Motty Cruz
2016-Oct-13 20:15 UTC
[asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit
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 at
pbx.mydomain.com>;tag=2859342B-CBC71460
SIP Display info: "1006"
SIP from address: sip:1006 at pbx.mydomain.com
SIP from tag: 2859342B-CBC71460
To: <sip:1006 at pbx.mydomain.com>
SIP to address: sip:1006 at 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 at 192.168.1.37
Contact: <sip:1006 at 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 at lists.digium.com [mailto:asterisk-users-bounces
at 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 at gmail.com>
escribi?:
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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Victor Villarreal
2016-Oct-13 22:54 UTC
[asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit
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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20161013/0a2af6d7/attachment.html>
motty cruz
2016-Oct-16 03:38 UTC
[asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit
Thank you for your help! Centos 7 firewall was enable. systemctl stop firewalld issue fixed. Thanks, On Thu, Oct 13, 2016 at 3:54 PM, Victor Villarreal <mefhigoseth at gmail.com> wrote:> 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. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 > http://www.asterisk.org/community/astricon-user-conference > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Thanks for your support, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20161015/c69965d3/attachment.html>