[asterisk-bugs] [JIRA] (ASTERISK-19050) Wrong transport for outgoing INVITE
Matt Jordan (JIRA)
noreply at issues.asterisk.org
Wed Feb 20 14:31:18 CST 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-19050?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Matt Jordan updated ASTERISK-19050:
-----------------------------------
Description:
INVITE is sent via UDP despite previous REGISTER via TCP and TCP being declared as primary outboud transport.
REGISTERing to external provider via TCP - Ok.
But outgoing INVITE to the same provider is sent via UDP, despite TCP being configured primary outgoing transport, 403 Forbidden as result.
All combinations of "transport=tcp" or "transport=tcp,udp" in general or peer section or in both - behave the same.
1.8.7.0 - the same.
1.6.2.21 - working as expected (INVITE via TCP)
sip.conf:
{noformat}
[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
{noformat}
extensions.conf:
{noformat}
[general]
static = yes
writeprotect = no
clearglobalvars = no
[default]
exten=>_7XXXXXXXXXX,1,Dial(SIP/${EXTEN}@2provider,30,r)
{noformat}
show's:
{noformat}
localhost*CLI> 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
localhost*CLI>
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: <none>
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: <not set>
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 : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : default
Subscr.Cont. : <Not set>
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>
{noformat}
debug:
{noformat}
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 at 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 at multifon.ru>;tag=as176e1f32
To: <sip:7AAAAAAAAAA at multifon.ru>
Call-ID: 20861aac7e7569e62c3608ff7d62cd9e at 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 at 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 at multifon.ru>;tag=as176e1f32
To: <sip:7AAAAAAAAAA at multifon.ru>;tag=2CB932463135364126DE1600
Call-ID: 20861aac7e7569e62c3608ff7d62cd9e at 127.0.0.1
CSeq: 105 REGISTER
Contact: <sip:7AAAAAAAAAA at 10.128.1.225:5060;transport=TCP>;expires=130
Content-Length: 0
Service-Route: <sip:7AAAAAAAAAA at 193.201.229.35:5060;transport=tcp;lr>
<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '20861aac7e7569e62c3608ff7d62cd9e at 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 at 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 at multifon.ru>;tag=as653efae6
To: <sip:7BBBBBBBBBB at multifon.ru>
Contact: <sip:7AAAAAAAAAA at 10.128.1.225:5060>
Call-ID: 21bd2e403656248b19d83565196334ea at 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 at multifon.ru>;tag=as653efae6
To: <sip:7BBBBBBBBBB at multifon.ru>
Call-ID: 21bd2e403656248b19d83565196334ea at 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 at multifon.ru>;tag=as653efae6
To: <sip:7BBBBBBBBBB at multifon.ru>;tag=aprqngfrt-k04r2m20000c6
Call-ID: 21bd2e403656248b19d83565196334ea at 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 at 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 at multifon.ru>;tag=as653efae6
To: <sip:7BBBBBBBBBB at multifon.ru>;tag=aprqngfrt-k04r2m20000c6
Contact: <sip:7AAAAAAAAAA at 10.128.1.225:5060>
Call-ID: 21bd2e403656248b19d83565196334ea at 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 at multifon.ru>;tag=as653efae6'
localhost*CLI>
Really destroying SIP dialog '21bd2e403656248b19d83565196334ea at multifon.ru' Method: INVITE
localhost*CLI>
Really destroying SIP dialog '20861aac7e7569e62c3608ff7d62cd9e at 127.0.0.1' Method: OPTIONS
localhost*CLI>
Really destroying SIP dialog '20861aac7e7569e62c3608ff7d62cd9e at 127.0.0.1' Method: REGISTER
localhost*CLI>
<< Hangup on console >>
localhost*CLI>
{noformat}
was:
INVITE is sent via UDP despite previous REGISTER via TCP and TCP being declared as primary outboud transport.
REGISTERing to external provider via TCP - Ok.
But outgoing INVITE to the same provider is sent via UDP, despite TCP being configured primary outgoing transport, 403 Forbidden as result.
All combinations of "transport=tcp" or "transport=tcp,udp" in general or peer section or in both - behave the same.
1.8.7.0 - the same.
