- Joined
- Dec 31, 2015
- Messages
- 7
- Reaction score
- 0
Hi,
I have a phone handset which I have taken to a remote location, not connected at all to our office where 3CX is running.
I have the 3CX app running externally with no issues, however when I take one of our physical handsets to a remote location, I can make and receive calls, but get no audio. Below is the log file of the event of an outbound call from an extension at the remote site to an extension in the office where the PBX is... I can see what the problem is, with regards to the IP address of SIP and RTP being different, but I am not sure how to overcome it.
Help would be greatly appreciated.
Nathan
-------------------------------------------------------
09-Jan-2016 06:33:03.597 [CM504001]: Endpoint Extn:100: new contact is registered. Contact(s): [sip:100@PUBLIC IP OF REMOTE SITE:39300 / 100]
09-Jan-2016 06:33:01.663 [CM504002]: Endpoint Extn:100: a contact is unregistered. Contact(s): []
09-Jan-2016 06:30:56.268 Leg L:75.2[VMail] is terminated: Cause: BYE from PBX
09-Jan-2016 06:30:56.267 [CM503008]: Call(C:75): Call is terminated
09-Jan-2016 06:30:56.266 Leg L:75.1[Extn] is terminated: Cause: BYE from "PUBLIC IP OF REMOTE SITE":39300
09-Jan-2016 06:30:53.652 Currently active calls - 1: [75]
09-Jan-2016 06:30:51.455 [CM503007]: Call(C:75): VMail:999 has joined, contact <sip:[email protected]:5483>
09-Jan-2016 06:30:51.453 [CM503007]: Call(C:75): Extn:100 has joined, contact <sip:100@PUBLIC IP OF REMOTE SITE:39300>
09-Jan-2016 06:30:51.452 L:75.2[VMail] has joined to L:75.1[Extn]
09-Jan-2016 06:30:51.451 NAT/ALG check:L:75.2[VMail] RESPONSE 200 on 'INVITE' - basic check passed. No information for extended checks
09-Jan-2016 06:30:51.300 [CM503025]: Call(C:75): Calling T:VMail:999@[Dev:sip:[email protected]:5483;rinstance=bc4a7b8e1363726f] for L:75.1[Extn]
09-Jan-2016 06:30:51.259 [CM503005]: Call(C:75): Forwarding: T:VMail:999@[Dev:sip:[email protected]:5483;rinstance=bc4a7b8e1363726f]
09-Jan-2016 06:30:51.259 L:75.1[Extn] forwards call from Extn:102 to VMail:999 based on rule Fwd[Available/NotReg]
09-Jan-2016 06:30:51.259 L:75.1[Extn] failed to reach Extn:102, reason Not Registered
09-Jan-2016 06:30:51.259 [CM503017]: Call(C:75): Target is not registered: Extn:102
09-Jan-2016 06:30:51.257 [CM503001]: Call(C:75): Incoming call from Extn:100 to <sip:102@OFFICE PBX ADDRESS:5060>
09-Jan-2016 06:30:51.257 NAT/ALG check:L:75.1[Extn] REQUEST 'INVITE' - some of SIP/SDP headers may contain inconsistent information or modified by intermediate hop
'audio' media IP is not equal to the IP specified in contact header:
'audio' media IP:192.168.1.23
Contact IP
UBLIC IP OF REMOTE SITE
'audio' media IP is not equal to the SIP packet source(IP
ort):
'audio' media IP:192.168.1.23
Received from
UBLIC IP OF REMOTE SITE
09-Jan-2016 06:30:51.256 [MS003005] C:75.1: Failed to create Endpoint: (destination=PUBLIC IP OF REMOTE SITE)
EndPoint: [email protected]:5482(EXTERNAL)
LOGID=C:75.1 Status: MSEP_FAILED
RTP
BX PUBLIC ADDRESS:9104
RTCP
BX PUBLIC ADDRESS:9105
STUN RTP:0.0.0.0:0
STUN RTCP:0.0.0.0:0
Coder:
NOT SET
101:telephony-event
Party ptime:20
Party RTP:0.0.0.0:0
Party RTCP:0.0.0.0:0
Decoders:
<empty>
I have a phone handset which I have taken to a remote location, not connected at all to our office where 3CX is running.
