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:
id | 2
name | edi-2001
ipaddr | 10.106.116.25
port | 60512
regseconds | 1508745709
defaultuser | edi-2001
fullcontact | sip:edi-2001@10.106.116.25:60512^3Btransport=TCP^3Bob
regserver |
useragent | MicroSIP/3.15.10
lastms | 1
host | dynamic
type | friend
context | mydomain.com
permit |
deny |
secret |
md5secret | cfa9fcae484a589f99736f5c437d9007
remotesecret |
transport | tcp
dtmfmode |
directmedia | no
nat | force_rport,comedia
callgroup |
pickupgroup |
language | en
disallow | all
allow | gsm,alaw,ulaw,g729,h261,h263,h264,vp8
insecure |
trustrpid |
progressinband |
promiscredir |
useclientcode |
accountcode |
setvar |
callerid | edi-2001
amaflags |
callcounter |
busylevel |
allowoverlap |
allowsubscribe | yes
videosupport | yes
maxcallbitrate |
rfc2833compensate |
mailbox | edi-2001@mydoamin.com
session-timers |
session-expires |
session-minse |
session-refresher |
t38pt_usertpsource |
regexten |
fromdomain |mydomain.com
fromuser |
qualify | 120000
defaultip |
rtptimeout |
rtpholdtimeout |
sendrpid |
outboundproxy |
callbackextension | yes
timert1 |
timerb |
qualifyfreq |
constantssrc |
contactpermit |
contactdeny |
usereqphone |
textsupport |
faxdetect |
buggymwi |
auth |
fullname |
trunkname |
cid_number |
callingpres |
mohinterpret |
mohsuggest |
parkinglot |
hasvoicemail |
subscribemwi |
vmexten |
autoframing |
rtpkeepalive |
call-limit |
g726nonstandard |
ignoresdpversion |
allowtransfer |
dynamic |
path |
supportpath |