1.6.2.21 - working as expected (INVITE via TCP)
sip.conf:
[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
extensions.conf:
[general]
static = yes
writeprotect = no
clearglobalvars = no
[default]
exten=>_7XXXXXXXXXX,1,Dial(SIP/${EXTEN}@2provider,30,r)
show's:
localhost*CLI> 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
localhost*CLI>
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: <none>
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: <not set>
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 : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : default
Subscr.Cont. : <Not set>
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>
debug:
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 at 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 at multifon.ru>;tag=as176e1f32
To: <sip:7AAAAAAAAAA at multifon.ru>
Call-ID: 20861aac7e7569e62c3608ff7d62cd9e at 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 at 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 at multifon.ru>;tag=as176e1f32
To: <sip:7AAAAAAAAAA at multifon.ru>;tag=2CB932463135364126DE1600
Call-ID: 20861aac7e7569e62c3608ff7d62cd9e at 127.0.0.1
CSeq: 105 REGISTER
Contact: <sip:7AAAAAAAAAA at 10.128.1.225:5060;transport=TCP>;expires=130
Content-Length: 0
Service-Route: <sip:7AAAAAAAAAA at 193.201.229.35:5060;transport=tcp;lr>
<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '20861aac7e7569e62c3608ff7d62cd9e at 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 at 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 at multifon.ru>;tag=as653efae6
To: <sip:7BBBBBBBBBB at multifon.ru>
Contact: <sip:7AAAAAAAAAA at 10.128.1.225:5060>
Call-ID: 21bd2e403656248b19d83565196334ea at 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 at multifon.ru>;tag=as653efae6
To: <sip:7BBBBBBBBBB at multifon.ru>
Call-ID: 21bd2e403656248b19d83565196334ea at 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 at multifon.ru>;tag=as653efae6
To: <sip:7BBBBBBBBBB at multifon.ru>;tag=aprqngfrt-k04r2m20000c6
Call-ID: 21bd2e403656248b19d83565196334ea at 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 at 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 at multifon.ru>;tag=as653efae6
To: <sip:7BBBBBBBBBB at multifon.ru>;tag=aprqngfrt-k04r2m20000c6
Contact: <sip:7AAAAAAAAAA at 10.128.1.225:5060>
Call-ID: 21bd2e403656248b19d83565196334ea at 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 at multifon.ru>;tag=as653efae6'
localhost*CLI>
Really destroying SIP dialog '21bd2e403656248b19d83565196334ea at multifon.ru' Method: INVITE
localhost*CLI>
Really destroying SIP dialog '20861aac7e7569e62c3608ff7d62cd9e at 127.0.0.1' Method: OPTIONS
localhost*CLI>
Really destroying SIP dialog '20861aac7e7569e62c3608ff7d62cd9e at 127.0.0.1' Method: REGISTER
localhost*CLI>
<< Hangup on console >>
localhost*CLI>
> Wrong transport for outgoing INVITE
> -----------------------------------
>
> Key: ASTERISK-19050
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-19050
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/Messaging, Channels/chan_sip/TCP-TLS
> Affects Versions: 1.8.7.2
> Environment: Minimal CentOS 5.7, Asterisk 1.8.7.0 from repo, updated to 1.8.7.2, behind the NAT.
> Reporter: Dmitry Alekseev
>
> INVITE is sent via UDP despite previous REGISTER via TCP and TCP being declared as primary outboud transport.
> REGISTERing to external provider via TCP - Ok.
> But outgoing INVITE to the same provider is sent via UDP, despite TCP being configured primary outgoing transport, 403 Forbidden as result.
> All combinations of "transport=tcp" or "transport=tcp,udp" in general or peer section or in both - behave the same.
> 1.8.7.0 - the same.
> 1.6.2.21 - working as expected (INVITE via TCP)
> sip.conf:
> {noformat}
> [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
> {noformat}
> extensions.conf:
> {noformat}
> [general]
> static = yes
> writeprotect = no
> clearglobalvars = no
> [default]
> exten=>_7XXXXXXXXXX,1,Dial(SIP/${EXTEN}@2provider,30,r)
> {noformat}
> show's:
> {noformat}
> localhost*CLI> 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
> localhost*CLI>
> 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: <none>
> 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: <not set>
> 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 : <Set>
> MD5Secret : <Not set>
> Remote Secret: <Not set>
> Context : default
> Subscr.Cont. : <Not set>
> 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>
> {noformat}
> debug:
> {noformat}
> 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 at 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 at multifon.ru>;tag=as176e1f32
> To: <sip:7AAAAAAAAAA at multifon.ru>
> Call-ID: 20861aac7e7569e62c3608ff7d62cd9e at 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 at 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 at multifon.ru>;tag=as176e1f32
> To: <sip:7AAAAAAAAAA at multifon.ru>;tag=2CB932463135364126DE1600
> Call-ID: 20861aac7e7569e62c3608ff7d62cd9e at 127.0.0.1
> CSeq: 105 REGISTER
> Contact: <sip:7AAAAAAAAAA at 10.128.1.225:5060;transport=TCP>;expires=130
> Content-Length: 0
> Service-Route: <sip:7AAAAAAAAAA at 193.201.229.35:5060;transport=tcp;lr>
> <------------->
> --- (9 headers 0 lines) ---
> Scheduling destruction of SIP dialog '20861aac7e7569e62c3608ff7d62cd9e at 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 at 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 at multifon.ru>;tag=as653efae6
> To: <sip:7BBBBBBBBBB at multifon.ru>
> Contact: <sip:7AAAAAAAAAA at 10.128.1.225:5060>
> Call-ID: 21bd2e403656248b19d83565196334ea at 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 at multifon.ru>;tag=as653efae6
> To: <sip:7BBBBBBBBBB at multifon.ru>
> Call-ID: 21bd2e403656248b19d83565196334ea at 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 at multifon.ru>;tag=as653efae6
> To: <sip:7BBBBBBBBBB at multifon.ru>;tag=aprqngfrt-k04r2m20000c6
> Call-ID: 21bd2e403656248b19d83565196334ea at 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 at 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 at multifon.ru>;tag=as653efae6
> To: <sip:7BBBBBBBBBB at multifon.ru>;tag=aprqngfrt-k04r2m20000c6
> Contact: <sip:7AAAAAAAAAA at 10.128.1.225:5060>
> Call-ID: 21bd2e403656248b19d83565196334ea at 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 at multifon.ru>;tag=as653efae6'
> localhost*CLI>
> Really destroying SIP dialog '21bd2e403656248b19d83565196334ea at multifon.ru' Method: INVITE
> localhost*CLI>
> Really destroying SIP dialog '20861aac7e7569e62c3608ff7d62cd9e at 127.0.0.1' Method: OPTIONS
> localhost*CLI>
> Really destroying SIP dialog '20861aac7e7569e62c3608ff7d62cd9e at 127.0.0.1' Method: REGISTER
> localhost*CLI>
> << Hangup on console >>
> localhost*CLI>
> {noformat}
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