asterisk v13 compatibility

Asked by tomislav.m.zagreb on 2018-02-20

Hi Adam,

I copied the question from mailing list about compatibility with asterisk v13:

I have problem with quexs v1.15.2 and 1.15.10 and asterisk v13.19. After
login to web interface starting call (nothing happens) there is a line i
asterisk full log:

WARNING[15001] chan_sip.c: Purely numeric hostname (67331), and not a
peer--rejecting!

thanks
Tom

Question information

Language:
English Edit question
Status:
Solved
For:
queXS Edit question
Assignee:
No assignee Edit question
Solved by:
tomislav.m.zagreb
Solved:
2018-03-01
Last query:
2018-03-01
Last reply:
2018-03-01
Adam Zammit (adamzammit) said : #1

Hi Tom,

I need a bit more information to help sort this out.

Are you running the "VoIP monitoring" process? Does it report any errors?

Have you had previous versions of queXS / Asterisk working before?

When you defined your extensions in queXS did you include the prefix of "SIP/" or "IAX/"?

Adam

Hi Adam,

we have two quexs/asterisk servers. One in prod, asterisk 11 quexs 15.2, everything works ok. Second server is for testing, centos 7 quexs 15.2 asterisk 13.19 (freepbx SNG7-PBX-64bit-1712-2). Quexs on new server is working ok if we switch to asterisk 11 on prod server trough config.inc.local.php

No, we are not running voip monitor. Extension is with SIP/

Tom

Adam Zammit (adamzammit) said : #3

Can you confirm the manager.conf settings in asterisk 13 are the same/similar to the asterisk 11 server (especially read and write permissions, must have originate set)?

Are you using freepbx or just asterisk?

Also please confirm that the "ORIGINATE_CONTEXT" setting in your config.inc.local.php file matches the Asterisk context for outbound dialing.

yes it is, manager.conf v13:
permit=127.0.0.1/255.255.255.0
permit=192.168.0.0/255.255.255.0
permit=10.9.0.0/255.255.255.0
read = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
writetimeout = 5000

 manager.conf v11:
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
permit=192.168.0.0/255.255.255.0
permit=10.9.0.0/255.255.255.0
read = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
writetimeout = 5000

We are using freepbx: iso: freepbx SNG7-PBX-64bit-1712-2

Yes, originate_context is default, from-internal:
 [2018-02-22 23:52:49] VERBOSE[26355][C-00000001] pbx.c: Executing [91xxxxx@from-internal:1] Macro("PJSIP/67331-00000001", "user-callerid,LIMIT,EXTERNAL,") in new stack

Adam Zammit (adamzammit) said : #5

Ok I think you should try and test the AMI directly - using a "telnet" command - connect to the Asterisk AMI:

telnet IP-OF-ASTERISK-SERVER 5038

Then enter:

Action: Login
Username: astuser
Secret: astpassword
Events: off

Action: Originate
Channel: 1234
Exten: 5678
Context: from-internal
Priority: 1
Callerid: 1234

Where 1234 is the extension to dial from
And 5678 is the number to dial.

Please see if that returns any errors

You could try changing the Context of the originate command if it fails to "default"

Adam

call from pjsip/67331 extension to my cellphone 91333xxxx, everything ok, cell phone rang.
Does anyone else have problem with asterisk v13 ? thanks Adam for the effort

xxxx@xxxx~> telnet localhost 5038
Trying ::1...
telnet: connect to address ::1: Connection refused
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/2.10.3

Action:Login
ActionID:1
Username: admin
Secret:xxxx
events:offResponse: Error
Message: Missing action in request

Response: Success
ActionID: 1
Message: Authentication accepted

Action:Originate
Channel:PJSIP/67331
Exten:91333xxxx
Context:from-internal
Priority:1Response: Error
Message: Missing action in request

Response: Success
Message: Originate successfully queued

full:
...
[2018-02-28 22:30:00] VERBOSE[1429][C-0000000c] app_dial.c: Called IAX2/908/091333xxxx
[2018-02-28 22:30:00] VERBOSE[2196][C-0000000c] chan_iax2.c: Call accepted by 192.168.0.10:4569 (format gsm)
[2018-02-28 22:30:00] VERBOSE[2196][C-0000000c] chan_iax2.c: Format for call is (gsm)
[2018-02-28 22:30:01] VERBOSE[1429][C-0000000c] app_dial.c: IAX2/908-692 is making progress passing it to PJSIP/67331-00000016
[2018-02-28 22:30:17] VERBOSE[1429][C-0000000c] app_dial.c: IAX2/908-692 answered PJSIP/67331-00000016
...

Adam Zammit (adamzammit) said : #7

These are the commands that queXS should be sending to your asterisk server.

Please just confirm the extension has been entered as PJSIP/67331 in queXS and that it is reflected as that in the quexs database (extensions table).

Otherwise I can't see why it wouldn't work.

Adam

Hi Adam,

that's it, sip instead of pjsip in quexs/admin panel, my bad. We also had to expand structure for ext from char(10) to ... ext:pjsip/67331.

thanks Adam again.

Tom

Adam Zammit (adamzammit) said : #9

Thanks Tom,

I'll increase the width of that column in the next release of queXS.

Adam