[Asterisk-Users] WARNING[49159]

Tue Oct 14 10:27:13 MST 2003

Hi Martin!

here is:

s="Tue, 14 Oct 2003 17:55:00 GMT", <sip:33 at 192.168.0.31>;expires=3600
Expires: 159
Content-Length: 0

9 headers, 0 lines
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sip.microcity.com.br SIP/2.0
Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK31bef983
From: <sip:miklos at sip.microcity.com.br>;tag=as119e76aa
To: <sip:miklos at sip.microcity.com.br>
Call-ID: 17f7245e3138da193d974dc31d78c7bb at 127.0.0.1
CSeq: 199 REGISTER
User-Agent: Asterisk PBX
Expires: 160
Contact: <sip:33 at 192.168.0.31>
Event: registration
Content-length: 0

 (no NAT) to 200.251.160.60:5060
Sip read: CLI>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK31bef983
From: <sip:miklos at sip.microcity.com.br>;tag=as119e76aa
Call-ID: 17f7245e3138da193d974dc31d78c7bb at 127.0.0.1
CSeq: 199 REGISTER
Server: Intertex  ix66-release-2-0-4
To: <sip:miklos at sip.microcity.com.br>
Content-Length: 0

8 headers, 0 lines
Sip read: CLI>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK31bef983
From: <sip:miklos at sip.microcity.com.br>;tag=as119e76aa
To: <sip:miklos at sip.microcity.com.br>;tag=as0ebd4a9a
Call-ID: 17f7245e3138da193d974dc31d78c7bb at 127.0.0.1
CSeq: 199 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:miklos at 200.251.160.60>
Proxy-Authenticate: Digest realm="asterisk", nonce="5347caf4"
Content-Length: 0

11 headers, 0 lines
12 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sip.microcity.com.br SIP/2.0
Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK31bef983
From: <sip:miklos at sip.microcity.com.br>;tag=as119e76aa
To: <sip:miklos at sip.microcity.com.br>
Call-ID: 17f7245e3138da193d974dc31d78c7bb at 127.0.0.1
CSeq: 200 REGISTER
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="miklos", realm="asterisk",
algorithm="MD5", uri="sip:miklos at 200.251.160.60", nonce="5347caf4",
response="5af102c6033332abf311b8ec4c4eac72"
Expires: 160
Contact: <sip:33 at 192.168.0.31>
Event: registration
Content-length: 0

 (no NAT) to 200.251.160.60:5060
Sip read: CLI>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK31bef983
From: <sip:miklos at sip.microcity.com.br>;tag=as119e76aa
Call-ID: 17f7245e3138da193d974dc31d78c7bb at 127.0.0.1
CSeq: 200 REGISTER
Server: Intertex  ix66-release-2-0-4
To: <sip:miklos at sip.microcity.com.br>
Content-Length: 0

8 headers, 0 lines
Sip read: CLI>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK31bef983
From: <sip:miklos at sip.microcity.com.br>;tag=as119e76aa
To: <sip:miklos at sip.microcity.com.br>;tag=as0ebd4a9a
Call-ID: 17f7245e3138da193d974dc31d78c7bb at 127.0.0.1
CSeq: 200 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 160
Contact: <sip:miklos at 200.251.160.60>;expires=160
Date: Tue, 14 Oct 2003 17:24:44 GMT
Content-Length: 0

12 headers, 0 lines
Sip read: CLI>
REGISTER sip:35 at 192.168.0.31 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.33:5060
Call-ID: 00003a97-f0f0ca67 at 192.168.0.33
Contact: "35" <sip:35 at 192.168.0.33>
CSeq: 26288 REGISTER
From: <sip:35 at 192.168.0.31>;tag=00000952-f0f0f9a2
Supported: timer
To: "35" <sip:35 at 192.168.0.31>;tag=as2f9c027e
Proxy-Authorization: Digest
username="35",realm="asterisk",uri="sip:35 at 192.168.0.31",nonce="15dd12ff",re
sponse="e94ab01fd7e8aff59a9b787f2f2a9288"
User-Agent: ipDialog SipTone 1.2.0 rc V UA
Expires: 3600
Content-Length: 0

