Ipphones in remote location

[mbs1]

Hi,

I hope someone can help.

Xlite in a remote location ( behind a nat ) can register to the PBX at a remote location ( also behind a nat). The correct ports are opened on the firewall both ways. The remote soft phone can call a phone ( ipphone or xlite ) on the same subnet as the pbx, ringing is heard but on pickup the trace shows continious icmp messages. The trace also show the remote XLite as having resolved and sent its public IP address in the SIP Invite message. However the OK Contact message subsequently returned to the Xlite from the PBX shows the internal private ip address of the called party.The remote xlite then tries to send RTP packets to the private ip via the internet which fail and the ICMPs.

Has anyone got remote Extensions working. Would appreciate any help. Have got the RC1 ( 6 march ) version installed.

thanks

mbs

 

[dekatech]

Same problem with IP Phone

I have the same problem with the Polycom IP Phone. It registers and downloads its' sip.id and boot.cfg files and I receive dial tone on remote phone, however when trying to call the remote extension The server status shows as trying to connect to extensionnumber@localipaddress instead of the public ip address.

Any ideas??

 

[mbs1]

hi,
just to add to the first post, when i make a call from a remote location to the central site, the phones rings and on pickup, the remote site can hear the central site but not the other way round. I tried this with calling an extension on the central site, a provider for fwd and voipcheap. the same in all three cases. audio only from central to remote.

calls within the central site are ok

the trace on all three cases shows the comtact 200 ok from central giving the private ip of the pbx,the remote then trying to send rtp to the private across internet. i think this may be the root of the problem.

again, if anyone has tried and got this working, would appreciate help.

maybe this is a feature that will only be available in enterprise?.

3cx support pls help.

thanks

 

[Anonymous]

NAT issue, seems to be arround for a while.

If you can try to get a STUN server on either side. if that does not work port through a VPN.

 

[archie]

This should be fixed in latest version (RC1). Check it, please.

 

[mbs1]

itfarmer and archie, thanks ever so much for your responses.

I am currently using rC1 6th March and having these issues. if you need any logs etc let me know.

thanks again

mbs

 

[archie]

itfarmer and archie, thanks ever so much for your responses.

I am currently using rC1 6th March and having these issues. if you need any logs etc let me know.

thanks again

mbs

In the extension configuration page set the check-box - 'External Phone'

 

[mbs1]

thanks archie,

i will try that and post back.

appreciate your help

mbs

 

[mbs1]

hi all,

hope all is well.

I set the external option for external phones but i still cant get audio from remote to central.I have attached the trace as performed on the pbx server.A.B.C.D is the public ip of the remote ip phone. 192.168.201.7 is the private pbx ip and 192.168.201.105 is the private ip of internal phone.

the call is from A.B.C.D to 192.168.201.105. Sorry its a bit long. From my understanding Packet 43 is from the pbx server to the public ip of remote phone giving the IP address and rtp ports that it would accept to. Lokking at this packet the pbx is giving out the private ip and not the public ip of the central site and i suspect this may be the problem. if i do a trace from the remote site i would see a lot of requests from the remote ip phone to the private ip of the pbx. off course these would never get through.
the stun settings are set on the pbx to stun.3cx.com

can i please request asssistance from all experts please


No. Time Source Destination Protocol Info
21 54.751316 A.B.C.D 192.168.201.7 SIP/SDP Request: INVITE sip:[email protected], with session description

Frame 21 (722 bytes on wire, 722 bytes captured)

