxy-Authorization: Digest realm="sipgate.co.uk",nonce="4f7b15e16e978f46f73d28b5e7f176df57b71688",response="48bbdb7c4a5390ee7bccea87d6fea33f",username="1175451",uri="sip:0797xxxxxx@sipgate.co.uk;user=phone",algorithm=MD5
Content-Type: application/sdp
Content-Length: 327
v=0
o=- 24482 24482 IN IP4 192.168.1.13
s=-
c=IN IP4 192.168.1.13
t=0 0
m=audio 49730 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
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] 2012/04/03 17:13:24: SIP Rx udp:217.10.79.23:5060:
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 192.168.1.13:5060;received=81.143.137.173;branch=z9hG4bK-cf0472c374f4ecaa67c158f920221bfb;rport=5060
From: "SipGate" <sip:1175451@sipgate.co.uk>;tag=42546
To: <sip:0797xxxxxx@sipgate.co.uk>
Call-ID: 4e16a894@pbx
CSeq: 12950 INVITE
Content-Length: 0
[5] 2012/04/03 17:13:24: SIP Rx tls:193.239.14.1:2111:
PRACK sip:47@81.143.XXX.XXX:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.11.25:2111;branch=z9hG4bK-azaojtbpk1ev;rport
From: "Jardines" <sip:47@localhost>;tag=mczlnvi840
To: <sip:0797xxxxxx@localhost;user=phone>;tag=b45176d1ce
Call-ID: 3c27b10c0f1d-a7j5f14za2ri
CSeq: 2 PRACK
Max-Forwards: 70
Contact: <sip:47@192.168.11.25:2111;transport=tls;line=yngjfnmi>;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
[5] 2012/04/03 17:13:24: SIP Tx tls:193.239.14.1:2111:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.11.25:2111;branch=z9hG4bK-azaojtbpk1ev;rport=2111;received=193.239.14.1
From: "Jardines" <sip:47@localhost>;tag=mczlnvi840
To: <sip:0797xxxxxx@localhost;user=phone>;tag=b45176d1ce
Call-ID: 3c27b10c0f1d-a7j5f14za2ri
CSeq: 2 PRACK
Contact: <sip:47@192.168.1.13:5061;transport=tls>
User-Agent: snomONE/2011-4.5.0.1030 Beta Corona Austrinids
Content-Length: 0
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Hi
I've tried this update and I get error 1053 :the service did not respond to the start or control request in a timely fashion
Any clues?
Regards
Paul
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Oops, missed that totally.
The issue is that on the PBX, "Patton" trunk type is set to "SIP Registration", but the trunk is not registered. In .4025, we skip the unregistered trunks for outbound calls. You can do either - make sure that the trunk is registered OR change the trunk type to "SIP Gateway" to avoid the issue.
The registration type trunk Patton is not registered. Skipping it...
Thanks
That seems to have fixed it
Regards
Paul
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Hi
Any thoughts on my issues above pbx support?