12 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.33 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.33:5060
From: <sip:35 at 192.168.0.31>;tag=00000952-f0f0f9a2
To: "35" <sip:35 at 192.168.0.31>;tag=as2f9c027e
Call-ID: 00003a97-f0f0ca67 at 192.168.0.33
CSeq: 26288 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:35 at 192.168.0.31>
Content-Length: 0

 to 192.168.0.33:5060
Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.33:5060
From: <sip:35 at 192.168.0.31>;tag=00000952-f0f0f9a2
To: "35" <sip:35 at 192.168.0.31>;tag=as2f9c027e
Call-ID: 00003a97-f0f0ca67 at 192.168.0.33
CSeq: 26288 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:35 at 192.168.0.31>
Proxy-Authenticate: Digest realm="asterisk", nonce="423ebf17"
Content-Length: 0

 to 192.168.0.33:5060
Sip read: CLI>
REGISTER sip:35 at 192.168.0.31 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.33:5060
Call-ID: 00003a97-f0f0ca67 at 192.168.0.33
Contact: "35" <sip:35 at 192.168.0.33>
CSeq: 26289 REGISTER
From: <sip:35 at 192.168.0.31>;tag=0000483a-f0f0b8ca
Supported: timer
To: "35" <sip:35 at 192.168.0.31>
Proxy-Authorization: Digest
username="35",realm="asterisk",uri="sip:35 at 192.168.0.31",nonce="423ebf17",re
sponse="31561f3026dd65147adf46891c6b8129"
User-Agent: ipDialog SipTone 1.2.0 rc V UA
Expires: 3600
Content-Length: 0
localhost*CLI>

12 headers, 0 lines
Using latest request as basis request
Sending to 192.168.0.33 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.33:5060
From: <sip:35 at 192.168.0.31>;tag=0000483a-f0f0b8ca
To: "35" <sip:35 at 192.168.0.31>;tag=as3028bf6d
Call-ID: 00003a97-f0f0ca67 at 192.168.0.33
CSeq: 26289 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:35 at 192.168.0.31>
Content-Length: 0

 to 192.168.0.33:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.33:5060
From: <sip:35 at 192.168.0.31>;tag=0000483a-f0f0b8ca
To: "35" <sip:35 at 192.168.0.31>;tag=as3028bf6d
Call-ID: 00003a97-f0f0ca67 at 192.168.0.33
CSeq: 26289 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 180
Contact: <sip:35 at 192.168.0.31>;expires=180
Date: Tue, 14 Oct 2003 16:30:06 GMT
Content-Length: 0

 to 192.168.0.33:5060
11 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:35 at 192.168.0.33 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK15438552
From: "asterisk" <sip:asterisk at 192.168.0.31>;tag=as787ccf10
To: <sip:35 at 192.168.0.33>
Contact: <sip:asterisk at 192.168.0.31>
Call-ID: 2681db8442cd81b016feefa53f342ddf at 192.168.0.31
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36

Messages-Waiting: no
Voicemail: 0/1
 (no NAT) to 192.168.0.33:5060
Sip read: CLI>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK15438552
Call-ID: 2681db8442cd81b016feefa53f342ddf at 192.168.0.31
Contact: "35" <sip:35 at 192.168.0.33>
CSeq: 102 NOTIFY
From: "asterisk" <sip:asterisk at 192.168.0.31>;tag=as787ccf10
Supported: timer
To: <sip:35 at 192.168.0.33>;tag=000002f8-f0f0f208
Server: ipDialog SipTone 1.2.0 rc V UAS
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,SUBSCRIBE,INFO,NOTIFY
Content-Length: 0

11 headers, 0 lines
localhost*CLI>

----- Original Message ----- 
From: "Martin Pycko" <martinp at digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, October 14, 2003 12:39 PM
Subject: Re: [Asterisk-Users] WARNING[49159]

> It means that your SIP device sends some SIP packets and we can't parse
> the CSeq numbers. Can you paste the 'sip debug' of that ?
>
> regards
> Martin
>
> On Tue, 14 Oct 2003, listas iPfone wrote:
>
> > Hi All
> >
> > I receive that warning message:
> >
> > WARNING[49159]: File chan_sip.c, Line 2220 (__transmit_response): Unable
to dete
> > rmine sequence number from ''
> >
> > What is it?
> >
> > There is some documentation with all error messages?
> >
> > thanks
> >
> > miklos
>
> _______________________________________________
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> Asterisk-Users at lists.digium.com
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>