Session Initiation Protocol
Request-Line: INVITE sip:[email protected] SIP/2.0
Method: INVITE
Resent Packet: False
Message Header
Via: SIP/2.0/UDP A.B.C.D:5064;branch=z9hG4bKY6U1RHsvl8t1GDf3;rport
Max-Forwards: 70
User-Agent: IP PHONE 2 V1.52.007 CFG0
From: "101" <sip:[email protected]>;tag=bpsE0S5tofiJvMgC
SIP Display info: "101"
SIP from address: sip:[email protected]
SIP tag: bpsE0S5tofiJvMgC
To: "100" <sip:[email protected]>
SIP Display info: "100"
SIP to address: sip:[email protected]
Call-ID: [email protected]
Contact: <sip:[email protected]:5064>
Contact Binding: <sip:[email protected]:5064>
URI: <sip:[email protected]:5064>
SIP contact address: sip:[email protected]:5064
CSeq: 1 INVITE
Supported: replaces
Content-Type: application/sdp
Content-Length: 243
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 13990265 81066096 IN IP4 A.B.C.D
Owner Username: -
Session ID: 13990265
Session Version: 81066096
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: A.B.C.D
Session Name (s): SIP CALL
Connection Information (c): IN IP4 A.B.C.D
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: A.B.C.D
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 8030 RTP/AVP 3 18 0 101
Media Type: audio
Media Port: 8030
Media Proto: RTP/AVP
Media Format: GSM 06.10
Media Format: ITU-T G.729
Media Format: ITU-T G.711 PCMU
Media Format: 101
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 3 GSM/8000
Media Attribute (a): rtpmap:18 G729/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 18 G729/8000
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute Fieldname: fmtp
Media Attribute Value: 101 0-15



No. Time Source Destination Protocol Info
25 55.222144 192.168.201.7 A.B.C.D SIP Status: 100 Trying

Frame 25 (323 bytes on wire, 323 bytes captured)

Session Initiation Protocol
Status-Line: SIP/2.0 100 Trying
Status-Code: 100
Resent Packet: False
Message Header
Via: SIP/2.0/UDP A.B.C.D:5064;branch=z9hG4bKQP8mUkfGDhuNaYtQ;rport=5064
To: "100"<sip:[email protected]>;tag=a6491607
SIP Display info: "100"
SIP to address: sip:[email protected]
SIP tag: a6491607
From: "101"<sip:[email protected]>;tag=bpsE0S5tofiJvMgC
SIP Display info: "101"
SIP from address: sip:[email protected]
SIP tag: bpsE0S5tofiJvMgC
Call-ID: [email protected]
CSeq: 2 INVITE
Content-Length: 0

No. Time Source Destination Protocol Info
26 55.332017 192.168.201.7 192.168.201.105 SIP/SDP Request: INVITE sip:[email protected]:5060, with session description

Frame 26 (787 bytes on wire, 787 bytes captured)

Session Initiation Protocol
Request-Line: INVITE sip:[email protected]:5060 SIP/2.0
Method: INVITE
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 192.168.201.7:5060;branch=z9hG4bK-d87543-7b33fb266e0cee4f-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
Contact Binding: <sip:[email protected]:5060>
URI: <sip:[email protected]:5060>
SIP contact address: sip:[email protected]:5060
To: "100"<sip:[email protected]:5060>
SIP Display info: "100"
SIP to address: sip:[email protected]:5060
From: "101"<sip:[email protected]:5060>;tag=11462903
SIP Display info: "101"
SIP from address: sip:[email protected]:5060
SIP tag: 11462903
Call-ID: YWQwNzUzOWJhNWM3MDUxOWIxNmQ5NTg1ZGY5NGJjZGY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Content-Type: application/sdp
Content-Length: 246
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): 3cxPS 12818015825872 12818015825873 IN IP4 192.168.201.7
Owner Username: 3cxPS
Session ID: 12818015825872
Session Version: 12818015825873
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 192.168.201.7
Session Name (s): 3cxPS Audio call
Connection Information (c): IN IP4 192.168.201.7
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 192.168.201.7
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 8034 RTP/AVP 0 8 3 98
Media Type: audio
Media Port: 8034
Media Proto: RTP/AVP
Media Format: ITU-T G.711 PCMU
Media Format: ITU-T G.711 PCMA
Media Format: GSM 06.10
Media Format: 98
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 0 PCMU/8000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 8 PCMA/8000
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 3 GSM/8000
Media Attribute (a): rtpmap:98 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 98 telephone-event/8000

No. Time Source Destination Protocol Info
27 55.444866 192.168.201.105 192.168.201.7 SIP Status: 180 Ringing