thanks
Paul
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There you go
Thanks
.3981
[6] 2011/07/01 18:09:22: Received bindRequest for user localhost\48 [5] 2011/07/01 18:09:25: SIP Rx tls:192.168.1.8:2778: INVITE sip:819161@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.8:2778;branch=z9hG4bK-h41any7bdh3p;rport From: "Study" <sip:48@localhost>;tag=qzv6db7x6u To: <sip:819161@localhost;user=phone> Call-ID: 3c641d1fe2f7-4xrvrha9is0i CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:48@192.168.1.8:2778;transport=tls;line=mjvwc7ij>;reg-id=1 X-Serialnumber: 00041336B86D P-Key-Flags: keys="3" User-Agent: snom300/8.4.31 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: 520 v=0 o=root 391218360 391218360 IN IP4 192.168.1.8 s=call c=IN IP4 192.168.1.8 t=0 0 m=audio 56900 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:AbJQ3lgRvtJ7BbbTxRK15Rg3nXBsgROXGul4+G7J 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] 2011/07/01 18:09:25: Packet authenticated by transport layer [9] 2011/07/01 18:09:25: UDP: Opening socket on 0.0.0.0:58970 [9] 2011/07/01 18:09:25: UDP: Opening socket on 0.0.0.0:58971 [8] 2011/07/01 18:09:25: Could not find a trunk (2 trunks) [5] 2011/07/01 18:09:25: SIP Rx tls:192.168.1.8:2778: INVITE sip:819161@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.8:2778;branch=z9hG4bK-h41any7bdh3p;rport From: "Study" <sip:48@localhost>;tag=qzv6db7x6u To: <sip:819161@localhost;user=phone> Call-ID: 3c641d1fe2f7-4xrvrha9is0i CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:48@192.168.1.8:2778;transport=tls;line=mjvwc7ij>;reg-id=1 X-Serialnumber: 00041336B86D P-Key-Flags: keys="3" User-Agent: snom300/8.4.31 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: 520 v=0 o=root 391218360 391218360 IN IP4 192.168.1.8 s=call c=IN IP4 192.168.1.8 t=0 0 m=audio 56900 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:AbJQ3lgRvtJ7BbbTxRK15Rg3nXBsgROXGul4+G7J 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 [9] 2011/07/01 18:09:25: Using outbound proxy sip:192.168.1.8:2778;transport=tls because of flow-label [9] 2011/07/01 18:09:25: Last message repeated 3 times [6] 2011/07/01 18:09:25: Received bindRequest for user localhost\48 [5] 2011/07/01 18:09:25: SIP Tx tls:192.168.1.8:2778: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.1.8:2778;branch=z9hG4bK-h41any7bdh3p;rport=2778 From: "Study" <sip:48@localhost>;tag=qzv6db7x6u To: <sip:819161@localhost;user=phone>;tag=ebe544b72c Call-ID: 3c641d1fe2f7-4xrvrha9is0i CSeq: 1 INVITE Content-Length: 0 [7] 2011/07/01 18:09:25: Set packet length to 20 [6] 2011/07/01 18:09:25: Sending RTP for 3c641d1fe2f7-4xrvrha9is0i to 192.168.1.8:56900, codec not set yet [8] 2011/07/01 18:09:25: Call from an user 48 [8] 2011/07/01 18:09:25: To is <sip:819161@localhost;user=phone>, user 0, domain 1 [8] 2011/07/01 18:09:25: From user 48 [8] 2011/07/01 18:09:25: Set the To domain based on From user 48@localhost [8] 2011/07/01 18:09:25: Call state for call object 13: idle [7] 2011/07/01 18:09:25: set_codecs: for 3c641d1fe2f7-4xrvrha9is0i codecs "", codec_preference count 6 [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^(999)@.*!sip:\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^0800([0-9]*)@.*!sip:0800\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^00([0-9]*)@.*!sip:00\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^07([0-9]*)@.*!sip:07\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^907([0-9]*)@.*!sip:07\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^900([0-9]*)@.