Hi Alonso, sip.conf
[general] context=hunt_incoming port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=all nat=yes callerid = LITE externip= externhost= autocreatepeer=yes autodomain=yes localnet=xxx.xxx.xxx.xxx/xxx.xxx.xxx.xxx canreinvite=yes language=En allowtransfer=yes realm=telunet domain=192.168.1.5 maxexpiry=3600 defaultexpiry=200 useragent=LITE PBX usereqphone = yes dtmfmode = rfc2833 alwaysauthreject = no regcontext=sipregistrations rtptimeout=3600 rtpholdtimeout=300 rtcachefriends=yes ;--------------------------- SIP DEBUGGING --------------------------------------------------- sipdebug = yes registertimeout=60 registerattempts=5 callgroup=1 pickupgroup=1 callevents=yes ;register => <username>:<password>:<username>@<Sip Proxy IP or domain name> [authentication] [4001] type=friend context=outbound defaultuser=4001 secret=4001 callerid="EXT1" host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4002] type=friend context=outbound defaultuser=4002 secret=4002 callerid="EXT2" host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4003] type=friend context=outbound defaultuser=4003 secret=4003 callerid="EXT3" host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all On Fri, Nov 21, 2014 at 11:52 PM, Alonso Genis <[email protected]> wrote: > > ----- Mensagem original ----- > > > De: "akhilesh chand" <[email protected]> > > Para: "Asterisk Users Mailing List - Non-Commercial Discussion" > > <[email protected]> > > Enviadas: Sexta-feira, 21 de novembro de 2014 14:36:05 > > Assunto: [asterisk-users] Not able to register an Extension > > > Hi folk, > > > I'm trying to register an extension through softphone and got stuck.I got > > below error:- > > > [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via: > missing > > sent-by in Via header > > [Nov 22 07:31:28] ERROR[3522]: netsock2.c:269 ast_sockaddr_resolve: > > getaddrinfo("", "(null)", ...): Name or service not known > > [Nov 22 07:31:28] WARNING[3522]: chan_sip.c:16056 check_via: Could not > > resolve socket address for '' > > Sending to 192.168.1.2:5060 (NAT) > > [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via: > missing > > sent-by in Via header > > [Nov 22 07:31:28] ERROR[3522]: chan_sip.c:7897 process_via: error > processing > > via header > > [Nov 22 07:31:28] WARNING[3522]: chan_sip.c:10420 respprep: error > processing > > via header, will send response to originating address > > > Please let me know how could i solve the same and I will appreciate your > > suggestion. > > Please, send us your sip.conf, i suspect is a problem with your bindaddr > or name resolution. > Alonso. > > > > Thanks & Regards > > Akhilesh > > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