Frame 27 (472 bytes on wire, 472 bytes captured)

Session Initiation Protocol
Status-Line: SIP/2.0 180 Ringing
Status-Code: 180
Resent Packet: False
Message Header
From: "101"<sip:[email protected]:5060>;tag=11462903
SIP Display info: "101"
SIP from address: sip:[email protected]:5060
SIP tag: 11462903
To: "100"<sip:[email protected]:5060>;tag=c0a8c969-13c4-45f2d558-269fe9-5798
SIP Display info: "100"
SIP to address: sip:[email protected]:5060
SIP tag: c0a8c969-13c4-45f2d558-269fe9-5798
Call-ID: YWQwNzUzOWJhNWM3MDUxOWIxNmQ5NTg1ZGY5NGJjZGY.
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.201.7:5060;rport=5060;branch=z9hG4bK-d87543-7b33fb266e0cee4f-1--d87543-
Supported: replaces
User-Agent: SIP Phone
Contact: <sip:[email protected]:5060>
Contact Binding: <sip:[email protected]:5060>
URI: <sip:[email protected]:5060>
SIP contact address: sip:[email protected]:5060
Content-Length: 0

No. Time Source Destination Protocol Info
28 55.550222 192.168.201.7 A.B.C.D SIP Status: 180 Ringing

Frame 28 (368 bytes on wire, 368 bytes captured)

Session Initiation Protocol
Status-Line: SIP/2.0 180 Ringing
Status-Code: 180
Resent Packet: False
Message Header
Via: SIP/2.0/UDP A.B.C.D:5064;branch=z9hG4bKQP8mUkfGDhuNaYtQ;rport=5064
Contact: "101"<sip:[email protected]:5060>
Contact Binding: "101"<sip:[email protected]:5060>
URI: "101"<sip:[email protected]:5060>
SIP Display info: "101"
SIP contact address: sip:[email protected]:5060
To: "100"<sip:[email protected]>;tag=a6491607
SIP Display info: "100"
SIP to address: sip:[email protected]
SIP tag: a6491607
From: "101"<sip:[email protected]>;tag=bpsE0S5tofiJvMgC
SIP Display info: "101"
SIP from address: sip:[email protected]
SIP tag: bpsE0S5tofiJvMgC
Call-ID: [email protected]
CSeq: 2 INVITE
Content-Length: 0

No. Time Source Destination Protocol Info
29 59.604918 192.168.201.105 192.168.201.7 SIP/SDP Status: 200 OK, with session description

Frame 29 (727 bytes on wire, 727 bytes captured)

Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Status-Code: 200
Resent Packet: False
Message Header
From: "101"<sip:[email protected]:5060>;tag=11462903
SIP Display info: "101"
SIP from address: sip:[email protected]:5060
SIP tag: 11462903
To: "100"<sip:[email protected]:5060>;tag=c0a8c969-13c4-45f2d558-269fe9-5798
SIP Display info: "100"
SIP to address: sip:[email protected]:5060
SIP tag: c0a8c969-13c4-45f2d558-269fe9-5798
Call-ID: YWQwNzUzOWJhNWM3MDUxOWIxNmQ5NTg1ZGY5NGJjZGY.
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.201.7:5060;rport=5060;branch=z9hG4bK-d87543-7b33fb266e0cee4f-1--d87543-
Supported: replaces
User-Agent: SIP Phone
Contact: <sip:[email protected]:5060>
Contact Binding: <sip:[email protected]:5060>
URI: <sip:[email protected]:5060>
SIP contact address: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 227
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): SIP302 100 0 IN IP4 192.168.201.105
Owner Username: SIP302
Session ID: 100
Session Version: 0
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 192.168.201.105
Session Name (s): Audio Session
Session Information (i): Audio Session
Connection Information (c): IN IP4 192.168.201.105
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 192.168.201.105
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 8000 RTP/AVP 0 101
Media Type: audio
Media Port: 8000
Media Proto: RTP/AVP
Media Format: ITU-T G.711 PCMU
Media Format: 101
Media Attribute (a): ptime:20
Media Attribute Fieldname: ptime
Media Attribute Value: 20
Media Attribute (a): fmtp:101 0-11
Media Attribute Fieldname: fmtp
Media Attribute Value: 101 0-11
Media Attribute (a): rtpmap:0 PCMU/8000/1
Media Attribute Fieldname: rtpmap
Media Attribute Value: 0 PCMU/8000/1
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 101 telephone-event/8000

