[asterisk-users] Fwd: FW: loadstar and ok one

Vidura Senadeera vidurased at gmail.com
Wed Jun 22 18:01:14 CDT 2011


Hi All,



I have experiancing strenge issue with my production Asterisk system.



I'm using asterisk vertion 1.4.28 installed cent OS 5.



Issue decription.



I have SIP trunk from local carrier to their hosted PBX( broadsoft). Out
going calls over this trunk working fine and I can make a conversation with
landlines and mobiles but incomming not working. I can see calls are hitting
my IVR but no audio.

This system worked till yesterday without any issue.



Please find attached SIP traces from received from my carrier and what they
are saying is they not receiving proper information on SIP 200 message.



I'm attached my asterisk system traces also named asterisk SIP log.



Please looking to this and provide me help ASAP.



Thanks & Regards,
Vidura Senadeera,
Sri Lanka.
msn/yahoo/skype Ids - vidurased
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110623/0451b418/attachment.htm>
-------------- next part --------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.36:5061;branch=z9hG4bKooi3r32663oonttllrzts6lio
Call-ID: 63kothjzr4m4ks2jsimktm64ihrsrhlk at SoftX3000
From: <sip:717747766 at 10.10.1.36;user=phone>;tag=t4h4mk46-CC-24
To: <sip:600 at 10.20.1.66;user=phone>;tag=aprqtqg8kq3-mt4au32000020
CSeq: 1 CANCEL

-------------- next part --------------

---
Retransmitting #6 (NAT) to 10.8.55.194:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.55.194:5060;branch=z9hG4bKrjogsq209000rjk8s-3.1;received=10.8.55.194
From: <sip:773208775 at 10.8.55.194;user=phone>;tag=SD5j4u701-or643njn-CC-23
To: <sip:600 at 10.94.0.45;user=phone>;tag=as18171f50
Call-ID: SD5j4u701-8b8b9bbad4be07ce8971c1f3532859ab-ag220u0
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:600 at 10.94.0.45>
ontent-Type: application/sdp
Content-Length: 304

v=0
o=root 29336 29336 IN IP4 10.94.0.45
s=session
c=IN IP4 10.94.0.45
t=0 0
m=audio 10802 RTP/AVP 8 0 18 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
elastix*CLI>
<--- SIP read from 194.78.20.125:1092 --->
REGISTER sip:202.124.179.130 SIP/2.0
From: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81
To: "Tweco Wim"<sip:1103 at 202.124.179.130>
Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127
CSeq: 1860 REGISTER
Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbe-212749c0-7fc98295
Max-Forwards: 70
Supported: replaces,net2phone
User-Agent: LGE LIP 6812 v1.1.31s SN/00405A181B6A
Contact: <sip:1103 at 194.78.20.125:1093>
Expires: 60
Authorization: Digest username="1103",realm="asterisk",nonce="0dcbdd61",uri="sip:202.124.179.130",response="8ff1c4bfbd6ad4a9dc5db70640dfc431",algorithm=MD5
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 194.78.20.125 : 1092 (NAT)

<--- Transmitting (NAT) to 194.78.20.125:1092 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbe-212749c0-7fc98295;received=194.78.20.125
From: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81
To: "Tweco Wim"<sip:1103 at 202.124.179.130>
Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127
CSeq: 1860 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 194.78.20.125:1092 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbe-212749c0-7fc98295;received=194.78.20.125
From: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81
To: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=as5c46c474
Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127
CSeq: 1860 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3b3b16d9"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127' in 32000 ms (Method: REGISTER)
elastix*CLI>
<--- SIP read from 194.78.20.125:1092 --->
REGISTER sip:202.124.179.130 SIP/2.0
From: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81
To: "Tweco Wim"<sip:1103 at 202.124.179.130>
Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127
CSeq: 1861 REGISTER
Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbf-21274a88-2be0b1a2
Max-Forwards: 70
Supported: replaces,net2phone
User-Agent: LGE LIP 6812 v1.1.31s SN/00405A181B6A
Contact: <sip:1103 at 194.78.20.125:1093>
Expires: 60
Authorization: Digest username="1103",realm="asterisk",nonce="3b3b16d9",uri="sip:202.124.179.130",response="1d39bc50536a47031997a5c40cd65402",algorithm=MD5
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 194.78.20.125 : 1092 (NAT)

