I have edited this post, this one have more information.
Hi,
I’m pretty new here and hope someone can help me with that issue
I’m setting Asterisk as SIP client of a SIP Provider (GVT in Brazil).
I can register a peer normally. Receive calls but can’t make outbound calls.
Always receive 403 Forbidden.
The SIP Provider give me this information:
Host: www2.gvt.com.br
Domain: voxng03.gvt.com.br
Username: XXXXXXXX
Secret: MY_SECRET
The point is: voxng03.gvt.com.br is not on the internet. This is probably a realm on the internal network in the gvt.com.br.
When I set in asterisk host=voxng03.gvt.com.br, can’t make calls because this host is not reachable. So I have to put host=www2.gvt.com.br and so I only receive the same response: 403 Forbidden.
My register string is like this:
register=>XXXXXXXX@voxng03.gvt.com.br:MY_SECRET:XXXXXXXX@www2.gvt.com.br/XXXXXXXX
My peer settings:
[XXXXXXXX]
username = XXXXXXXX@voxng03.gvt.com.br
type = peer
insecure = invite,port
fromdomain = voxng03.gvt.com.br
fromuser = XXXXXXXX
secret = MY_SECRET
;host = voxng03.gvt.com.br -> with this host doesn’t work
host = www2.gvt.com.br
realm=voxng03.gvt.com.br
context = MY_INTERNAL_CONTEXT
auth = XXXXXXXX:MY_SECRET@voxng03.gvt.com.br
fullcontact = sip:voxng03.gvt.com.br
canreinvite = no
disallow = all
allow = g729
allow = ulaw
allow = alaw
My Dial command:
exten => _XXXXXXXX,1,Dial(SIP/XXXXXXXX/${EXTEN},90,tT)
This is my SIP REGISTER:
REGISTER sip:voxng03.gvt.com.br SIP/2.0
Via: SIP/2.0/UDP MY_EXTERN_IP:5060;branch=z9hG4bK6122c55b;rport
Max-Forwards: 70
From: sip:XXXXXXXX@voxng03.gvt.com.br;tag=as46a8f0af
To: sip:XXXXXXXX@voxng03.gvt.com.br
Call-ID: 5dd200387c84dec7320e5835308b0093@127.0.0.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX 1.6.0.6
Expires: 1800
Contact: sip:XXXXXXXX@MY_EXTERN_IP
Event: registration
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP MY_EXTERN_IP:5060;received=MY_EXTERN_IP;branch=z9hG4bK6122c55b;rport=5060
From: sip:XXXXXXXX@voxng03.gvt.com.br;tag=as46a8f0af
To: sip:XXXXXXXX@voxng03.gvt.com.br;tag=SDvhmq599-369a9b208be6e716bd03f57754d8f55e
Call-ID: 5dd200387c84dec7320e5835308b0093@127.0.0.1
CSeq: 102 REGISTER
WWW-Authenticate: Digest qop=“auth”,nonce=“c3fb5826c65d60bdd4fa16ad3e7c560a”,realm="voxng03.gvt.com.br"
Content-Length: 0
REGISTER sip:voxng03.gvt.com.br SIP/2.0
Via: SIP/2.0/UDP MY_EXTERN_IP:5060;branch=z9hG4bK25197b05;rport
Max-Forwards: 70
From: sip:XXXXXXXX@voxng03.gvt.com.br;tag=as22aeaeb9
To: sip:XXXXXXXX@voxng03.gvt.com.br
Call-ID: 5dd200387c84dec7320e5835308b0093@127.0.0.1
CSeq: 103 REGISTER
User-Agent: Asterisk PBX 1.6.0.6
Authorization: Digest username=“XXXXXXXX”, realm=“voxng03.gvt.com.br”, algorithm=MD5, uri=“sip:voxng03.gvt.com.br”, nonce=“c3fb5826c65d6fa16ad3e7c560a”, response=“f103b1f5c0e7b5fd07a0ac7384bcec70”, qop=auth, cnonce=“753cd41f”, nc=00000001
Expires: 1800
Contact: sip:XXXXXXXX@MY_EXTERN_IP
Event: registration
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.40.191.59:5060;received=201.40.191.59;branch=z9hG4bK25197b05;rport=5060
From: sip:68166489@voxng03.gvt.com.br;tag=as22aeaeb9
To: sip:68166489@voxng03.gvt.com.br;tag=SDvhmq599-a648ecf943d0d707b5f0c5d60fa4ada8
Call-ID: 5dd200387c84dec7320e5835308b0093@127.0.0.1
CSeq: 103 REGISTER
Contact: sip:68166489@189.10.133.116;expires=136
Contact: sip:68166489@201.40.191.59;expires=900
Content-Length: 0
When I try to make a call…
[Nov 2 13:51:46] DEBUG[20653]: pbx.c:3086 pbx_extension_helper: Launching ‘Dial’