*!sip:00\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:09:25: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 819161@localhost [5] 2011/07/01 18:09:25: Dialplan "Standard Dialplan": Match 819161@localhost to <sip:819161@192.168.1.200;user=phone> on trunk Patton [8] 2011/07/01 18:09:25: Play audio_moh/noise.wav [9] 2011/07/01 18:09:25: UDP: Opening socket on 0.0.0.0:59340 [9] 2011/07/01 18:09:25: UDP: Opening socket on 0.0.0.0:59341 [7] 2011/07/01 18:09:25: set_codecs: for 83572110@pbx codecs "", codec_preference count 6 [9] 2011/07/01 18:09:25: update_codecs for 83572110@pbx: adding codec pcmu/8000 to available list [9] 2011/07/01 18:09:25: update_codecs for 83572110@pbx: adding codec pcma/8000 to available list [9] 2011/07/01 18:09:25: update_codecs for 83572110@pbx: adding codec g722/8000 to available list [9] 2011/07/01 18:09:25: update_codecs for 83572110@pbx: adding codec g726-32/8000 to available list [9] 2011/07/01 18:09:25: update_codecs for 83572110@pbx: adding codec gsm/8000 to available list [9] 2011/07/01 18:09:25: update_codecs for 83572110@pbx: codec_preference size 6, available codecs size 6 [9] 2011/07/01 18:09:25: Resolve 12472: url sip:192.168.1.200 [9] 2011/07/01 18:09:25: Resolve 12472: udp 192.168.1.200 5060 [5] 2011/07/01 18:09:25: SIP Tx udp:192.168.1.200:5060: INVITE sip:819161@192.168.1.200;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-72902ae4db65e28d50e1980ed6df68bf;rport From: "Study" <sip:01246819161@localhost;user=phone>;tag=45409 To: <sip:819161@192.168.1.200;user=phone> Call-ID: 83572110@pbx CSeq: 900 INVITE Max-Forwards: 70 Contact: <sip:01246819161@192.168.1.13:5060;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: 327 v=0 o=- 41145 41145 IN IP4 192.168.1.13 s=- c=IN IP4 192.168.1.13 t=0 0 m=audio 59340 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 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
4025
[6] 2011/07/01 18:16:31: Received bindRequest for user localhost\48 [6] 2011/07/01 18:16:33: Last message repeated 2 times [7] 2011/07/01 18:16:33: SIP Rx tls:192.168.1.8:2782: INVITE sip:819161@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.8:2782;branch=z9hG4bK-v0svqzc2wfnk;rport From: "Study" <sip:48@localhost>;tag=xy3mypr1cv To: <sip:819161@localhost;user=phone> Call-ID: 3c641eccef30-an7w0fgu4eg2 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:48@192.168.1.8:2782;transport=tls;line=mjvwc7ij>;reg-id=1 X-Serialnumber: 00041336B86D P-Key-Flags: keys="3" User-Agent: snom300/8.4.31 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: 522 v=0 o=root 1034786031 1034786031 IN IP4 192.168.1.8 s=call c=IN IP4 192.168.1.8 t=0 0 m=audio 54922 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XhQqTHLltypTJWC5vDrHpGfZkxH45okk1VH+jdSi 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] 2011/07/01 18:16:33: Packet authenticated by transport layer [9] 2011/07/01 18:16:33: UDP: Opening socket on 0.0.0.0:60914 [9] 2011/07/01 18:16:33: UDP: Opening socket on 0.0.0.0:60915 [8] 2011/07/01 18:16:33: Could not find a trunk (2 trunks) [9] 2011/07/01 18:16:33: Using outbound proxy sip:192.168.1.8:2782;transport=tls because of flow-label [9] 2011/07/01 18:16:33: Last message repeated 3 times [7] 2011/07/01 18:16:33: SIP Tx tls:192.168.1.8:2782: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.1.8:2782;branch=z9hG4bK-v0svqzc2wfnk;rport=2782 From: "Study" <sip:48@localhost>;tag=xy3mypr1cv To: <sip:819161@localhost;user=phone>;tag=1abb0eb12d Call-ID: 3c641eccef30-an7w0fgu4eg2 CSeq: 1 INVITE Content-Length: 0 [7] 2011/07/01 18:16:33: Set packet length to 20 [6] 2011/07/01 18:16:33: Sending RTP for 3c641eccef30-an7w0fgu4eg2 to 192.168.1.8:54922, codec not set yet [8] 2011/07/01 18:16:33: Incoming call: Request URI sip:819161@localhost;user=phone, To is <sip:819161@localhost;user=phone> [8] 2011/07/01 18:16:33: Call from an user 48 [8] 2011/07/01 18:16:33: To is <sip:819161@localhost;user=phone>, user 0, domain 1 [8] 2011/07/01 18:16:33: From user 48 [8] 2011/07/01 18:16:33: Set the To domain based on From user 48@localhost [8] 2011/07/01 18:16:33: Call state for call object 1: idle [7] 2011/07/01 18:16:33: set_codecs: for 3c641eccef30-an7w0fgu4eg2 codecs "", codec_preference count 6 [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^(999)@.