<--- Transmitting (NAT) to 194.78.20.125:1092 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbf-21274a88-2be0b1a2;received=194.78.20.125
From: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81
To: "Tweco Wim"<sip:1103 at 202.124.179.130>
Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127
CSeq: 1861 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
Reliably Transmitting (NAT) to 194.78.20.125:1092:
OPTIONS sip:1103 at 194.78.20.125:1093 SIP/2.0
Via: SIP/2.0/UDP 202.124.179.130:5060;branch=z9hG4bK70e928aa;rport
From: "Unknown" <sip:Unknown at 202.124.179.130>;tag=as4531a54a
To: <sip:1103 at 194.78.20.125:1093>
Contact: <sip:Unknown at 202.124.179.130>
Call-ID: 64502bb31876dd3858d19c3f5470f727 at 202.124.179.130
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 22 Jun 2011 22:50:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
elastix*CLI>
<--- Transmitting (NAT) to 194.78.20.125:1092 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbf-21274a88-2be0b1a2;received=194.78.20.125
From: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81
To: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=as5c46c474
Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127
CSeq: 1861 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Expires: 60
Contact: <sip:1103 at 194.78.20.125:1093>;expires=60
Date: Wed, 22 Jun 2011 22:50:33 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127' in 32000 ms (Method: REGISTER)
elastix*CLI>
<--- SIP read from 194.78.20.125:1092 --->
SIP/2.0 200 OK
From: "Unknown"<sip:Unknown at 202.124.179.130>;tag=as4531a54a
To: <sip:1103 at 194.78.20.125:1093>;tag=94cbf448-a20007f-13c4-87cbf-186a602e-87cbf
Call-ID: 64502bb31876dd3858d19c3f5470f727 at 202.124.179.130
CSeq: 102 OPTIONS
Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,SUBSCRIBE,OPTIONS
Via: SIP/2.0/UDP 202.124.179.130:5060;rport=5060;branch=z9hG4bK70e928aa
Supported: replaces
Content-Type: application/sdp
Content-Length: 258

v=0
o=LGEIPP 0 0 IN IP4 10.32.0.127
s=SIP Call
c=IN IP4 194.78.20.125
t=0 0
m=audio 23000 RTP/AVP 18 4 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
--- (10 headers 12 lines) ---
Really destroying SIP dialog '64502bb31876dd3858d19c3f5470f727 at 202.124.179.130' Method: OPTIONS
elastix*CLI>
<--- SIP read from 10.94.0.18:5974 --->

<------------->
    -- Executing [h at ivr-3:1] Hangup("SIP/Suntel OUT-0000002c", "") in new stack
  == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/Suntel OUT-0000002c'
Really destroying SIP dialog 'SD5j4u701-8b8b9bbad4be07ce8971c1f3532859ab-ag220u0' Method: INVITE
<--- SIP read from 10.8.55.194:5060 --->
INVITE sip:600 at 10.94.0.45;user=phone SIP/2.0^M
Via: SIP/2.0/UDP 10.8.55.194:5060;branch=z9hG4bKrjogsq209000rjk8s-3.1^M
Call-ID: SD5j4u701-8b8b9bbad4be07ce8971c1f3532859ab-ag220u0^M
From: <sip:773208775 at 10.8.55.194;user=phone>;tag=SD5j4u701-or643njn-CC-23^M
To: <sip:600 at 10.94.0.45;user=phone>^M
CSeq: 1 INVITE^M
Contact: <sip:773208775 at 10.8.55.194:5060;user=phone;transport=udp>^M
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER^M
User-Agent: Huawei SoftX3000 V300R010^M
Supported: 100rel^M
Max-Forwards: 69^M
Content-Length: 274^M
Content-Type: application/sdp^M
^M
v=0^M
o=HuaweiSoftX3000 11035821 11035821 IN IP4 10.8.55.194^M
s=Sip Call^M
c=IN IP4 10.8.55.194^M
t=0 0^M
m=audio 39858 RTP/AVP 8 0 18 97^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:18 G729/8000^M
a=rtpmap:97 telephone-event/8000^M
a=fmtp:97 0-15^M
a=fmtp:18 annexb=yes^M

<------------->
[Jun 23 04:20:11] VERBOSE[29370] logger.c: --- (13 headers 12 lines) ---
[Jun 23 04:20:11] VERBOSE[29370] logger.c: Ignoring this INVITE request
[Jun 23 04:20:11] VERBOSE[29370] logger.c:
<--- Transmitting (NAT) to 10.8.55.194:5060 --->
SIP/2.0 100 Trying^M
Via: SIP/2.0/UDP 10.8.55.194:5060;branch=z9hG4bKrjogsq209000rjk8s-3.1;received=10.8.55.194^M



More information about the asterisk-users mailing list