– Executing [YYYYYYYY@ramais:3] Dial(“SIP/200-016b6a60”, “SIP/XXXXXXXX/YYYYYYYY,90,tT”) in new stack
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:19971 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw)
== Using SIP RTP CoS mark 5
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6033 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:3924 do_setnat: Setting NAT on RTP to Off
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:2424 obproxy_get: OBPROXY: Not applying OBproxy to this call
[Nov 2 13:51:46] DEBUG[20653]: acl.c:490 ast_ouraddrfor: Found IP address for this socket
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:2697 ast_sip_ouraddrfor: Target address 200.146.79.165 is not local, substituting externip
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:5453 sip_new: *** Our native formats are 0x4 (ulaw)
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:5454 sip_new: *** Joint capabilities are 0x4 (ulaw)
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:5455 sip_new: *** Our capabilities are 0x10c (ulaw|alaw|g729)
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:5456 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:5458 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw)
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:5484 sip_new: This channel will not be able to handle video.
[Nov 2 13:51:46] DEBUG[20653]: rtp.c:1970 ast_rtp_make_compatible: Seeded SDP of ‘SIP/XXXXXXXX-016a4500’ with that of ‘SIP/200-016b6a60’
[Nov 2 13:51:46] DEBUG[20653]: channel.c:3894 ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[Nov 2 13:51:46] DEBUG[20653]: channel.c:3894 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[Nov 2 13:51:46] DEBUG[20653]: channel.c:3894 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[Nov 2 13:51:46] DEBUG[20653]: channel.c:3894 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[Nov 2 13:51:46] DEBUG[20653]: channel.c:3894 ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[Nov 2 13:51:46] DEBUG[20653]: channel.c:3894 ast_channel_inherit_variables: Not copying variable MIXMONITOR_FILENAME.
[Nov 2 13:51:46] DEBUG[20653]: channel.c:3894 ast_channel_inherit_variables: Not copying variable ARQUIVO.
[Nov 2 13:51:46] DEBUG[20653]: channel.c:3894 ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Nov 2 13:51:46] DEBUG[20653]: channel.c:3894 ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[Nov 2 13:51:46] DEBUG[20653]: channel.c:3894 ast_channel_inherit_variables: Not copying variable SIPURI.
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:4298 sip_call: Outgoing Call for YYYYYYYY
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:4524 update_call_counter: Updating call counter for outgoing call
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:4320 sip_call: Our T38 capability (0), joint T38 capability (0)
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:8361 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:8362 add_sdp: ** Our prefcodec: 0x4 (ulaw)
Audio is at MY_EXTERN_IP port 16458
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:8515 add_sdp: – Done with adding codecs to SDP
[Nov 2 13:51:46] DEBUG[20653]: channel.c:2845 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=36)
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:8580 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw)
