[asterisk-users] after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port

[gincantalupo]

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:passwd at 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
>>
>> --
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> _____________________________________________________________________
> -- 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
>
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