No. Time Source Destination Protocol Info
43 59.707702 192.168.201.7 A.B.C.D SIP/SDP Status: 200 OK, with session description

Frame 43 (702 bytes on wire, 702 bytes captured)

Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Status-Code: 200
Resent Packet: False
Message Header
Via: SIP/2.0/UDP A.B.C.D:5064;branch=z9hG4bKQP8mUkfGDhuNaYtQ;rport=5064
Contact: "101"<sip:[email protected]:5060>
Contact Binding: "101"<sip:[email protected]:5060>
URI: "101"<sip:[email protected]:5060>
SIP Display info: "101"
SIP contact address: sip:[email protected]:5060
To: "100"<sip:[email protected]>;tag=a6491607
SIP Display info: "100"
SIP to address: sip:[email protected]
SIP tag: a6491607
From: "101"<sip:[email protected]>;tag=bpsE0S5tofiJvMgC
SIP Display info: "101"
SIP from address: sip:[email protected]
SIP tag: bpsE0S5tofiJvMgC
Call-ID: [email protected]
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Content-Type: application/sdp
Content-Length: 222
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): 3cxPS 12818015825826 12818015825828 IN IP4 192.168.201.7
Owner Username: 3cxPS
Session ID: 12818015825826
Session Version: 12818015825828
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 192.168.201.7
Session Name (s): 3cxPS Audio call
Connection Information (c): IN IP4 192.168.201.7
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 192.168.201.7
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 9106 RTP/AVP 3 0 98
Media Type: audio
Media Port: 9106
Media Proto: RTP/AVP
Media Format: GSM 06.10
Media Format: ITU-T G.711 PCMU
Media Format: 98
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 3 GSM/8000
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 0 PCMU/8000
Media Attribute (a): rtpmap:98 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 98 telephone-event/8000


thanks very much

mbs

 

[mbs1]

hi all

Just to add to the above.

I checked the trace on an outbound call from a ipphone on the same subnet as the pbx to a remote Ipiphone.

the trace shows the invite going from the pbx to the remote ipiphone with the connection ip address as the local private ip address of the pbx server.
maybe this could aslo explain why there is no audio from the remote to the central end.

pls help

thanks

mbs

 

[archie]

Please, check your logs if STUN client was able to resolve your external IP address.

 

[mbs1]

hi archie,

thanks for getting back.

I have noticed that on the pbx server status, i only get the External IPaddress resolved correctly if i restart the main service. Otherwise i can see frequent stun client requests but no responses.

let me get a trace

thanks again, help appreciated

mbs

 

[mbs1]

hi

i have been checking the server status. i can see a message

stunclient Send Initial Stun request to 194.221.62.209 but there is no response on the log to this request.

A trace on the pbx pc shows a
stun binding request to 194.221.62.209
Stun binding response 194.221.62.209

the only time i get a response to the stun client request is when the server get booted or i restart the pbx service.

i have got udp 3478 and 3479 allowed out

sorry this query is dragging on.

appreciate your help

mbs

 

[mbs1]

hi archie,

i saw in one of the posts that a newer version will have introduced new advanced options, which allows you to bind your device to our Media Server. It should solve one-way audio problem.

This may cure my problem as well.

On the general settings, are the stun settings changeable. I can enter a different stun server but if i restart the service, it goes back to stun.3cx.com. if i dont restart, when i logon to the admin screen, it is reset to stun.3cx.com.

also the log screen shows scheduled intial requests to the ip address of stun.3cx.com but no response. i only get a response if i restart the service.is this normal.

sorry asking a lot of questions.