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:2362 initialize_initreq: Initializing initreq for method INVITE - callid 6f6466061bb340753cc4ae8159b4fd50@voxng03.gvt.com.br
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 0 [ 44]: INVITE sip:YYYYYYYY@www2.gvt.com.br SIP/2.0
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP MY_EXTERN_IP:5060;branch=z9hG4bK0e851ee0;rport
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 2 [ 16]: Max-Forwards: 70
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 3 [ 70]: From: “Roberto Linck” sip:XXXXXXXX@voxng03.gvt.com.br;tag=as30a46b6b
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 4 [ 35]: To: sip:YYYYYYYY1@www2.gvt.com.br
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 5 [ 38]: Contact: sip:XXXXXXXX@MY_EXTERN_IP
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 6 [ 60]: Call-ID: 6f6466061bb340753cc4ae8159b4fd50@voxng03.gvt.com.br
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 7 [ 16]: CSeq: 102 INVITE
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 8 [ 32]: User-Agent: Asterisk PBX 1.6.0.6
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 9 [ 35]: Date: Tue, 02 Nov 2010 15:51:46 GMT
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 10 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 11 [ 26]: Supported: replaces, timer
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 12 [ 29]: Content-Type: application/sdp
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 13 [ 19]: Content-Length: 289
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Header 14 [ 0]:
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Body 0 [ 3]: v=0
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Body 1 [ 48]: o=root 464492712 464492712 IN IP4 189.10.133.116
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Body 2 [ 22]: s=Asterisk PBX 1.6.0.6
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Body 3 [ 23]: c=IN IP4 MY_EXTERN_IP
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Body 4 [ 5]: t=0 0
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Body 5 [ 29]: m=audio 16458 RTP/AVP 0 8 101
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Body 6 [ 20]: a=rtpmap:0 PCMU/8000
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Body 7 [ 20]: a=rtpmap:8 PCMA/8000
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Body 9 [ 15]: a=fmtp:101 0-16
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Body 10 [ 25]: a=silenceSupp:off - - - -
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Body 11 [ 10]: a=ptime:20
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:6337 parse_request: Body 12 [ 10]: a=sendrecv
Reliably Transmitting (no NAT) to 200.146.79.165:5060:
INVITE sip:YYYYYYYY@www2.gvt.com.br SIP/2.0
Via: SIP/2.0/UDP 189.10.133.116:5060;branch=z9hG4bK0e851ee0;rport
Max-Forwards: 70
From: “Roberto Linck” sip:XXXXXXXX@voxng03.gvt.com.br;tag=as30a46b6b
To: sip:YYYYYYYY@www2.gvt.com.br
Contact: sip:68166489@MY_EXTERN_IP
Call-ID: 6f6466061bb340753cc4ae8159b4fd50@voxng03.gvt.com.br
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.6
Date: Tue, 02 Nov 2010 15:51:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 464492712 464492712 IN IP4 189.10.133.116
s=Asterisk PBX 1.6.0.6
c=IN IP4 MY_EXTERN_IP
t=0 0
m=audio 16458 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:2916 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #1088
[Nov 2 13:51:46] DEBUG[20653]: chan_sip.c:2598 __sip_xmit: Trying to put ‘INVITE sip’ onto UDP socket destined for 200.146.79.165:5060
