External extension

[p800aul]

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

[pbx support]

In which version this was working?

BTW, the log did not show that the call was connected.

[Vodia PBX]

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.

[p800aul]

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

[Vodia PBX]

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.

[p800aul]

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

[Vodia PBX]

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

[Bill H]

Try using the information in the attached document.

Snom IP Route.zip

[p800aul]

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

 

:-)

[Vodia PBX]

If you have two interfaces then you must (1) make sure that the traffic towards the public internet is using the designated interface (thats a topic for the routing table, which can be influenced with the metric or otherwise explicitly with the route command) and (2) make sure that the PBX replaces the IP address on the interface that points to the internet with the IP address that the router will insert later. This can be done with "IP Routing List" of the PBX, for example "192.168.0.0/255.255.0.0/192.168.1.2 0.0.0.0/0.0.0.0/123.124.125.126" (note the space between the route entries) if 192.168.1.2 is the private IP address of the interface that points into the LAN and 123.124.125.126 is the public IP address that is used on the router.

 

I know this is all very complicated, it would have been a better world if SIP have been designed with NAT problems in mind. STUN does also not solve the problem, because this is the server, not the client :-(