I think i am almost there and if this gets solved many will benefit

thanks

mbs

 

[mbs1]

hi

i saw in one of the posts a reference to a registry key msexternalinterface.

it is currently set to the private ip address of my pbx server. in this config, a central ipphone can make a call to a voip gateway configured for fwd and as a test the echo test works. the ipphone is configured to use the pbx as a outbound proxy as well. i can see from the trace that the pbx acts as a proxy between the ipphone and the fwd gateway. not phone and pb on same subnet.

However in this configuration, the internal ipphone can dial a remote ipphone registered to the pbx ( external option set ) but audio is only one way ( central to out not the other way round). i think the reason for this is the pbx giving its contact ip address as the private internal one to the remote end , trace given in previous post. Again i can see the pbx acting as a proxy between the central ipphone and the remote one.

IF i change the value of msexternalinterface to the public ipaddress of the firewall the the following happens. The remote ipphone can call the fwd gateway and the echo test works. 2 remote ipphone can call each other and there is 2 way audio. Note the remote phones are configured to use stun. from the pbx trace i can see that the pbx is not really acting as a proxy, it tells both ends what rtp ports are in use and the rtp comms is then direct between the endppoints. Success?.......... only partially as this configuration stops the central ipphone on same subnet as the pbx from receiving audio from the fwd gateway. As the pbx is not acting as a proxy it gives the rtp ports to both ends. what is stopping the audio is the config of the firewall which allows inbound rtp to the pbx only. if i allow the relevant rtp port inbound to the ipphone then it works. So i suppose for a site with a small nos of central ipphones the firewall could be configured to allow different inbound rtp to different inbound ports and the phones would have to be configured to use these ports.


Maybe the new version may fix this.

can anyone help, please

credit to the superuser who shared the registry key info as it helped me understand a bit better that there is a real good chance that this may just work.

thanks

mbs

 

[archie]

I have noticed that on the pbx server status, i only get the External IPaddress resolved correctly if i restart the main service. Otherwise i can see frequent stun client requests but no responses.

mbs

Hi,

It is normal behavior. You will get message about External IP resolved, only for first time, or after currently resolved addressort doesn't match newly resolved ones. That is why you see periodic requests, but no message about responses, because they match previous responces.

 

[mbs1]

thanks archie,

any idea on the contact ip being set to the private ip of the pbx or the msexternalinterface setting determining the mode od operation of the\ proxy.

am i right in thinking that the pbx should proxy the rtp for all calls ?

i appreciate your help.

thanks

mbs

 

[archie]

any idea on the contact ip being set to the private ip of the pbx or the msexternalinterface setting determining the mode od operation of the\ proxy.

am i right in thinking that the pbx should proxy the rtp for all calls ?

What exactly contact IP do you mean? The address in 'Contact' field of outgoing INVITE should be your external IP resolved by STUN for any endpoint that is marked as external. Though, there's a bug in RC1 due to which, for extensions marked as external, the RTP (receiving) address is local IP or the IP specified in msexternalinterface, but not STUN-resolved one. We've fixed it in upcoming RC2.

PBX will proxy RTP if one of endpoints is marked as an external. Otherwise, it will try to establish direct connection, unless one of endpoints is IVR, Voice Mailbox, or is placed on-hold.

 

[mbs1]

Hi archie.

Thanks Very much for our reply.

by contact ip i mean the connection ip address sent to the external endpoint for it to send rtp packets to. i now understand that RC2 will resolve this issue.

Thanks very much, with your help i have learnt a lot.

regards

mbs

 

[dekatech]

Been following this thread, as I have the exact same issues.

Can someine calrify for me.

Does this mean that in RC2 I will be able to have a phone register and communicate with my 3CX server from a remote location outside of my LAN?

If so,
Understanding the bug issue in RC1, if I set my msexternalinterface to the external ip (WAN), should'nt I be able to check the external extension box and make phone calls from the remote location through my 3CX server?

I have been successful at getting the external extension to make calls using my provider to outbound numbers, but extension to extension dialing is not working....

Travis