– Called XXXXXXXX/YYYYYYYY
== Begin MixMonitor Recording SIP/200-016b6a60
SIP/2.0 100 Trying
Via: SIP/2.0/UDP MY_EXTERN_IP:5060;received=201.40.191.59;branch=z9hG4bK0e851ee0;rport=5060
From: “Roberto Linck” sip:XXXXXXXX@voxng03.gvt.com.br;tag=as30a46b6b
To: sip:YYYYYYYY@www2.gvt.com.br
Call-ID: 6f6466061bb340753cc4ae8159b4fd50@voxng03.gvt.com.br
CSeq: 102 INVITE
<------------->
[Nov 2 13:51:46] DEBUG[18105]: chan_sip.c:6337 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying
[Nov 2 13:51:46] DEBUG[18105]: chan_sip.c:6337 parse_request: Header 1 [ 93]: Via: SIP/2.0/UDP MY_EXTERN_IP:5060;received=MY_EXTERN_IP;branch=z9hG4bK0e851ee0;rport=5060
[Nov 2 13:51:46] DEBUG[18105]: chan_sip.c:6337 parse_request: Header 2 [ 70]: From: “Roberto Linck” sip:XXXXXXXX@voxng03.gvt.com.br;tag=as30a46b6b
[Nov 2 13:51:46] DEBUG[18105]: chan_sip.c:6337 parse_request: Header 3 [ 35]: To: sip:YYYYYYYY@www2.gvt.com.br
[Nov 2 13:51:46] DEBUG[18105]: chan_sip.c:6337 parse_request: Header 4 [ 60]: Call-ID: 6f6466061bb340753cc4ae8159b4fd50@voxng03.gvt.com.br
[Nov 2 13:51:46] DEBUG[18105]: chan_sip.c:6337 parse_request: Header 5 [ 16]: CSeq: 102 INVITE
[Nov 2 13:51:46] DEBUG[18105]: chan_sip.c:6337 parse_request: Header 6 [ 0]:
— (6 headers 0 lines) —
[Nov 2 13:51:46] DEBUG[18105]: chan_sip.c:6099 find_call: = Found Their Call ID: 6f6466061bb340753cc4ae8159b4fd50@voxng03.gvt.com.br Their Tag Our tag: as30a46b6b
[Nov 2 13:51:46] DEBUG[18105]: chan_sip.c:3108 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #1088 - INVITE (got response)
[Nov 2 13:51:46] DEBUG[18105]: chan_sip.c:3115 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '6f6466061bb340753cc4ae8159b4fd50@voxng03.gvt.com.br’ Request 102: Found
[Nov 2 13:51:46] DEBUG[18105]: chan_sip.c:14929 handle_response_invite: SIP response 100 to standard invite
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP MY_EXTERN_IP:5060;received=MY_EXTERN_IP;branch=z9hG4bK0e851ee0;rport=5060
From: “Roberto Linck” sip:XXXXXXXX@voxng03.gvt.com.br;tag=as30a46b6b
To: sip:YYYYYYYY@www2.gvt.com.br;tag=SD3p7ie99-1146397198
Call-ID: 6f6466061bb340753cc4ae8159b4fd50@voxng03.gvt.com.br
CSeq: 102 INVITE
User-Agent: Nortel SESM 12.0.6.0
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc
Content-Length: 0
I tried the same using X-Lite and can make calls normally.
Request-Line: INVITE sip:YYYYYYYY@voxng03.gvt.com.br;transport=udp SIP/2.0
Method: INVITE
Request-URI: sip:YYYYYYYY@voxng03.gvt.com.br;transport=udp
Request-URI User Part: YYYYYYYY
Request-URI Host Part: voxng03.gvt.com.br
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.XXX.XXX:12846;branch=z9hG4bK-d8754z-b44572367bb4a973-1—d8754z-;rport
Transport: UDP
Sent-by Address: 192.168.XXX.XXX
Sent-by port: 12846
Branch: z9hG4bK-d8754z-b44572367bb4a973-1—d8754z-
RPort: rport
Max-Forwards: 70
Contact: <sip:XXXXXXXX@MY_EXTERN_IP:12846;transport=udp >
Contact Binding: <sip:XXXXXXXX@MY_EXTERN_IP:12846;transport=udp >
To: sip:YYYYYYYY@voxng03.gvt.com.br
SIP to address: sip:YYYYYYYY@voxng03.gvt.com.br
SIP to address User Part: YYYYYYYY
SIP to address Host Part: voxng03.gvt.com.br
From: "YYYYYYYY"sip:XXXXXXXX@voxng03.gvt.com.br;tag=a1 f0e83d
SIP Display info: "sip:XXXXXXXX"
SIP from address: sip:sip:XXXXXXXX@voxng03.gvt.com.br
SIP from address User Part: sip:XXXXXXXX
SIP from address Host Part: voxng03.gvt.com.br
SIP tag: a1f0e83d
Call-ID: YWViODc0YTkwZTIyZTY5OGNkZTNkOWMxMDY4YjM5Mzc.
