Displaying 10 results from an estimated 10 matches for "allowexternalinvit".
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allowexternalinvites
2007 Nov 27
5
SIP port 5060 closed - how do I open it?
...g calls, but I cannot receive any incoming calls. A
port scan of my * server shows that port 5060 is closed. How do I open this
port? In my users.conf, I have set [trunk_1] to hassip=yes and port=5060.
Also, in the global SIP.conf file
bindport=5060
bindaddr=0.0.0.0
allowexternaldomains=no
allowexternalinvites=no
Do I have to set allowexternalinvites or allowexternaldomains to yes to
accept INVITEs from my ITSP? I've already configured the system to allow
traffic from their IP address.
Thanks for the help!
Regards,
Zaheer
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2007 Aug 09
1
strange warning
...t based on stale nonce received from '<sip:diet at magnum.axvoice.com>'
I dont know what is the problem. Can somebody explain me this? Below is my
client configuration.
[general]
bindport=9060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
context=incoming
allowexternalinvites=yes
register=> diet:pepsi at magnum.axvoice.com:9060
registertimeout=10 ;(default 20 secs)
registerattempts=10 ;set it to zero for infinit attempts
Following is the server sip account im using for my client asterisk to
register:
[diet]
username=diet
type=friend
secret=pepsi
qualify=...
2006 Nov 08
1
Re: Asterisk and Max TNT PRI to SIP Authentication Issue
> what is the sip.conf for 1239
> which I'm going to assume is a extension on the TNT
>
> Barry
>
> JR Richardson wrote:
> > Hi All,
> >
> > I have a lab setup with two asterisk servers and a MAX TNT in the
> > middle like this:
> >
> > asterisk sip >< sip TNT pri >< pri asterisk
exten 1239 is the CID Number from the
2009 May 20
0
inbound SIP funnies
...t's going on
exten => 555,n,Playback(vq20)
exten => 555,n,Goto(default,555,3) ; repeat
So as far as I can tell, we should be accepting the connection and
playing the voicefile (yup - I know this would be open to the internet,
that's the intention).
Sip.conf also has:
allowexternalinvites=yes
allowexternaldomains=yes
so it should be working I think...
This is a 1.4.15 based asterisk
Thanks
Adrian
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2007 Oct 02
0
Supervised call transfer problem
...14:23:58 ERROR[27281]: cdr_custom.c:129 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : No such file or d
ectory
Here is my sip.conf
[general]
srvlookup=yes ;allows DNS lookups of server names
register => 390641043101:password at sip.uni.it/100
allowexternalinvites=yes
disallow=all
allow=g729
allow=alaw
context=outbound
[sip.uni.it]
type=peer
disallow=all
allow=g729
allow=alaw
dtmfmode=rfc2833
host=sip.uni.it
fromdomain=sip.uni.it
insecure=very
qualify=yes
fromuser=390641043101
authuser=390641043101
username=390641043101
secret=password
canreinvite=no...
2006 Apr 20
0
Re: Asterisk-Users Digest, Vol 21, Issue 113
...for the test
users. The warning indication is no jumping anymore and the voice is not
delayed. This is my sip.conf:
[general]
context=default
;allowguest=no
;realm=mydomain.tld
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
;domain=mydomain.tld
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4
;allowexternalinvites=no
;autodomain=yes
;pedantic=yes
;tos=184
;tos=lowdelay
;maxexpiry=3600
;defaultexpiry=120
;notifymimetype=text/plain
;checkmwi=10
;vmexten=voicemail
;videosupport=yes
;recordhistory=yes
disallow=all
allow=g729
allow=gsm
allow=ulaw
jitterbuffer=yes
maxjitterbuffer=1500
;allow=ilbc
;musicclass=def...
2006 Apr 08
0
Re: [asterisk-dev] bug or bad chan_sip.c
...=5050
bindaddr=nxs.yi.org
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=1000
allow=all
musicclass=default
language=fr
insecure=very
allowguest=yes
rtptimeout=60
rtpholdtimeout=300
useragent=PBX
dtmfmode = rfc2833
checkmwi=20
promiscredir=no
nat=yes
autodomain=no
domain=nxs.yi.org,sip
allowexternalinvites=yes
rtcachefriends=yes
rtupdate=yes
rtautoclear=yes
ignoreregexpire=yes
and extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
//////////////////////////////////////////////////////
[globals]
[mainmenu]
exten => s,1,Answer()
exten =>
s,n,GotoIfTime(09:30-21:00|mo...
2006 Apr 08
0
Re: [asterisk-dev] bug or bad chan_sip.c
...=5050
bindaddr=nxs.yi.org
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=1000
allow=all
musicclass=default
language=fr
insecure=very
allowguest=yes
rtptimeout=60
rtpholdtimeout=300
useragent=PBX
dtmfmode = rfc2833
checkmwi=20
promiscredir=no
nat=yes
autodomain=no
domain=nxs.yi.org,sip
allowexternalinvites=yes
rtcachefriends=yes
rtupdate=yes
rtautoclear=yes
ignoreregexpire=yes
and extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
//////////////////////////////////////////////////////
[globals]
[mainmenu]
exten => s,1,Answer()
exten =>
s,n,GotoIfTime(09:30-21:00|mo...
2009 Aug 04
0
SIP server behind NAT
...d to (0.0.0.0 binds to all)
> srvlookup=yes ; Enable DNS SRV lookups on outbound calls
> ;domain=mydomain.tld ; Set default domain for this host
> ;domain=mydomain.tld,mydomain-incoming
> ;domain=1.2.3.4 ; Add IP address as local domain
> ;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
> ;autodomain=yes ; Turn this on to have Asterisk add local host
> ;pedantic=yes ; Enable slow, pedantic checking for Pingtel
> ;tos=184 ; Set IP QoS to either a keyword or...
2006 Dec 18
0
pap2/wrt54gs/asterisk
...coming
; Add domain and configure incoming context
; for external calls to this domain
;domain=192.168.1.130 ; Add IP address as local domain
;domain=192.168.1.135 ; You can have several "domain" settings
;allowexternalinvites=no ; Disable INVITE and REFER to non-local
domains
; Default is yes
;autodomain=yes ; Turn this on to have Asterisk add
local host
; name and local IP to domain list.
;pedantic=yes ; Enable...