Asterisk send register to my softphones

dendene · October 20, 2017, 5:13pm

HI;

i have made an asterisk server, when my softphones connect to it. asterisk sends REGISTERs periodically (each 18 seconds approx)

i have these messages into the console:

chan_sip.c:16029 transmit_register: Probably a DNS error for registration to edi-1@, trying REGISTER again (after 20 seconds)
[Oct 20 18:59:54] NOTICE[8722]: chan_sip.c:15858 sip_reg_timeout: – Registration for ‘edi-1@’ timed out, trying again (Attempt #30

I don’t know why it talks about DNS, i have set srvlookup=no

why does asterisk send a lot of registers to my clients?

jcolp · October 20, 2017, 5:19pm

Asterisk only does what it is configured to do. It won’t send a REGISTER somewhere unless it has been told to. I’d suggest providing the configuration and also SIP traces (sip set debug on) to show what is going on.

david551 · October 22, 2017, 11:53am

It still needs to use DNS on domain names and reverse name lookups.

dendene · October 23, 2017, 8:03am

Hi Jcolp;

herein my sip.conf:

[general]
registertimeout=0
rtcachefriends=yes
context=public ; Default context for incoming calls. Defaults to 'default’
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
allowguest=no

disallow=all
allow=gsm:80 ; changed from 120 to 80 to match conf mixing interval
allow=ulaw
allow=alaw

tcpenable=yes ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)

                            ; to connect at any given time. (default: 100)

transport=tcp ; Set the default transports. The order determines the primary default transport.
; If tcpenable=no and the transport set is tcp, we will fallback to UDP.

srvlookup=no ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
;pedantic=yes ; Enable checking of tags in headers,
; international character conversions in URIs
; and multiline formatted headers for strict
; SIP compatibility (defaults to “yes”)

useragent=PBX
; Allows you to change the user agent string
; The default user agent string also contains the Asterisk
; version. If you don’t want to expose this, change the
; useragent string.
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
; Other options:

videosupport=yes ; Turn on support for SIP video. You need to turn this
; on in this section to get any video support at all.
; You can turn it off on a per peer basis if the general
; video support is enabled, but you can’t enable it for
; one peer only without enabling in the general section.
; If you set videosupport to “always”, then RTP ports will
; always be set up for video, even on clients that don’t
; support it. This assists callfile-derived calls and
; certain transferred calls to use always use video when
; available. [yes|NO|always]

alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
; for any reason, always reject with an identical response
; equivalent to valid username and invalid password/hash
; instead of letting the requester know whether there was
; a matching user or peer for their request. This reduces
; the ability of an attacker to scan for valid SIP usernames.
; This option is set to “yes” by default.

;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
; SIP history is output to the DEBUG logging channel

accept_outofcall_message=yes
outofcall_message_context=astsms_chan_sip
auth_message_requests=yes

realm=mydomain.com
fromdomain=mydoamin.com
language=en
domain=mydomain.com

when i make sip set debug on i have this:

Really destroying SIP dialog ‘4d71ebcc2a6c084444212b960324c992@mydomain.com’ Method: OPTIONS
[Oct 23 09:56:13] ERROR[21377]: chan_sip.c:6067 create_addr_from_peer: ‘UDP’ is not a valid transport for ‘edi-2001’. we only use ‘TCP’! ending call.
[Oct 23 09:56:13] WARNING[21377]: chan_sip.c:16029 transmit_register: Probably a DNS error for registration to edi-2001@, trying REGISTER again (after 20 seconds)
[Oct 23 09:56:13] NOTICE[21377]: chan_sip.c:15858 sip_reg_timeout: – Registration for ‘edi-2001@’ timed out, trying again (Attempt #14)
Really destroying SIP dialog ‘1460681b7568fe901913aa70443fa4a7@mydomain.com’ Method: REGISTER
[Oct 23 09:56:13] ERROR[21377]: chan_sip.c:6067 create_addr_from_peer: ‘UDP’ is not a valid transport for ‘edi-1’. we only use ‘TCP’! ending call.
[Oct 23 09:56:13] WARNING[21377]: chan_sip.c:16029 transmit_register: Probably a DNS error for registration to edi-1@, trying REGISTER again (after 20 seconds)
[Oct 23 09:56:13] NOTICE[21377]: chan_sip.c:15858 sip_reg_timeout: – Registration for ‘edi-1@’ timed out, trying again (Attempt #17)

I have disabled udp, but when i activate it, i see register requests sent to my sip phones.

I use realtime asterisk, my peers are in sippeers table:

herein one user conf:

dendene · October 23, 2017, 8:44am

Hi david551;

is there a conf to disable DNS lookup completely please?

david551 · October 23, 2017, 9:50am

I wouldn’t expect one.

dendene · October 27, 2017, 2:41pm

I have resolved the problem.

I have started asterisk configuration from zero, now, asterisk doesn’t send register to my phones.

however i ignore which parameter is responsible for the old behaviour.

surely, Asterisk is a Strong product, we can do a lot of things with it.