*!sip:\1@\r;user=phone!i against 819161@localhost [6] 2011/07/01 18:16:33: The registration type trunk Patton is not registered. Skipping it... [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^0800([0-9]*)@.*!sip:0800\1@\r;user=phone!i against 819161@localhost [6] 2011/07/01 18:16:33: The registration type trunk Patton is not registered. Skipping it... [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^00([0-9]*)@.*!sip:00\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^07([0-9]*)@.*!sip:07\1@\r;user=phone!i against 819161@localhost [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^907([0-9]*)@.*!sip:07\1@\r;user=phone!i against 819161@localhost [6] 2011/07/01 18:16:33: The registration type trunk Patton is not registered. Skipping it... [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^900([0-9]*)@.*!sip:00\1@\r;user=phone!i against 819161@localhost [6] 2011/07/01 18:16:33: The registration type trunk Patton is not registered. Skipping it... [9] 2011/07/01 18:16:33: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 819161@localhost [6] 2011/07/01 18:16:33: The registration type trunk Patton is not registered. Skipping it... [8] 2011/07/01 18:16:33: call port 0: state code from 0 to 404 [7] 2011/07/01 18:16:33: Set packet length to 20 [7] 2011/07/01 18:16:33: SIP Tx tls:192.168.1.8:2782: SIP/2.0 404 Not Found Via: SIP/2.0/TLS 192.168.1.8:2782;branch=z9hG4bK-v0svqzc2wfnk;rport=2782 From: "Study" <sip:48@localhost>;tag=xy3mypr1cv To: <sip:819161@localhost;user=phone>;tag=1abb0eb12d Call-ID: 3c641eccef30-an7w0fgu4eg2 CSeq: 1 INVITE Contact: <sip:48@192.168.1.13:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.1.4025 Content-Length: 0 [6] 2011/07/01 18:16:33: Received searchRequest, equalityMatch (description=telephoneNumber, value=819161) [7] 2011/07/01 18:16:33: SIP Rx tls:192.168.1.8:2782: ACK sip:819161@localhost;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.8:2782;branch=z9hG4bK-v0svqzc2wfnk;rport From: "Study" <sip:48@localhost>;tag=xy3mypr1cv To: <sip:819161@localhost;user=phone>;tag=1abb0eb12d Call-ID: 3c641eccef30-an7w0fgu4eg2 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:48@192.168.1.8:2782;transport=tls;line=mjvwc7ij>;reg-id=1 Proxy-Require: buttons Content-Length: 0
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It's been a while since we released the last bug fix version(2011-4.2.0.3981) for snomONE. In the meanwhile, we heard the feedback from you in the form of forum requests, support tickets, etc. We tried to resolve most of the known issues during this time and as a result of this is the latest bug fix release 2011-4.2.1.4025.
For details such as release notes and download links please refer the below link
http://wiki.snomone.com/index.php?title=Release_notes
Thank you all!!!
Any reason why the patton trunk will not dial out using this the sipgate still works.
So previous version 2011-4.2.0.3981 worked incoming - outgoing sipgate and worked incoming - outgoing patton 2xfxo
Upgrade to 2011-4.2.1.4025, sipgate incoming - outgoing, patton incoming only - no outgoing (engaged tone).
Downgrade back to 2011-4.2.0.3981 everything fine again.
