Good day,
After upgrading to Asterisk 1.8.5.0, The sip connections are not working anymore. When I try to connect from the softphone, I would get a request timeout error.
I checked on the server and it appears that port 5060 is not listening. How do I start the port?
I don’t have a firewall running, and phones could connect before the upgrade.
Any suggestions?
System
Asterisk 1.8.5.0
Freepbx 2.9.0.7
Rhino PCI E1 card (Dahdi)
I asterisk started?
LOL, Yes it is.
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:[email protected]]” | sua::CSIPRegistrationWatcher::OnTimer
[11-07-18]13:38:10.195 | Debug | CCM | “[URI:[email protected]]” | 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:[email protected]
From: "“Ben”"sip:[email protected];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:[email protected];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:[email protected];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 ]
Best guess is that you are using TCP as transport on X-lite and UDP on Asterisk.
Transport settings on X-lite are set to automatic and on the extension is set to UDP only.
Another thing, on the freepbx dashboard under Freepbx Connections in the statistics box the bar that shows connected extensions is not visible.
Sip module is not loaded.
When I enter ‘module show like sip’, I receive “0 modules loaded message”.
And when I try to load the module, I get a “module load chan_sip.so: failed”
How do I proceed?
amportal kill
Rename file /var/log/asterisk/full to something else.
amportal start
Now go through the log file to see why it does not load sip.
Or…
Install FreePBX Distro.
Backup FreePBX first.
I renamed the log file but a new one was not created.
So i decided to reinstall freepbx from a distro. Now i get text in the background on the freepbx web page and the following notifications.
retrieve_conf failed, config not applied
Reload failed because retrieve_conf encountered an error: 255
Added 20 minutes ago
(freepbx.RCONFFAIL)
Notice: Deprecated Directory used by 1 IVRs more…
I decided to uninstall asterisk and freepbx completly. Here is how I did it.
In asterisk source directory
make uninstall-all
Uninstalling freepbx
rm -rf /var/www/html [if there are no other websites]
And I installed asterisk18 and freepbx from distribution.
yum -y install asterisk18 asterisk18-core asterisk18-configs asterisk18-dahdi asterisk18-doc asterisk18-odbc asterisk18-res_fax_digium asterisk-voicemail
yum -y install freepbx*
All is ok now, but I cannot get the trunk to work. I cannot receive nor make outbound calls.
Got it to work as well.
I had to include the dahdi-channels.conf file in chan_dahdi.conf file at the end like this.
#include dahdi-channels.conf
Thank you Mikael for assistance. Now off to get the fax service to work.