I got this Problem currently. Feel very strange
I can display in CLI that I have registered the Server.
My number describe below is virtual number(due to privacy).
XFT01*CLI> sip show registry
Host Username Refresh State Reg.Time
202.0.179.3:5060 85235015001 105 Registered Sun, 09 Sep 2007 19:07:30
However when I make a call from outside world. I can forward to internal extension. However when pickup call… It Drop. Has the same problem.
Audio is at 220.232.XXX.XXX port 15728
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (no NAT) to 202.0.179.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bK10e7955c3;received=202.0.179.3
From: sip:65011001@202.0.179.3;user=phone;tag=7ac82219
To: sip:85235015001@220.232.XXX.XXX;user=phone;tag=as4129607b
Call-ID: 2dc12890932eb00fa38c65153a633b65@sx3000
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:85235015001@220.232.XXX.XXX
Content-Type: application/sdp
Content-Length: 309
v=0
o=root 8279 8279 IN IP4 220.232.XXX.XXX
s=session
c=IN IP4 220.232.XXX.XXX
t=0 0
m=audio 15728 RTP/AVP 0 8 4 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
– <SIP/85235015001-09857798> Playing ‘vm-theperson’ (language ‘en’)
XFT01*CLI>
<— SIP read from 202.0.179.3:5060 —>
ACK sip:85235015001@220.232.XXX.XXX:5060;user=phone SIP/2.0
From: sip:65011001@202.0.179.3;user=phone;tag=7ac82219
To: sip:85235015001@220.232.XXX.XXX;user=phone;tag=as4129607b
CSeq: 1 ACK
Call-ID: 2dc12890932eb00fa38c65153a633b65@sx3000
Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bK28a8b6b3a
Max-Forwards: 70
Content-Length: 0
<------------->
— (8 headers 0 lines) —
XFT01*CLI>
<— SIP read from 202.0.179.3:5060 —>
BYE sip:85235015001@220.232.XXX.XXX:5060;user=phone SIP/2.0
From: sip:65011001@202.0.179.3;user=phone;tag=7ac82219
To: sip:85235015001@220.232.XXX.XXX;user=phone;tag=as4129607b
CSeq: 2 BYE
Call-ID: 2dc12890932eb00fa38c65153a633b65@sx3000
Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bK81827d9e4
Reason: Q.850;cause=“100”;text="unknown"
Max-Forwards: 70
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Sending to 202.0.179.3 : 5060 (no NAT)
<— Transmitting (no NAT) to 202.0.179.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bK81827d9e4;received=202.0.179.3
From: sip:65011001@202.0.179.3;user=phone;tag=7ac82219
To: sip:85235015001@220.232.XXX.XXX;user=phone;tag=as4129607b
Call-ID: 2dc12890932eb00fa38c65153a633b65@sx3000
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:85235015001@220.232.XXX.XXX
Content-Length: 0
When make outgoing call, it reply with 503 Services Unavailable. But it was funny that there is no problem if I use Linksys RTP300. Can make call sucessfully.
Audio is at 220.232.XXX.XXX port 16506
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 202.0.179.3:5060:
INVITE sip:18503@202.0.179.3 SIP/2.0
Via: SIP/2.0/UDP 220.232.XXX.XXX:5060;branch=z9hG4bK179b8f6f;rport
From: “phone” sip:85235011500@220.232.XXX.XXX;tag=as2f5bd6e1
To: sip:18503@202.0.179.3
Contact: sip:85235015001@220.232.149.192
all-ID: 092b449521d9b2c9296ef1d412954a76@220.232.XXX.XXX
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 09 Sep 2007 11:15:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 335
v=0
o=root 8279 8279 IN IP4 220.232.XXX.XXX
s=session
c=IN IP4 220.232.XXX.XXX
t=0 0
m=audio 16506 RTP/AVP 0 8 4 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