I have the 3CX app running externally with no issues, however when I take one of our physical handsets to a remote location, I can make and receive calls, but get no audio. Below is the log file of the event of an outbound call from an extension at the remote site to an extension in the office where the PBX is... I can see what the problem is, with regards to the IP address of SIP and RTP being different, but I am not sure how to overcome it.
Help would be greatly appreciated.
Nathan
-------------------------------------------------------
09-Jan-2016 06:33:03.597 [CM504001]: Endpoint Extn:100: new contact is registered. Contact(s): [sip:100@PUBLIC IP OF REMOTE SITE:39300 / 100]
09-Jan-2016 06:33:01.663 [CM504002]: Endpoint Extn:100: a contact is unregistered. Contact(s): []
09-Jan-2016 06:30:56.268 Leg L:75.2[VMail] is terminated: Cause: BYE from PBX
09-Jan-2016 06:30:56.267 [CM503008]: Call(C:75): Call is terminated
09-Jan-2016 06:30:56.266 Leg L:75.1[Extn] is terminated: Cause: BYE from "PUBLIC IP OF REMOTE SITE":39300
09-Jan-2016 06:30:53.652 Currently active calls - 1: [75]
09-Jan-2016 06:30:51.455 [CM503007]: Call(C:75): VMail:999 has joined, contact <sip:[email protected]:5483>
09-Jan-2016 06:30:51.453 [CM503007]: Call(C:75): Extn:100 has joined, contact <sip:100@PUBLIC IP OF REMOTE SITE:39300>
09-Jan-2016 06:30:51.452 L:75.2[VMail] has joined to L:75.1[Extn]
09-Jan-2016 06:30:51.451 NAT/ALG check:L:75.2[VMail] RESPONSE 200 on 'INVITE' - basic check passed. No information for extended checks
09-Jan-2016 06:30:51.300 [CM503025]: Call(C:75): Calling T:VMail:999@[Dev:sip:[email protected]:5483;rinstance=bc4a7b8e1363726f] for L:75.1[Extn]
09-Jan-2016 06:30:51.259 [CM503005]: Call(C:75): Forwarding: T:VMail:999@[Dev:sip:[email protected]:5483;rinstance=bc4a7b8e1363726f]
09-Jan-2016 06:30:51.259 L:75.1[Extn] forwards call from Extn:102 to VMail:999 based on rule Fwd[Available/NotReg]
09-Jan-2016 06:30:51.259 L:75.1[Extn] failed to reach Extn:102, reason Not Registered
09-Jan-2016 06:30:51.259 [CM503017]: Call(C:75): Target is not registered: Extn:102
09-Jan-2016 06:30:51.257 [CM503001]: Call(C:75): Incoming call from Extn:100 to <sip:102@OFFICE PBX ADDRESS:5060>
09-Jan-2016 06:30:51.257 NAT/ALG check:L:75.1[Extn] REQUEST 'INVITE' - some of SIP/SDP headers may contain inconsistent information or modified by intermediate hop
'audio' media IP is not equal to the IP specified in contact header:
'audio' media IP:192.168.1.23
Contact IP
'audio' media IP is not equal to the SIP packet source(IP
'audio' media IP:192.168.1.23
Received from
09-Jan-2016 06:30:51.256 [MS003005] C:75.1: Failed to create Endpoint: (destination=PUBLIC IP OF REMOTE SITE)
EndPoint: [email protected]:5482(EXTERNAL)
LOGID=C:75.1 Status: MSEP_FAILED
RTP
RTCP
STUN RTP:0.0.0.0:0
STUN RTCP:0.0.0.0:0
Coder:
NOT SET
101:telephony-event
Party ptime:20
Party RTP:0.0.0.0:0
Party RTCP:0.0.0.0:0
Decoders:
<empty>