[Asterisk-Users] WARNING[49159]
Martin Pycko
martinp at digium.com
Tue Oct 14 11:32:46 MST 2003
I don't see any warnings in your trace.
regards
Martin
On Tue, 14 Oct 2003, listas iPfone wrote:
> 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
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> _______________________________________________
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