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...line. I can make outgoing 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 -----...

Hello, all. We have successfully connected our new Asterisk 1.6.1.1 PBX to Vitelity's network and have been very happy with them thus far. However, we'd like to use domains in our sip.conf to facilitate routing in our multi-tenant environment. We also like to set allowexternaldomains=no for security. However, this breaks our inbound PSTN calling from Vitelity. Is it possible to use allowexternaldomains=no with VoIP carriers? We tried setting fromdomain=<a local domain>. We tried defining a Vitelity domain based upon both vitelity.net and IP address (although I'm r...

...9] WARNING[21272]: chan_sip.c:13837 register_verify: Registration denied because of contact ACL Note, that the server IP is 172.16.16.11 and the SBC internal Interface IP is 172.16.16.6 the following lines have been added to sip.conf dynamic_exclude_static = yes autodomain=yes domain=172.16.16.6 allowexternaldomains=no In addition, in the general endpoint template in sip.conf, I have the lines contactdeny=0.0.0.0/0.0.0.0 contactpermit=172.16.16.0/255.255.255.0 host=dynamic What else am I missing? Thanks \RR -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.di...

...: "username"<sip:s at 216.26.109.22>;tag=as4fffe111 Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Max-Forwards: 70 Content-Length: 0 Here's the configs: subscribecontext = device-hints allowexternaldomains = yes allowguest = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = yes checkmwi = 10 compactheaders = no defaultexpiry = 120 dumphistory = no externip = 216.26.109.22 g726nonstandard = no jbenable = yes jbforce = no jblog = no localn...

...contents: bindport = 5070 ; using this since Kamailio is at 5060 bindaddr = PU.BL.IC.IP tcpenable = yes ;no limitonpeers = yes rtcachefriends = yes ; for realtime rtupdate=yes tos_sip=cs3 tos_audio=ef useragent=MyAsterisk realm = myrealm.com autodomain=no domain=PU.BL.IC.IP domain=testers.com allowexternaldomains=no allowguest=no avpf=yes encryption=yes transport=ws,udp icesupport=yes srvlookup=yes And here's an example of a ws client in my realtime peer table: id: 4 name: 660 ipaddr: PU.BL.IC.IP port: 5060 regseconds: 1406368294...

...gt; 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.htm

...wn it. Waiting for a while (minute or two), retrieving DEVICE_STATE(SIP/device) again - no changes! Awaited UNAVAILABLE. doing from CLI: sip qualify peer device load no result. What I did not configured? My sip.conf [general] context = default allowguest = no bindport = 5060 bindaddr = 0.0.0.0 allowexternaldomains = no allowoverlap = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = no checkmwi = 10 compactheaders = no defaultexpiry = 120 domain=sop-korniychuk domain=172.30.8.13 domain=172.30.8.13:5060 dumphistory = no externrefresh = 10 g726non...

...tries have been made in sip.conf: [general] context=default udpbindaddr=0.0.0.0 bindport=5060 srvlookup=no language=en contactpermit=127.0.0.1/255.255.255.0 contactpermit=10.0.0.2/255.255.255.0 sipdebug=yes allowsubscribe=no localnet=10.0.0.1/255.255.255.0 localnet=10.0.0.2/255.255.255.0 nat=never allowexternaldomains=no domain=10.0.0.1 matchexterniplocally=yes autodomain=yes directmedia=yes disallow=all allow=gsm allow=ulaw allow=alaw ;entry for phones [100] type=friend context=phones host=dynamic [102] type=friend context=phones host=dynamic ;entry for users [user1] type=friend context=on_this_system secre...

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

Hi The subject line says it all how do I enable this style of call. Pointers to the dns setup and asterisk setup would be great or even search words for google, as I am not sure how to search for this type of request. Alex -- There is no instance of a country having benefited from prolonged warfare -- Sun Tzu - The Art of War -------------- next part -------------- A

...g to rtpengine. Here's my sip.conf: bindport = 5070 ;Kamailio is at port 5060, and it's always used as outbound proxy bindaddr = PU.BL.IC.IP tcpenable = yes limitonpeers = yes rtcachefriends = yes rtupdate=yes tos_sip=cs3 tos_audio=ef realm = testers.com autodomain=no domain=testers.com allowexternaldomains=no allowguest=no ;avpf=yes ; encryption=yes transport=ws,wss,udp icesupport=yes srvlookup=yes nat=force_rport,comedia videosupport=yes directmedia=no And here's the way I've defined my websocket peer to my sippeers table: id: 4 name: 660 ipaddr: PU.BL.IC.IP...