search for: autodomain

Hello List, I am using Samba as PDC for some time now and I want do make the process of joining the domain somewhat more elegant ;) I use partimage to deploy images of my Windows XP-installation to the computers in my lab. In order to automate the process of joining these computers to the domain, I use some small perl-scripts. 1. After installing new software, etc. I have to create a new

...uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet autodomain=yes ; Turn this on to have Asterisk add local host ; name and local IP to domain list. ; and multiline formatted headers for strict localnet=192.168.2.0/23 qualify=no And this for my local settings: [201] type=friend...

...in from an external server I get this: Oct 1715:36:43 NOTICE[4040]: chan_sip.c:10774 handle_request_register: Failed to authenticate user "+16143691415" /(this is the number making the call)/ <sip:+16143601415@XXX.XXX.XXX.XXXX>;tag=as0147d2b3 I have tried using various domains and autodomain=yes, etc.... cannot find anything online concerning this, probably because it's new.... Anyone else have an idea?

...CL (violating IP ) [Mar 10 11:53:59] 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 attachme...

...rpid domain testers.com sippasswd2 and my sip.conf: [general] bindport = 5070 bindaddr = PU.BL.IC.IP udpbindaddr = PU.BL.IC.IP tcpenable = yes limitonpeers = yes rtcachefriends = no tos_sip=cs3 tos_audio=ef realm = testers.com autodomain=yes domain=PU.BL.IC.IP domain=testers.com transport=ws,wss,udp outboundproxy=PU.BL.IC.IP:5060 I'd appreciate Your advice. cheers, Olli -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140908/058...

...anch=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 localnet = internal subnet maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default n...

...kups 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 numeric val > tos=lowdelay ; lowdelay,throughput,reliab...

...mpiled with res_srtp module. Here's my sip.conf 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...

...ARNING[95290]: chan_sip.c:12539 reload_config: Empty context specified at line 43 for domain 'domain.com' and the domain does not appear when I do a sip show domains. It isn't recognised as local, CHECKSIPDOMAIN doesn't do what I want and calls I want are rejected. If I specify autodomain=yes, then the IP address and canonical hostname of the box are added to the domain list, and sip show domains shows them with a context of (default). It would appear that for incoming calls from PSTN gateways at least this does what I want, in that the context specified in the type=peer block i...

...| | dundipbx3 | secret | XXXXXXXXXXXXX | 0 | | dundipbx3 | type | peer | 0 | | dundipbx3 | username | dundipbx3 | 0 | | general | allowguest | no | 0 | | general | autodomain | no | 0 | | general | bindaddr | xxx.yyy.142.203 | 0 | | general | bindport | 5060 | 0 | | dundipbx1 | permit | xxx.yyy.142.204 | 0 | | dundipbx2 | permit | xxx.yyy.1...

Hi. I'm trying to set up a bunch of SIP phones to register in various domains on the local subnets connected to an Asterisk appliance, yet be able to support Internet SIP calling. My config looks like this: [general] context=INVALID allowguest=yes autodomain=no ... domain=redfish-solutions.com,redfish-internet ... [internal](!) type=friend qualify=50 nat=yes host=dynamic canreinvite=no vmexten=voicemail [sipura](!) ; nothing for now, add SLA's later [redfish](!) context=redfish-internal [garage_1](internal,sipura,redfish) callerid=Redfish Solu...

...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 g726nonstandard = no notifyringing = yes srvlookup = yes t1min = 100 t38pt_udptl = no ;tos_audio = none ;t...

...indaddr=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 secret=password regcontext=on_this_system regexten=100 usereq...

...arning 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=default ;language=en...

...172.b.c.d is the address of the Metaswitch) [general] disallow = all allguest = yes allow = all allowguest = yes autocreatepeer = yes autodomain = yes bindaddr = 206.b.c.d bindport = 5060 callerid = "metaswitch" <> canreinvite = no context = test...

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

...ontext=sip realm=nxs.yi.org bindport=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...

...ontext=sip realm=nxs.yi.org bindport=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...

...been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-27108 - Crash using 'data get' CLI command (Reported by Sean Bright) * ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. (Reported by Alexander Traud) * ASTERISK-27100 - channel: ast_waitfordigit_full fails to clear flag in an error branch. (Reported by Corey Farrell) * ASTERISK-27090 - PJSIP: Deadlock using TCP transport...

...been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-27108 - Crash using 'data get' CLI command (Reported by Sean Bright) * ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. (Reported by Alexander Traud) * ASTERISK-27100 - channel: ast_waitfordigit_full fails to clear flag in an error branch. (Reported by Corey Farrell) * ASTERISK-27090 - PJSIP: Deadlock using TCP transport...