Hi,
I have an interesting problem that the dial out via sip always generates 603
error
The following is the sip debug
Your help is appreciated.
CK
== Using SIP RTP CoS mark 5
-- Executing [998560848 at DLPN_DP1:1] Dial("SIP/6100-0000005b",
"SIP/13398560848 at hkbn2b") in new stack
== Using SIP RTP CoS mark 5
Audio is at 113.253.230.26 port 11316
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:13398560848 at s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport
Max-Forwards: 70
From: "cklee at mobile"
<sip:35944101hk at s2hkbntel.net<sip%3A35944101hk at
s2hkbntel.net>>;tag=as1d554c43
To: <sip:13398560848 at s2hkbntel.net:5060>
Contact: <sip:35944101hk at 113.253.230.26 <sip%3A35944101hk at
113.253.230.26>>
Call-ID: 34c9241622c72c7d26b13fdc22d95530 at s2hkbntel.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Date: Sun, 15 Aug 2010 13:47:41 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 2083113394 2083113394 IN IP4 113.253.230.26
s=Asterisk PBX 1.6.2.10
c=IN IP4 113.253.230.26
t=0 0
m=audio 11316 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 13398560848 at hkbn2b
<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 100 Trying
t: <sip:13398560848 at s2hkbntel.net:5060>
f: "cklee at mobile" <sip:35944101hk at
s2hkbntel.net<sip%3A35944101hk at s2hkbntel.net>>;tag=as1d554c43
i: 34c9241622c72c7d26b13fdc22d95530 at s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.230.26:5060
;received=113.253.230.70;rport;branch=z9hG4bK575022bd
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/2.0 487 Request Terminated
t: <sip:13398560848 at s2hkbntel.net:5060>;tag=1652716799
f: "cklee at mobile" <sip:35944101hk at
s2hkbntel.net<sip%3A35944101hk at s2hkbntel.net>>;tag=as1d554c43
i: 34c9241622c72c7d26b13fdc22d95530 at s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.230.26:5060
;received=113.253.230.70;rport;branch=z9hG4bK575022bd
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 203.80.89.139:5060:
ACK sip:13398560848 at s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport
Max-Forwards: 70
From: "cklee at mobile"
<sip:35944101hk at s2hkbntel.net<sip%3A35944101hk at
s2hkbntel.net>>;tag=as1d554c43
To: <sip:13398560848 at s2hkbntel.net:5060>;tag=1652716799
Contact: <sip:35944101hk at 113.253.230.26 <sip%3A35944101hk at
113.253.230.26>>
Call-ID: 34c9241622c72c7d26b13fdc22d95530 at s2hkbntel.net
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.10
Content-Length: 0
---
Scheduling destruction of SIP dialog '
34c9241622c72c7d26b13fdc22d95530 at s2hkbntel.net' in 6400 ms (Method:
INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [998560848 at DLPN_DP1:2] Hangup("SIP/6100-0000005b",
"") in
new stack
== Spawn extension (DLPN_DP1, 998560848, 2) exited non-zero on
'SIP/6100-0000005b'
Really destroying SIP dialog '34c9241622c72c7d26b13fdc22d95530 at
s2hkbntel.net'
Method: INVITE
ns*CLI> sip set debug off
SIP Debugging Disabled
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Hi,
I have an interesting problem that the dial out via sip always generates 603
error
The following is the sip debug
Your help is appreciated.
CK
== Using SIP RTP CoS mark 5
-- Executing [998560848 at DLPN_DP1:1] Dial("SIP/6100-0000005b",
"SIP/13398560848 at hkbn2b") in new stack
== Using SIP RTP CoS mark 5
Audio is at 113.253.230.26 port 11316
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:13398560848 at s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport
Max-Forwards: 70
From: "cklee at mobile"
<sip:35944101hk at s2hkbntel.net<sip%3A35944101hk at
s2hkbntel.net>>;tag=as1d554c43
To: <sip:13398560848 at s2hkbntel.net:5060>
Contact: <sip:35944101hk at 113.253.230.26 <sip%3A35944101hk at
113.253.230.26>>
Call-ID: 34c9241622c72c7d26b13fdc22d95530 at s2hkbntel.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Date: Sun, 15 Aug 2010 13:47:41 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 2083113394 2083113394 IN IP4 113.253.230.26
s=Asterisk PBX 1.6.2.10
c=IN IP4 113.253.230.26
t=0 0
m=audio 11316 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 13398560848 at hkbn2b
<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 100 Trying
t: <sip:13398560848 at s2hkbntel.net:5060>
f: "cklee at mobile" <sip:35944101hk at
s2hkbntel.net<sip%3A35944101hk at s2hkbntel.net>>;tag=as1d554c43
i: 34c9241622c72c7d26b13fdc22d95530 at s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.230.26:5060
;received=113.253.230.70;rport;branch=z9hG4bK575022bd
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/2.0 487 Request Terminated
t: <sip:13398560848 at s2hkbntel.net:5060>;tag=1652716799
f: "cklee at mobile" <sip:35944101hk at
s2hkbntel.net<sip%3A35944101hk at s2hkbntel.net>>;tag=as1d554c43
i: 34c9241622c72c7d26b13fdc22d95530 at s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.230.26:5060
;received=113.253.230.70;rport;branch=z9hG4bK575022bd
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 203.80.89.139:5060:
ACK sip:13398560848 at s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.230.26:5060;branch=z9hG4bK575022bd;rport
Max-Forwards: 70
From: "cklee at mobile"
<sip:35944101hk at s2hkbntel.net<sip%3A35944101hk at
s2hkbntel.net>>;tag=as1d554c43
To: <sip:13398560848 at s2hkbntel.net:5060>;tag=1652716799
Contact: <sip:35944101hk at 113.253.230.26 <sip%3A35944101hk at
113.253.230.26>>
Call-ID: 34c9241622c72c7d26b13fdc22d95530 at s2hkbntel.net
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.10
Content-Length: 0
---
Scheduling destruction of SIP dialog '
34c9241622c72c7d26b13fdc22d95530 at s2hkbntel.net' in 6400 ms (Method:
INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [998560848 at DLPN_DP1:2] Hangup("SIP/6100-0000005b",
"") in
new stack
== Spawn extension (DLPN_DP1, 998560848, 2) exited non-zero on
'SIP/6100-0000005b'
Really destroying SIP dialog '34c9241622c72c7d26b13fdc22d95530 at
s2hkbntel.net'
Method: INVITE
ns*CLI> sip set debug off
SIP Debugging Disabled
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