[Asterisk-Users] Sip clients not longer registering
David Filion
dfilion at dotality.com
Thu Nov 4 10:54:33 MST 2004
Karl Brose wrote:
> The REGISTER requests that your SIP UAs are sending as listed are not
> requests to *register*, but request to *unregister*
> The contacts are '*' and expirations are '0'
> Granted that Asterisk doesn't do registrations correctly, but it does
> need a proper registration request with a contact and an Expires value
> > 0 to enter something in its location database.
>
>
>
>
>
>
>
> David Filion wrote:
>
>> Hi,
>>
>> We have been using Asterisk since version 0.9x with little or no
>> problems. However, for an unknow reasons, our sip clients can
>> nolonger register. We updated to Asterisk 1.0.2 hoping that would
>> solve the problem, but no luck.
>>
>> Here is the entry from sip.conf for one of our clients:
>>
>> [10012200]
>> host=dynamic
>> nat=yes
>> type=friend
>> mailbox=220 at 1001
>> username=10012200
>> secret=XXXXXXXX
>> context=1001
>> port=5060
>> quality=1000
>> dtmfmode=rfc2833
>> canreinvite=no
>> callerid="Muffin Man" <12223334444>
>> disallow=all
>> allow=g729
>>
>> The settings have been check in the sip client (a gs 486) and they
>> match. Below is a couple of sip sessions from when the user device
>> attempts to register:
>>
>> sip*CLI>
>>
>> Sip read:
>> REGISTER sip:111.222.333.444 SIP/2.0
>> Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4dd151b65533090
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>
>> Contact: *
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> Expires: 0
>> User-Agent: Grandstream HT486 1.0.5.16
>> Max-Forwards: 70
>> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
>> Content-Length: 0
>>
>>
>> 12 headers, 0 lines
>> Using latest request as basis request
>> Sending to 555.666.777.888 : 5060 (non-NAT)
>> Transmitting (NAT):
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as3055bbba
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Transmitting (NAT):
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as3055bbba
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> WWW-Authenticate: Digest realm="asterisk", nonce="626079d1"
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Scheduling destruction of call '15dd35988522bdf2 at 192.168.1.6' in
>> 15000 ms
>> sip*CLI>
>>
>> Sip read:
>> REGISTER sip:111.222.333.444 SIP/2.0
>> Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4dd151b65533090
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>
>> Contact: *
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> Expires: 0
>> User-Agent: Grandstream HT486 1.0.5.16
>> Max-Forwards: 70
>> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
>> Content-Length: 0
>>
>>
>> 12 headers, 0 lines
>> Using latest request as basis request
>> Sending to 555.666.777.888 : 5060 (NAT)
>> Transmitting (NAT):
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as3055bbba
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Transmitting (NAT):
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as3055bbba
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> WWW-Authenticate: Digest realm="asterisk", nonce="626079d1"
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Scheduling destruction of call '15dd35988522bdf2 at 192.168.1.6' in
>> 15000 ms
>> sip*CLI>
>>
>> Sip read:
>> REGISTER sip:111.222.333.444 SIP/2.0
>> Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4dd151b65533090
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>
>> Contact: *
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> Expires: 0
>> User-Agent: Grandstream HT486 1.0.5.16
>> Max-Forwards: 70
>> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
>> Content-Length: 0
>>
>>
>> 12 headers, 0 lines
>> Using latest request as basis request
>> Sending to 555.666.777.888 : 5060 (NAT)
>> Transmitting (NAT):
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as3055bbba
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Transmitting (NAT):
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as3055bbba
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> WWW-Authenticate: Digest realm="asterisk", nonce="626079d1"
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Scheduling destruction of call '15dd35988522bdf2 at 192.168.1.6' in
>> 15000 ms
>> sip*CLI>
>>
>> sip*CLI>
>>
>> Sip read:
>> REGISTER sip:111.222.333.444 SIP/2.0
>> Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4dd151b65533090
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>
>> Contact: *
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> Expires: 0
>> User-Agent: Grandstream HT486 1.0.5.16
>> Max-Forwards: 70
>> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
>> Content-Length: 0
>>
>>
>> 12 headers, 0 lines
>> Using latest request as basis request
>> Sending to 555.666.777.888 : 5060 (NAT)
>> Transmitting (NAT):
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as3055bbba
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Transmitting (NAT):
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as3055bbba
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> WWW-Authenticate: Digest realm="asterisk", nonce="626079d1"
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Scheduling destruction of call '15dd35988522bdf2 at 192.168.1.6' in
>> 15000 ms
>> sip*CLI>
>>
>> Sip read:
>> REGISTER sip:111.222.333.444 SIP/2.0
>> Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4dd151b65533090
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>
