search for: allowexternalinvit

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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 -------------- next part -------------- An HTML attach...
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090520/a9aeb1ba/attachment...
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...