-- Called 85235015001/65011001
XFT01*CLI>
<— SIP read from 202.0.179.3:5060 —>
SIP/2.0 100 Trying
From: “phone” sip:85235015001@220.232.XXX.XXX;tag=as2f5bd6e1
To: sip:65011001@202.0.179.3
CSeq: 102 INVITE
Call-ID: 092b449521d9b2c9296ef1d412954a76@220.232.XXX.XXX
Via: SIP/2.0/UDP 220.232.XXX.XXX:5060;branch=z9hG4bK179b8f6f;rport=5060
Content-Length: 0
<------------->
— (7 headers 0 lines) —
tmhf01*CLI>
<— SIP read from 202.0.179.3:5060 —>
SIP/2.0 407 Proxy Authentication Required
From: “Kitman” sip:85235015001@220.232.XXX.XXX;tag=as2f5bd6e1
To: sip:65011001@202.0.179.3;tag=6809f112
CSeq: 102 INVITE
Call-ID: 092b449521d9b2c9296ef1d412954a76@220.232.XXX.XXX
Via: SIP/2.0/UDP 220.232.XXX.XXX:5060;branch=z9hG4bK179b8f6f;rport=5060
Proxy-Authenticate: Digest realm=“huawei.com”,nonce=“19:15:22:5902”, stale=false,algorithm=MD5
Reason: Q.850;cause=“0”;text="unknown"
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Transmitting (no NAT) to 202.0.179.3:5060:
ACK sip:65011001@202.0.179.3 SIP/2.0
Via: SIP/2.0/UDP 220.232.XXX.XXX:5060;branch=z9hG4bK179b8f6f;rport
From: “phone” sip:85235015001@220.232.XXX.XXX;tag=as2f5bd6e1
To: sip:65011001@202.0.179.3;tag=6809f112
Contact: sip:85235015001@220.232.XXX.XXX
Call-ID: 092b449521d9b2c9296ef1d412954a76@220.232.XXX.XXX
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Audio is at 220.232.XXX.XXX port 16506
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 202.0.179.3:5060:
INVITE sip:65011001@202.0.179.3 SIP/2.0
Via: SIP/2.0/UDP 220.232.XXX.XXX:5060;branch=z9hG4bK04b80928;rport
From: “phone” sip:85235015001@220.232.XXX.XXX;tag=as2f5bd6e1
To: sip:65011001@202.0.179.3
Contact: sip:85235015001@220.232.XXX.XXX
Call-ID: 092b449521d9b2c9296ef1d412954a76@220.232.XXX.XXX
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username=“85235015001”, realm=“huawei.com”, algorithm=MD5, uri="sip:65011001@202.0.179.3", nonce=“19:15:22:5902”, response=“1cd6ee8d4c9cdfe7dfdee18cb55ddf26”, opaque=""
Date: Sun, 09 Sep 2007 11:15:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 335
v=0
o=root 8279 8280 IN IP4 220.232.XXX.XXX
s=session
c=IN IP4 220.232.XXX.XXX
t=0 0
m=audio 16506 RTP/AVP 0 8 4 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
tmhf01*CLI>
<— SIP read from 202.0.179.3:5060 —>
SIP/2.0 100 Trying
From: “phone” sip:85235015001@220.232.XXX.XXX;tag=as2f5bd6e1
To: sip:65011001@202.0.179.3
CSeq: 103 INVITE
Call-ID: 092b449521d9b2c9296ef1d412954a76@220.232.XXX.XXX
Via: SIP/2.0/UDP 220.232.XXX.XXX:5060;branch=z9hG4bK04b80928;rport=5060
Content-Length: 0
<------------->
— (7 headers 0 lines) —
tmhf01*CLI>
<— SIP read from 202.0.179.3:5060 —>
SIP/2.0 503 Service Unavailable
From: “phone” sip:85235015001@220.232.XXX.XXX;tag=as2f5bd6e1
To: sip:65011001@202.0.179.3;tag=1b6d1533
CSeq: 103 INVITE
Call-ID: 092b449521d9b2c9296ef1d412954a76@220.232.XXX.XXX
Via: SIP/2.0/UDP 220.232.XXX.XXX:5060;branch=z9hG4bK04b80928;rport=5060
Reason: Q.850;cause=“98”;text="unknown"
Content-Length: 0
My provider calls CMPHONE Hong Kong. I am using Asterisk 1.4.11
Config as follows:
register => 85235015001:secretpwd@202.0.179.3
[35015001-cm]
type=peer
host=202.0.179.3
fromdomain=huawei.com
fromuser=85235015001
secret=secretpwd
username=85235015001
insecure=very
context=from-pstn
authname=85235015001
dtmfmode=auto
canreinvite=no
Is there anyone could provide me some hints or workable config for this?