>> Contact: *
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> Expires: 0
>> User-Agent: Grandstream HT486 1.0.5.16
>> Max-Forwards: 70
>> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
>> Content-Length: 0
>>
>>
>> 12 headers, 0 lines
>> Using latest request as basis request
>> Sending to 555.666.777.888 : 5060 (NAT)
>> Transmitting (NAT):
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as3055bbba
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Transmitting (NAT):
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as3055bbba
>> Call-ID: 15dd35988522bdf2 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> WWW-Authenticate: Digest realm="asterisk", nonce="626079d1"
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Scheduling destruction of call '15dd35988522bdf2 at 192.168.1.6' in
>> 15000 ms
>> sip*CLI>
>>
>> sip*CLI>
>>
>> Sip read:
>> REGISTER sip:111.222.333.444 SIP/2.0
>> Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4b8b5f73edf0c59
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>
>> Contact: *
>> Call-ID: 1b1cabd87b023547 at 192.168.1.6
>> CSeq: 100 REGISTER
>> Expires: 0
>> User-Agent: Grandstream HT486 1.0.5.16
>> Max-Forwards: 70
>> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
>> Content-Length: 0
>>
>>
>> 12 headers, 0 lines
>> Using latest request as basis request
>> Sending to 555.666.777.888 : 5060 (non-NAT)
>> Transmitting (NAT):
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4b8b5f73edf0c59;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as7b9e3aed
>> Call-ID: 1b1cabd87b023547 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Transmitting (NAT):
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4b8b5f73edf0c59;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as7b9e3aed
>> Call-ID: 1b1cabd87b023547 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> WWW-Authenticate: Digest realm="asterisk", nonce="0cb6551e"
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Scheduling destruction of call '1b1cabd87b023547 at 192.168.1.6' in
>> 15000 ms
>> sip*CLI> si
>>
>> Sip read:
>> REGISTER sip:111.222.333.444 SIP/2.0
>> Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4b8b5f73edf0c59
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>
>> Contact: *
>> Call-ID: 1b1cabd87b023547 at 192.168.1.6
>> CSeq: 100 REGISTER
>> Expires: 0
>> User-Agent: Grandstream HT486 1.0.5.16
>> Max-Forwards: 70
>> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
>> Content-Length: 0
>>
>>
>> 12 headers, 0 lines
>> Using latest request as basis request
>> Sending to 555.666.777.888 : 5060 (NAT)
>> Transmitting (NAT):
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4b8b5f73edf0c59;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as7b9e3aed
>> Call-ID: 1b1cabd87b023547 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Transmitting (NAT):
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4b8b5f73edf0c59;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as7b9e3aed
>> Call-ID: 1b1cabd87b023547 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> WWW-Authenticate: Digest realm="asterisk", nonce="0cb6551e"
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Scheduling destruction of call '1b1cabd87b023547 at 192.168.1.6' in
>> 15000 ms
>> sip*CLI> sip
>>
>> Sip read:
>> REGISTER sip:111.222.333.444 SIP/2.0
>> Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4b8b5f73edf0c59
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>
>> Contact: *
>> Call-ID: 1b1cabd87b023547 at 192.168.1.6
>> CSeq: 100 REGISTER
>> Expires: 0
>> User-Agent: Grandstream HT486 1.0.5.16
>> Max-Forwards: 70
>> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
>> Content-Length: 0
>>
>>
>> 12 headers, 0 lines
>> Using latest request as basis request
>> Sending to 555.666.777.888 : 5060 (NAT)
>> Transmitting (NAT):
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4b8b5f73edf0c59;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as7b9e3aed
>> Call-ID: 1b1cabd87b023547 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Transmitting (NAT):
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/UDP
>> 555.666.777.888;branch=z9hG4bKd4b8b5f73edf0c59;received=555.666.777.888;rport=1024
>>
>> From: "David Filion" <sip:10012200 at 111.222.333.444>;tag=706ef4f92086984a
>> To: <sip:10012200 at 111.222.333.444>;tag=as7b9e3aed
>> Call-ID: 1b1cabd87b023547 at 192.168.1.6
>> CSeq: 100 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Contact: <sip:10012200 at 111.222.333.444>
>> WWW-Authenticate: Digest realm="asterisk", nonce="0cb6551e"
>> Content-Length: 0
>>
>>
>> to 555.666.777.888:1024
>> Scheduling destruction of call '1b1cabd87b023547 at 192.168.1.6' in
>> 15000 ms
>> 11 headers, 0 lines
>>
>>
>> The Asterisk server has a public ip (the above ips have been changed
>> from actual values). The client is behind a firewall using nat. Up
>> until this registration problem, nat never posed a problem.
>>
>> Any suggestions? So far we have not been able to identify any
>> external problems (firewall, os, hardware, etc) that maybe causing
>> this problem.
>>
>> David Filion
>>
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
Well here is a twist. Last night we reinstalled 1.0.1, no change to the
configs, and all the phones registered with out a problem! The only
reason we switched to 1.0.2 is because the phones started to lose
registration under 1.0.1.
David
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