Regards
Paul
patton trunk set up:
# Trunk 5 in domain localhost
Name: Patton
Type: register
To: sip
RegPass: ********
Direction:
Disabled: false
Global: false
Display:
RegAccount:
RegRegistrar: 192.168.1.200
RegKeep:
RegUser:
Icid:
Require:
OutboundProxy: 192.168.1.200
Ani:
DialExtension: 72
Prefix:
Trusted: false
AcceptRedirect: false
RfcRtp: false
Analog: false
SendEmail:
UseUuid: false
Ring180: false
Failover: only_5xx
Privacy: false
Glob:
RequestTimeout:
Codecs:
CodecLock: true
Expires: 3600
FromUser:
Tel: true
TranscodeDtmf: false
AssociatedAddresses:
InterOffice: false
DialPlan:
Colines:
DialogPermission:
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The first thing that most probably causes problems is that the number in the From/To headers contain a "-". This is a valid character, as SIP URI can contain practically any character. In a SIP URI 978-746-2777 ir not equal to 9787462777. The gateway should by all means send the number without the dashes in the middle. (I guess the XXX is not in the real trace and just there to hide your real number).
And the other thing is that how can the To header have "anonymous"? The caller does not want to reveal information about who he wants to call?! There is something strange.
I've just done a further test bypassing the Patton using a sipgate line here are the results
From: "01246813xxx" <sip:01246813xxx@sipgate.co.uk;user=phone>;tag=48244
To: <sip:00441246887xxx@sipgate.co.uk;user=phone>
I see what you mean now, I guess it's the patton as both trunks are set up the same on the snom.
i'll get back to them and come back here if i get an answer
thanks
Regards
Paul
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I am talking about the "To" header
To: <sip:anonymous@192.168.1.13:5060;user=phone>;tag=vegrlg
Sorry but that's not much clearer
Anyway attached is a log file of an incoming missed call, hopefully this will help.
Regards
Paullogtrace.txt
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The 'To" is "anonymous"? Can you send the original (missed call) SIP messages?
I'm sorry I don't know what you mean the "to" is "anonymous" in the entire log. The call comes in via the FXO on a patton, the pbx picks it up on a hunt which rings all the handsets in the group, if no one picks up the call goes to an auto attendant which offers the opportunity to transfer to a mobile or leave a message.
if you can tell me where i might get the information you need to help me further i will go get it.
thanks
Paul
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Did you set a country code in your domain? That would explain why you see the strange formatting.
Nope both country and area code are blank.
The log shows the number comming in with a -
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-0e741a19b0c76a914c0a84e112ab6d9d;rport=5060
From: <sip:01246-813XX3@192.168.1.200:5060;user=phone>;tag=26412
To: <sip:anonymous@192.168.1.13:5060;user=phone>;tag=vegrlg
Call-ID: 225982df@pbx
CSeq: 25856 INVITE
Contact: <sip:42@192.168.1.2:3626;transport=udp;line=i49ilr>
Supported: 100rel, replaces, norefersub
User-Agent: snom-m9/9.3.9-a
Content-Length: 0
I've checked with Patton the FXO i'm using and they don't think it's them
Regards
Paul
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Hi
I'm sure this is simple and down to my inexperience with this system. I have a snom one system with a m9 set and a couple of 300. when i miss a call the number from landlines present like this 0114-278 376 as an example when i try to call this number i get a number unobtainable tone (quick beeps). incoming mobiles for some reason present as 07973222222 for example and dial without issue.
is there something I'm missing?
regards
Paul
The 10 license version was marketed as snomONE free and the software was limited to (2) non snom devices without the transfer ability. It was widely understood that snom was subsidizing the software to sell their phones. I understand the new software license, based upon the marketing and license is for evaluation and have no problem with the pricing for new installations, I actually think it is quite fair.
)
Installer download
in General Setup
Posted
This is not personal Snom One,I'm not asking you to go without lunch. My point (and it will be the last time I make it) is the software you provided was free, it was and is useful while we bought snom kit, so we used the free software and bought the snom kit, even though some of it was rubbish Snom m9 (Your UK sales Director told me it was rubbish.)I am not asking for more free software I'm am suggesting that the software (the .exe files) are made available to those that want to maintain the status quo, while (just maybe) we get the time and the inclination to upgrade to the paid version :-)
That's it I guess
You seem a little cheesed off with these questions and for that I'm sorry.
Regards
Paul