CSeq: 1 INVITE
Sequence Number: 1
Method: INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 410
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 12932851883775432 1 IN IP4 192.168.1.193
Session Name (s): CounterPath X-Lite 4.0
Connection Information ©: IN IP4 192.168.XXX.XXX
Time Description, active time (t): 0 0
Session Attribute (a): ice-ufrag:9aafec
Session Attribute (a): ice-pwd:86451fae77632c837296596270727bff
Media Description, name and address (m): audio 56748 RTP/AVP 107 0 8 101
Media Attribute (a): rtpmap:107 BV32/16000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): sendrecv
Media Attribute (a): candidate:1 1 UDP 659136 192.168.XXX.XXX 56748 typ host
Media Attribute (a): candidate:1 2 UDP 659134 192.168.XXX.XXX 56749 typ host
SIP response from SIP Provider
Status-Line: SIP/2.0 407 Proxy Authentication Required
Status-Code: 407
[Resent Packet: False]
[Request Frame: 59]
[Response Time (ms): 48]
Message Header
Via: SIP/2.0/UDP 192.168.XXX.XXX:12846;received=my_extern_ip;branch =z9hG4bK-d8754z-b44572367bb4a973-1—d8754z-;rport=12846
Transport: UDP
Sent-by Address: 192.168.XXX.XXX
Sent-by port: 12846
Received: my_extern_ip
Branch: z9hG4bK-d8754z-b44572367bb4a973-1—d8754z-
RPort: 12846
To: sip:YYYYYYYY@voxng03.gvt.com.br;tag=SDi5k7399-b59feeac6822c5ef52d0c44d7a91fedf
SIP to address: sip:YYYYYYYY@voxng03.gvt.com.br
SIP to address User Part: YYYYYYYY
SIP to address Host Part: voxng03.gvt.com.br
SIP tag: SDi5k7399-b59feeac6822c5ef52d0c44d7a91fedf
From: "XXXXXXXX"sip:XXXXXXXX@voxng03.gvt.com.br;tag=a1 f0e83d
SIP Display info: "XXXXXXXX"
SIP from address: sip:XXXXXXXX@voxng03.gvt.com.br
SIP from address User Part: XXXXXXXX
SIP from address Host Part: voxng03.gvt.com.br
SIP tag: a1f0e83d
Call-ID: YWViODc0YTkwZTIyZTY5OGNkZTNkOWMxMDY4YjM5Mzc.
CSeq: 1 INVITE
Sequence Number: 1
Method: INVITE
Proxy-Authenticate: Digest qop=“auth”,nonce="8121cbb3b0ac7d04c576f9182f8966f8 ",realm="voxng03.gvt.com.br"
Authentication Scheme: Digest
QOP: "auth"
Nonce Value: "8121cbb3b0ac7d04c576f9182f8966f8"
Realm: "voxng03.gvt.com.br"
Content-Length: 0
Request-Line: ACK sip:YYYYYYYY@voxng03.gvt.com.br;transport=udp SIP/2.0
Method: ACK
Request-URI: sip:YYYYYYYY@voxng03.gvt.com.br;transport=udp
Request-URI User Part: YYYYYYYY
Request-URI Host Part: voxng03.gvt.com.br
[Resent Packet: False]
[Request Frame: 59]
[Response Time (ms): 50]
Message Header
Via: SIP/2.0/UDP 192.168.XXX.XXX:12846;branch=z9hG4bK-d8754z-b44572367bb4a973-1—d8754z-;rport
Transport: UDP
Sent-by Address: 192.168.XXX.XXX
Sent-by port: 12846
Branch: z9hG4bK-d8754z-b44572367bb4a973-1—d8754z-
RPort: rport
Max-Forwards: 70
To: sip:YYYYYYYY@voxng03.gvt.com.br;tag=SDi5k7399-b59feeac6822c5ef52d0c44d7a91fedf
SIP to address: sip:YYYYYYYY@voxng03.gvt.com.br
SIP to address User Part: 0XXXXXXXX
SIP to address Host Part: voxng03.gvt.com.br
SIP tag: SDi5k7399-b59feeac6822c5ef52d0c44d7a91fedf
From: "XXXXXXXX"sip:XXXXXXXX@voxng03.gvt.com.br;tag=a1 f0e83d
SIP Display info: "XXXXXXXX"
SIP from address: sip:XXXXXXXX@voxng03.gvt.com.br
SIP from address User Part: XXXXXXXX
SIP from address Host Part: voxng03.gvt.com.br
SIP tag: a1f0e83d
Call-ID: YWViODc0YTkwZTIyZTY5OGNkZTNkOWMxMDY4YjM5Mzc.
CSeq: 1 ACK
Sequence Number: 1
Method: ACK
Content-Length: 0
I have understood the problem is the To: in SIP header at the INVITE. Using X-Lite the To: uses voxng03.gvt.com.br but in Asterisk the To: is with the host parameter in sip.conf (www2.gvt.com.br) so I can’t figure this out.
Can anyone tell me what I can do to figure this out or maybe some wrong settings in my Asterisk.
Roberto Linck
robertolinck@gmail.com