Dear community,
I have been pulling my hair on a sip trunk for which I can’t receive calls from.
My NAT’ing is setup correctly, as I do receive (anonymous) calls for another sip trunk (for another provider).
Trunk config
Trunk name: phonzo
PEER Details:
username=47XXXXXXX
type=friend
secret=*********
realm=phonzo.com
qualify=yes
nat=yes
insecure=very
host=sip.phonzo.com
fromuser=47XXXXXXX
fromdomain=phonzo.com
dtmfmode=inband
disallow=all
canreinvite=no
allow=ulaw&alaw
USER context: XXXXXXX
USER Details:
username=47XXXXXXX
type=friend
secret=*********
realm=phonzo.com
qualify=yes
nat=yes
insecure=very
host=sip.phonzo.com
fromuser=47XXXXXXX
fromdomain=phonzo.com
dtmfmode=inband
disallow=all
canreinvite=no
allow=ulaw&alaw
Registration string:
47XXXXXXX:*********@sip.phonzo.com:5060/88
This seems to work just fine, as I am registered and the peer is qualifying ok.
Asterisk output:
asterisk*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
sip.phonzo.com:5060 N 47XXXXXXX 285 Registered Mon, 09 Jan 2012 21:48:15
asterisk*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
XXXXXXX/47XXXXXXX 80.232.37.178 N 5060 OK (69 ms)
phonzo/47XXXXXXX 80.232.37.178 N 5060 OK (68 ms)
…
I have not tried outgoing calls, as I am not interested in using phonzo for that. But other sip trunks work for outgoing and incoming calls.
Here’s what I get when I enable debugging on this trunk>
Debug
Looking for s in from-sip-external (domain sip.phonzo.com)
<— Transmitting (NAT) to 80.232.37.178:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.232.37.178:5060;branch=z9hG4bK-d8754z-b6fade02bacbce41-1—d8754z-;received=80.232.37.178;rport=5060
From: sip:80.232.37.178:5060;tag=9a1eb35e
To: sip:sip.phonzo.com;tag=as6af3e16d
Call-ID: natpingYWJjZjJlNjQzODFlNTczMmE3MjBhOGRkYzUyYmE5ZmI.
CSeq: 1 OPTIONS
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:xxx.xxx.xxx.xxx:5060
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘natpingYWJjZjJlNjQzODFlNTczMmE3MjBhOGRkYzUyYmE5ZmI.’ in 32000 ms (Method: OPTIONS)
I have enabled guests and anonymous sip calls. This works just fine for another trunk.
Anyone?
Cheers
/Thomas