Hello!
Minimal CentOS 5.7, Asterisk 1.8.7.0 from repo, updated to 1.8.7.2, behind the NAT.
REGISTERing to external provider via TCP - Ok.
But outgoing INVITE to the same provider is sent via UDP, despite TCP being (I hope) configured primary outgoing transport, 403 Forbidden as result.
1.8.7.0 - the same.
1.6.2.21 - working as expected (INVITE via TCP)
sip.conf:
[code][general]
context=default
allowguest=no
allowoverlap=no
alwaysauthreject=yes
udpbindaddr = 0.0.0.0
tcpenable = yes
tcpbindaddr = 0.0.0.0
transport=tcp,udp
srvlookup=yes
nat = yes
disallow=all
allow=alaw
allow=g729
allow=g723
allow=ulaw
dtmfmode=inband
canreinvite=no
externip=AAA.BBB.CCC.DDD
register = tcp://7AAAAAAAAAA@multifon.ru:password:7AAAAAAAAAA@sbc.megafon.ru/7AAAAAAAAAA
[2provider]
type=peer
host = multifon.ru
username = 7AAAAAAAAAA
secret = password
context = default
outboundproxy = sbc.megafon.ru
fromdomain = multifon.ru
fromuser = 7AAAAAAAAAA
authuser = 7AAAAAAAAAA
insecure = port,invite
;transport=tcp,udp[/code]
extensions.conf:
[code][general]
static = yes
writeprotect = no
clearglobalvars = no
[default]
exten=>_7XXXXXXXXXX,1,Dial(SIP/${EXTEN}@2provider,30,r)
[/code]
show’s:
[code]localhostCLI> core show version
Asterisk 1.8.7.2 built by root @ localhost.localdomain on a i686 running Linux on 2011-12-09 17:52:28 UTC
localhostCLI>
localhost*CLI> sip show settings
Global Settings:
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: 0.0.0.0:5060
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX 1.8.7.2
SDP Session Name: Asterisk PBX 1.8.7.2
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Legacy userfield parse: No
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
SIP address remapping: Disabled, no localnet list
Externhost:
Externaddr: AAA.BBB.CCC.DDD:0
Externrefresh: 10
Global Signalling Settings:
Codecs: 0x10d (g723|ulaw|alaw|g729)
Codec Order: alaw:20,g729:20,g723:30,ulaw:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
Allowed transports: TCP,UDP
Outbound transport: TCP
Context: default
Force rport: Yes
DTMF: inband
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
localhost*CLI> sip show peer 2provider
- Name : 2provider
Secret :
MD5Secret :
Remote Secret:
Context : default
Subscr.Cont. :
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
FromUser : 7AAAAAAAAAA
FromDomain : multifon.ru Port 5060
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : No
Callerid : “” <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : port,invite
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
Outb. proxy : sbc.megafon.ru
DTMFmode : inband
Timer T1 : 500
Timer B : 32000
ToHost : multifon.ru
Addr->IP : 193.201.229.35:5060
Defaddr->IP : (null)
Prim.Transp. : TCP
Allowed.Trsp : TCP,UDP
Def. Username: 7AAAAAAAAAA
SIP Options : (none)
Codecs : 0x10d (g723|ulaw|alaw|g729)
Codec Order : (alaw:20,g729:20,g723:30,ulaw:20)
Auto-Framing : No
100 on REG : No
Status : Unmonitored
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
localhost*CLI>[/code]
debug:
[code]localhost*CLI> sip set debug peer 2provider
localhost*CLI>
SIP Debugging Enabled for IP: 193.201.229.35
localhost*CLI>
[Dec 14 16:20:19] NOTICE[3535]: chan_sip.c:12596 sip_reregister: – Re-registration for 7AAAAAAAAAA@sbc.megafon.ru
localhost*CLI>
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 193.201.229.35:5060:
REGISTER sip:multifon.ru SIP/2.0
Via: SIP/2.0/TCP 10.128.1.225:5060;branch=z9hG4bK3cedc749;rport
Max-Forwards: 70
From: sip:7AAAAAAAAAA@multifon.ru;tag=as176e1f32
To: sip:7AAAAAAAAAA@multifon.ru
Call-ID: 20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX 1.8.7.2
Authorization: Digest username=“7AAAAAAAAAA”, realm=“BREDBAND”, algorithm=MD5, uri=“sip:multifon.ru”, nonce=“MTMyMzgzNjIzNDpfLjfk775Bd2kMNs4bxIr3”, response=“2fdc731e5f472d43843edc9d95b8ea9e”, opaque=“MTMyMzgzNjIzNDpfLjfk775Bd2kMNs4bxIr3”, qop=auth, cnonce=“74cc256c”, nc=00000003
