Re: [asterisk-users] after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port
Hi Shitian, here's my sip.conf, but unfortunately I cannot make some other tests with Asterisk 1.8 since the PBX is in production now with Asterisk 1.4.26.2 which seems to work very fine. Thank you G NOTE: tried to change nat and canreinvite parameters but with no success. [general] disallow = all allow = alaw allow = ulaw allow = g726 allow = g723.1 allow = gsm notifyringing = yes limitonpeer = yes notifyhold = yes monitor-format = wav musicclass = default callerid = unknown callcounter = yes allowguest = no context = inbound busylimit = 1 srvlookup = no port = 5060 transport = udp bindaddr = 0.0.0.0 notifybusy = yes register = 123456789:pas...@psip1.mclink.it:5060/123456789 ; [123456789] ; Options from provider (provider.sip-mclink) host = psip1.mclink.it nat = yes canreinvite = yes type = peer context = outbound qualify = yes port = 5060 fromdomain = psip1.mclink.it insecure = very language = it fromuser = 123456789 username = 123456789 secret = passwd On 07/02/2012 12:32 AM, Shitian Long wrote: if you check out your sip.conf. On Jun 29, 2012, at 5:54 PM, gincantalupo wrote: Hi all, after upgrading my Asterisk 1.4.26.2 to 1.8.11.0 I cannot register to my VoIP provider because it says I'm trying to connect to port 55150 (that's what the call center guy told me)...but I'm not. In my sip I've set port=5060, not 55150. The strange thing is that the rport inside SIP packets (sip set debug) coming back from my provider is set to 55150.seen on both Asterisk 1.4 and 1.8 Does anybody have any idea? Thank you. Giorgio -- _ -- 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
Re: [asterisk-users] after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port
Il 05/07/2012 16.43, gincantalupo ha scritto: here's my sip.conf, but unfortunately I cannot make some other tests with Asterisk 1.8 since the PBX is in production now with Asterisk 1.4.26.2 which seems to work very fine. I'm using the same provider on many sites without special issues. My sip.conf follows, tested time ago on 1.4, ported with minor changes to 1.6.2 (now in production) then ported to 1.8 without changes (lab test only). [general] context=public-direct-dialin allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes useragent=TeeBX alwaysauthreject=yes videosupport=no notifybusy=yes counteronpeer=yes notifyhold=no pedantic=yes callcounter=yes defaultexpiry=120 minexpiry=60 maxexpiry=3600 localnet=172.31.255.0/24 localnet=172.31.254.0/24 ; MCLink register = username:p...@psip1.mclink.it/username [mclink-06x] type=peer defaultuser=username secret=pass fromuser=username host=psip1.mclink.it context=mclink-06x-incoming fromdomain=psip1.mclink.it language=it-it nat=yes qualify=2000 directmedia=no insecure=port,invite dtmfmode=rfc2833 disallow=all allow=alaw allow=gsm call-limit=5 -- _ -- 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
Re: [asterisk-users] after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port
if you check out your sip.conf. On Jun 29, 2012, at 5:54 PM, gincantalupo wrote: Hi all, after upgrading my Asterisk 1.4.26.2 to 1.8.11.0 I cannot register to my VoIP provider because it says I'm trying to connect to port 55150 (that's what the call center guy told me)...but I'm not. In my sip I've set port=5060, not 55150. The strange thing is that the rport inside SIP packets (sip set debug) coming back from my provider is set to 55150.seen on both Asterisk 1.4 and 1.8 Does anybody have any idea? Thank you. Giorgio -- _ -- 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
[asterisk-users] after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port
Hi all, after upgrading my Asterisk 1.4.26.2 to 1.8.11.0 I cannot register to my VoIP provider because it says I'm trying to connect to port 55150 (that's what the call center guy told me)...but I'm not. In my sip I've set port=5060, not 55150. The strange thing is that the rport inside SIP packets (sip set debug) coming back from my provider is set to 55150.seen on both Asterisk 1.4 and 1.8 Does anybody have any idea? Thank you. Giorgio -- _ -- 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
