Port 5060: Not Listening

Good day,

Asterisk 1.8.5.0
Freepbx 2.9.0.7
X-lite 4

Since I upgraded to asterisk 1.8.5.0, I cannot connect the SIP phones to the server. I checked the port connections on the server but it seems that port 5060 is not listening.

From the client, I get a timeout error. Here are the logs from X-lite 4 softphone:
[11-07-18]13:38:10.195 | Debug | CCM | “Re-trying to REGISTER[URI:1003@192.168.0.72]” | sua::CSIPRegistrationWatcher::OnTimer
[11-07-18]13:38:10.195 | Debug | CCM | “[URI:1003@192.168.0.72]” | sua::CSIPRegistration::Start
[11-07-18]13:38:10.195 | Debug | Resip | "RESIP:DUM:BaseCreator::makeInitialRequest:

16C9D870" |
[11-07-18]13:38:10.196 | Debug | Resip | “RESIP:DUM:RegistrationCreator::RegistrationCreator: 16C9D870” |
[11-07-18]13:38:10.196 | Debug | Resip | “RESIP:DUM: ************* Created DialogSet(UAC) – Njc4ZTA0OTFmZjM4ZWY2YmM1YTg3YjVhMmZlOTU2YjI.-d857e095*************” |
[11-07-18]13:38:10.196 | Debug | Resip | "RESIP:DUM:SEND:

REGISTER sip:192.168.0.72 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-1d7826def8ed2df0-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:1003;rinstance=5a43e8240ab733c1
To: "“Ben”"sip:1003@192.168.0.72
From: "“Ben”"sip:1003@192.168.0.72;tag=d857e095
Call-ID: Njc4ZTA0OTFmZjM4ZWY2YmM1YTg3YjVhMmZlOTU2YjI.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0

" |
[11-07-18]13:38:10.196 | Debug | Resip | "RESIP:TRANSACTION:Adding application timer: " |
[11-07-18]13:38:10.196 | Info | Resip | “RESIP:DUM:Got a DumFeatureMessage16CD28C0” |
[11-07-18]13:38:10.196 | Debug | Resip | “RESIP:DUM:has obp” |
[11-07-18]13:38:10.196 | Debug | Resip | “RESIP:DUM:Next hop is 192.168.0.72” |
[11-07-18]13:38:10.196 | Debug | Resip | “RESIP:DUM:DialogId::DialogId: Njc4ZTA0OTFmZjM4ZWY2YmM1YTg3YjVhMmZlOTU2YjI.-d857e095-” |
[11-07-18]13:38:10.196 | Debug | Resip | “RESIP:DUM:Using outbound proxy: sip:1003@192.168.0.72;lr -> SipReq: REGISTER 192.168.0.72 tid=1d7826def8ed2df0 cseq=REGISTER contact=1003 / 1 from(tu)” |
[11-07-18]13:38:10.197 | Debug | Resip | “RESIP:TRANSACTION:Adding timer: Timer F tid=1d7826def8ed2df0 ms=32000” |
[11-07-18]13:38:10.197 | Debug | Resip | “RESIP:DNS:DnsResult::lookup sip:1003@192.168.0.72;lr” |
[11-07-18]13:38:10.197 | Debug | Resip | “RESIP:DNS:Numeric result so return immediately: [ V4 192.168.0.72:5060 TCP target domain=192.168.0.72 mFlowKey=0 ]” |
[11-07-18]13:38:10.197 | Debug | Resip | “RESIP:TRANSPORT:Could not find a connection for [ V4 192.168.0.72:5060 TCP target domain=192.168.0.72 mFlowKey=0 ]” |
[11-07-18]13:38:10.201 | Debug | Resip | “RESIP:TRANSPORT:Best Route - subnet=192.168.0.64 net-mask=255.255.255.192 next-hop=0.0.0.0 if-index=11” |
[11-07-18]13:38:10.201 | Debug | Resip | “RESIP:TRANSPORT:IP Table entry 1/3 if-index=10 NIC IP=192.168.229.103 NIC Mask=255.255.255.192” |
[11-07-18]13:38:10.201 | Debug | Resip | “RESIP:TRANSPORT:IP Table entry 2/3 if-index=1 NIC IP=127.0.0.1 NIC Mask=255.0.0.0” |
[11-07-18]13:38:10.202 | Debug | Resip | “RESIP:TRANSPORT:IP Table entry 3/3 if-index=11 NIC IP=192.168.0.73 NIC Mask=255.255.255.192” |
[11-07-18]13:38:10.202 | Debug | Resip | “RESIP:TRANSPORT:Looked up source for destination: [ V4 192.168.0.72:5060 TCP target domain=192.168.0.72 mFlowKey=0 ] -> [ V4 192.168.0.73:0 TCP target domain=192.168.0.72 mFlowKey=0 ] sent-by= sent-port=0” |
[11-07-18]13:38:10.202 | Debug | Resip | “RESIP:TRANSPORT:findTransportBySource([ V4 192.168.0.73:0 TCP target domain=192.168.0.72 mFlowKey=0 ])” |
[11-07-18]13:38:10.202 | Debug | Resip | “RESIP:TRANSPORT:findTransport (any port, any interface) => Transport: [ V4 0.0.0.0:13771 TCP target domain=unspecified mFlowKey=0 ]” |
[11-07-18]13:38:10.202 | Debug | Resip | "RESIP:TRANSPORT:Transmitting to [ V4 192.168.0.72:5060 TCP target domain=192.168.0.72 mFlowKey=0 ] tlsDomain= via [ V4 192.168.0.73:13771 TCP target domain=192.168.0.72 mFlowKey=0 ]

REGISTER sip:192.168.0.72 SIP/2.0

Any suggestions?

HI!

The port 5060 should be a UDP port, and the UDP ports don’t listen, only the TCP ports listen. Make shure the extensions have the 5060 as port, and use the asterisk sip settings tool from the freepbx, if is not present on your tools menu, installit from the updates.

Hope this helps. Regards.

Even though they don’t use a listen system call, UDP ports will still show up on netstat -a.