Expires: 120
Contact: sip:7AAAAAAAAAA@10.128.1.225:5060;transport=TCP
Content-Length: 0
localhost*CLI>
<— SIP read from TCP:193.201.229.35:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.128.1.225:5060;received=AAA.BBB.CCC.DDD;branch=z9hG4bK3cedc749;rport=36591
From: sip:7AAAAAAAAAA@multifon.ru;tag=as176e1f32
To: sip:7AAAAAAAAAA@multifon.ru;tag=2CB932463135364126DE1600
Call-ID: 20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1
CSeq: 105 REGISTER
Contact: sip:7AAAAAAAAAA@10.128.1.225:5060;transport=TCP;expires=130
Content-Length: 0
Service-Route: sip:7AAAAAAAAAA@193.201.229.35:5060;transport=tcp;lr
<------------->
— (9 headers 0 lines) —
Scheduling destruction of SIP dialog ‘20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1’ in 32000 ms (Method: REGISTER)
[Dec 14 16:20:19] NOTICE[3547]: chan_sip.c:20155 handle_response_register: Outbound Registration: Expiry for sbc.megafon.ru is 130 sec (Scheduling reregistration in 115 s)
localhost*CLI> console dial 7BBBBBBBBBB
localhost*CLI>
Audio is at 5060
localhost*CLI>
Adding codec 0x8 (alaw) to SDP
localhost*CLI>
Adding codec 0x4 (ulaw) to SDP
localhost*CLI>
Reliably Transmitting (NAT) to 193.201.229.35:5060:
INVITE sip:7BBBBBBBBBB@multifon.ru SIP/2.0
Via: SIP/2.0/UDP 10.128.1.225:5060;branch=z9hG4bK2d2b39d7;rport
Max-Forwards: 70
From: “asterisk” sip:7AAAAAAAAAA@multifon.ru;tag=as653efae6
To: sip:7BBBBBBBBBB@multifon.ru
Contact: sip:7AAAAAAAAAA@10.128.1.225:5060
Call-ID: 21bd2e403656248b19d83565196334ea@multifon.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.7.2
Date: Wed, 14 Dec 2011 10:20:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 202
v=0
o=root 518834463 518834463 IN IP4 10.128.1.225
s=Asterisk PBX 1.8.7.2
c=IN IP4 10.128.1.225
t=0 0
m=audio 16650 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
localhost*CLI>
<— SIP read from UDP:193.201.229.35:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.128.1.225:5060;received=AAA.BBB.CCC.DDD;branch=z9hG4bK2d2b39d7;rport=1027
From: “asterisk” sip:7AAAAAAAAAA@multifon.ru;tag=as653efae6
To: sip:7BBBBBBBBBB@multifon.ru
Call-ID: 21bd2e403656248b19d83565196334ea@multifon.ru
CSeq: 102 INVITE
<------------->
— (6 headers 0 lines) —
<— SIP read from UDP:193.201.229.35:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.128.1.225:5060;received=AAA.BBB.CCC.DDD;branch=z9hG4bK2d2b39d7;rport=1027
From: “asterisk” sip:7AAAAAAAAAA@multifon.ru;tag=as653efae6
To: sip:7BBBBBBBBBB@multifon.ru;tag=aprqngfrt-k04r2m20000c6
Call-ID: 21bd2e403656248b19d83565196334ea@multifon.ru
CSeq: 102 INVITE
Reason: Q.850;cause=55;text=“Call Terminated”
<------------->
— (7 headers 0 lines) —
Transmitting (NAT) to 193.201.229.35:5060:
ACK sip:7BBBBBBBBBB@multifon.ru SIP/2.0
Via: SIP/2.0/UDP 10.128.1.225:5060;branch=z9hG4bK2d2b39d7;rport
Max-Forwards: 70
From: “asterisk” sip:7AAAAAAAAAA@multifon.ru;tag=as653efae6
To: sip:7BBBBBBBBBB@multifon.ru;tag=aprqngfrt-k04r2m20000c6
Contact: sip:7AAAAAAAAAA@10.128.1.225:5060
Call-ID: 21bd2e403656248b19d83565196334ea@multifon.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.7.2
Content-Length: 0
localhost*CLI>
[Dec 14 16:20:43] WARNING[3535]: chan_sip.c:19680 handle_response_invite: Received response: “Forbidden” from ‘“asterisk” sip:7AAAAAAAAAA@multifon.ru;tag=as653efae6’
localhost*CLI>
Really destroying SIP dialog ‘21bd2e403656248b19d83565196334ea@multifon.ru’ Method: INVITE
localhost*CLI>
Really destroying SIP dialog ‘20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1’ Method: OPTIONS
localhost*CLI>
Really destroying SIP dialog ‘20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1’ Method: REGISTER
localhost*CLI>
<< Hangup on console >>
localhost*CLI>
[/code]