[asterisk-users] SIP response 500 "Server Internal Error"

[asterisk asterisk]

Hi,

I have problem in initiating an dial out call with  SIP response 500 "Server
Internal Error"

The sip debug as


  == Using SIP RTP CoS mark 5
Audio is at 113.253.226.92 port 18284
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.80.89.139:5060:
INVITE sip:27101271 at s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.226.92:5060;branch=z9hG4bK5b563aea;rport
Max-Forwards: 70
From: "IAX-cklee" <sip:35944101hk at s2hkbntel.net<sip%3A35944101hk at s2hkbntel.net>
>;tag=as4cffc48a
To: <sip:27101271 at s2hkbntel.net:5060>
Contact: <sip:35944101hk at 113.253.226.92 <sip%3A35944101hk at 113.253.226.92>>
Call-ID: 051db26e59f7163b2458cb9e67ff5a2f at s2hkbntel.net
CSeq: 102 INVITE
User-Agent: Asterisk
Remote-Party-ID: "IAX-cklee" <sip:6101 at s2hkbntel.net<sip%3A6101 at s2hkbntel.net>
>;privacy=off;screen=yes
Date: Mon, 09 Aug 2010 15:03:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 1216883305 1216883305 IN IP4 113.253.226.92
s=Asterisk PBX 1.6.2.10
c=IN IP4 113.253.226.92
t=0 0
m=audio 18284 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called 27101271 at hkbn2b

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 100 Trying
t: <sip:27101271 at s2hkbntel.net:5060>
f: "IAX-cklee" <sip:35944101hk at s2hkbntel.net<sip%3A35944101hk at s2hkbntel.net>
>;tag=as4cffc48a
i: 051db26e59f7163b2458cb9e67ff5a2f at s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.226.92:5060;rport;branch=z9hG4bK5b563aea
Server: MCS5x00_3.0
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 500 Server Internal Error
t: <sip:27101271 at s2hkbntel.net:5060>;tag=301677433
f: "IAX-cklee" <sip:35944101hk at s2hkbntel.net<sip%3A35944101hk at s2hkbntel.net>
>;tag=as4cffc48a
i: 051db26e59f7163b2458cb9e67ff5a2f at s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.226.92:5060;rport;branch=z9hG4bK5b563aea
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 500 "Server Internal Error" back from 203.80.89.139
Transmitting (NAT) to 203.80.89.139:5060:
ACK sip:27101271 at s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.226.92:5060;branch=z9hG4bK5b563aea;rport
Max-Forwards: 70
From: "IAX-cklee" <sip:35944101hk at s2hkbntel.net<sip%3A35944101hk at s2hkbntel.net>
>;tag=as4cffc48a
To: <sip:27101271 at s2hkbntel.net:5060>;tag=301677433
Contact: <sip:35944101hk at 113.253.226.92 <sip%3A35944101hk at 113.253.226.92>>
Call-ID: 051db26e59f7163b2458cb9e67ff5a2f at s2hkbntel.net
CSeq: 102 ACK
User-Agent: Asterisk
Remote-Party-ID: "IAX-cklee" <sip:6101 at s2hkbntel.net<sip%3A6101 at s2hkbntel.net>
>;privacy=off;screen=yes
Content-Length: 0

I have no idea how to make it work.

CK
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