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[motty.cruz at gmail.com]

Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled

[1006]
type=friend
username=1006
secret=mysecret
context=sip-phone
call-limit=1
callerid="iuser" <1006>
disallow=all
host=dynamic
allow=all

any ideas?

Thanks,
Motty

[mefhigoseth at gmail.com]

Hi Motty,
Please, set  Verbose  to 3 and Debug to 3 At Asterisk CLI. Then "sip set debug on".
Now try to register again. At last, " sip  de debug off".
Examine tour console  or  full log file to find some clue ir send me back some trace.
Cheers.
El oct. 13, 2016 1:45 PM, "Motty Cruz" <motty.cruz@gmail.com (motty.cruz@gmail.com)> escribió:

[motty.cruz at gmail.com]

Hello Victor,
I did set debug on, but I don’t see any errors. I did tcpdump, client is trying to register:  here is the header of a udp packet

User Datagram Protocol, Src Port: 55300, Dst Port: 5060
Session Initiation Protocol (REGISTER)
    Request-Line: REGISTER sip:pbx.mydomain.com:5060 SIP/2.0
        Method: REGISTER
        Request-URI: sip:pbx.mydomain.com:5060
        [Resent Packet: True]
        [Suspected resend of frame: 14]
    Message Header
        Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK7b2855394DB988BE
            Transport: UDP
            Sent-by Address: 192.168.1.37
            Sent-by port: 5060
            Branch: z9hG4bK7b2855394DB988BE
        From: "1006" <sip:1006@pbx.mydomain.com>;tag=2859342B-CBC71460
            SIP Display info: "1006"
            SIP from address: sip:1006@pbx.mydomain.com
            SIP from tag: 2859342B-CBC71460
        To: <sip:1006@pbx.mydomain.com>
            SIP to address: sip:1006@pbx.mydomain.com
                SIP to address User Part: 1006
                SIP to address Host Part: pbx.mydomain.com
        CSeq: 1 REGISTER
        Call-ID: 6cbe37bb-cca69d70-85d0431d@192.168.1.37
        Contact: <sip:1006@192.168.1.37:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"
        User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.10.0689
        Accept-Language: en
        Max-Forwards: 70
        Expires: 90
        Content-Length: 0

Sip.conf
[1006]
type=friend
username=1006
secret=mysecret
context=sip-phone
call-limit=5
callerid="iuser" <1006>
disallow=all
host=dynamic
allow=all
nat=yes

Is NAT value set to yes OK? Servers is on public IP, client is on private network.

Thanks,
Motty

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Victor Villarreal
Sent: Thursday, October 13, 2016 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit


Hi Motty,
Please, set Verbose to 3 and Debug to 3 At Asterisk CLI. Then "sip set debug on".
Now try to register again. At last, " sip de debug off".
Examine tour console or full log file to find some clue ir send me back some trace.
Cheers.

El oct. 13, 2016 1:45 PM, "Motty Cruz" <motty.cruz@gmail.com (motty.cruz@gmail.com)> escribió:
Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled

[1006]
type=friend
username=1006
secret=mysecret
context=sip-phone
call-limit=1
callerid="iuser" <1006>
disallow=all
host=dynamic
allow=all

any ideas?

Thanks,
Motty



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[mefhigoseth at gmail.com]

Ok.
Please, note that 192.168.1.37 (I suspect) is the internal  LAN address Of the Polycom hardphone. If this is true, then you have  NAT issues.
The REGISTER message are received by your PBX, but when respond, Asterisk send the next SIP message to the IP informed by the phone, that is the internal LAN address. The messages do not reach back to the hardphone.
You need to setup a STUN server in the Polycom hardphone settings. Please, check the manual. Search in Google some public  STUN server to put in the settings.
Last, the idea behind the "sip set debug" command was view the complete SIP messages conversation, not search for an error.
On NAT escenarios, remember:
* The NATed phones need to know the public  IP of the NATing router. Either by manual setting  or  by STUN protocol.
* Reduce the time between REGISTERs attempt, if the client  have a dynamic IP connection.
* Use the "localnet" SIP settings in Asterisk, so the PBX can distingish what Network need contacted via NAT and what not.
Cheers.

[motty.cruz at gmail.com]

Thank you for your help! Centos 7 firewall was enable.