[asterisk-users] Aastra 6739i Support

[Joshua Tressler]

All,

I have multiple Asterisk servers in various locations running various
1.4 and 1.6 versions (lab and production) and am having trouble with a
new Aastra 6739i (3.0.1.2015) registering. Below is my request to
support and they have looked it over and don't see anything wrong:

 

Support, Can not get a 6739i to register with 3 different Asterisk
servers with varying configurations/versions but can get it to register
with 1 other in particular. From all of the SIP traces that I've
captured, after the 401 Unauthorized is sent from Asterisk to the phone
(including a WWW-Authenticate), the phone continues to send additional
REGISTER messages without the Authorization Digest response. The phones
DOES however send this back when it receives a message from the 1
server. I have gone through the configuration on all of the different
machines and can't find anything that would appear to change the
messages. Note also, that each of these 4 servers have various 6757i
phones in a nearly identical configuration working properly. Is there
something that has changed on the 6739i that would cause this?



See below for a snip of the debug on the Asterisk console:

 

REGISTER sip:10.100.250.10:5060 SIP/2.0

Via: SIP/2.0/UDP 172.16.99.161;branch=z9hG4bK46044b1bcfd826368

Max-Forwards: 70

From: <sip:5441 at 10.100.250.10:5060>;tag=be7edc2ddb

To: <sip:5441 at 10.100.250.10:5060>

Call-ID: 22fd19f44649a42f

CSeq: 26366 REGISTER

Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO

Allow-Events: talk, hold, conference

Contact:
<sip:5441 at 172.16.99.161:5060;transport=udp>;+sip.instance="<urn:uuid:000
00000-0000-1000-8000-00085D13D80D>";expires=3600

Supported: gruu

User-Agent: Aastra 6739i/3.0.1.38

Content-Length: 0

 

SIP/2.0 100 Trying

Via: SIP/2.0/UDP
10.100.250.10:5060;branch=z9hG4bK46044b1bcfd826368;received=172.16.99.16
1

From: <sip:5441 at 10.100.250.10:5060>;tag=be7edc2ddb

To: <sip:5441 at 10.100.250.10:5060>

Call-ID: 22fd19f44649a42f

CSeq: 26366 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:5441 at 10.100.250.10>

Content-Length: 0

 

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP
10.100.250.10:5060;branch=z9hG4bK46044b1bcfd826368;received=172.16.99.16
1

From: <sip:5441 at 10.100.250.10:5060>;tag=be7edc2ddb

To: <sip:5441 at 10.100.250.10:5060>;tag=as0913c4d5

Call-ID: 22fd19f44649a42f

CSeq: 26366 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

WWW-Authenticate: Digest algorithm=MD5, realm="ourrealm.net",
nonce="71fd68ea"

Content-Length: 0

 

REGISTER sip:10.100.250.10:5060 SIP/2.0

Via: SIP/2.0/UDP 172.16.99.161;branch=z9hG4bK46044b1bcfd826368

Max-Forwards: 70

From: <sip:5441 at 10.100.250.10:5060>;tag=be7edc2ddb

To: <sip:5441 at 10.100.250.10:5060>

Call-ID: 22fd19f44649a42f

CSeq: 26366 REGISTER

Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO

Allow-Events: talk, hold, conference

Contact:
<sip:5441 at 172.16.99.161:5060;transport=udp>;+sip.instance="<urn:uuid:000
00000-0000-1000-8000-00085D13D80D>";expires=3600

Supported: gruu

User-Agent: Aastra 6739i/3.0.1.38

Content-Length: 0

 

SIP/2.0 100 Trying

Via: SIP/2.0/UDP
10.100.250.10:5060;branch=z9hG4bK46044b1bcfd826368;received=172.16.99.16
1

From: <sip:5441 at 10.100.250.10:5060>;tag=be7edc2ddb

To: <sip:5441 at 10.100.250.10:5060>

Call-ID: 22fd19f44649a42f

CSeq: 26366 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:5441 at 10.100.250.10>

Content-Length: 0

 

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP
10.100.250.10:5060;branch=z9hG4bK46044b1bcfd826368;received=172.16.99.16
1

From: <sip:5441 at 10.100.250.10:5060>;tag=be7edc2ddb

To: <sip:5441 at 10.100.250.10:5060>;tag=as0913c4d5

Call-ID: 22fd19f44649a42f

CSeq: 26366 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

WWW-Authenticate: Digest algorithm=MD5, realm="ourrealm.net",
nonce="71fd68ea"

Content-Length: 0

 

 

SIP.conf:

[general]

context=default                

allowguest=no                   

allowoverlap=no               

realm=ourrealm.net           

bindport=5060                   

bindaddr=0.0.0.0               

srvlookup=yes                   

maxexpirey=900

limitonpeers=yes

notifyringing=yes

notifyhold=yes

callerid="Operator <0>"

subscribecontext=office-blf

rtcachefriends=yes

trustrpid=yes

generaterpid=yes

sendrpid=yes

 

DB Entry for Peer:

mysql> select * from sip_friends where name='5441'\G

*************************** 1. row ***************************

              id: 48

            name: 5441

            host: dynamic

             nat: no

            type: friend

     accountcode: NULL

        amaflags: NULL

      call-limit: 99

       callgroup: NULL

        callerid: Josh

  cancallforward: yes

     canreinvite: no

         context: office

       defaultip: NULL

        dtmfmode: inband

        fromuser: NULL

      fromdomain: NULL

        insecure: invi

        language: NULL

         mailbox: NULL

       md5secret: NULL

            deny: NULL

          permit: NULL

            mask: NULL

     musiconhold: NULL

     pickupgroup: NULL

         qualify: NULL

        regexten: NULL

     restrictcid: NULL

      rtptimeout: NULL

  rtpholdtimeout: NULL

          secret: abc123

          setvar: NULL

        disallow: all

           allow: ulaw

     fullcontact: 

          ipaddr: 

            port: 0

       regserver: NULL

      regseconds: 0

        username: 5441

     defaultuser: 

subscribecontext: NULL

 

 

Does anyone have any suggestions or run across a similar issue? The odd
thing is the box it is working with is running 1.6.2 and when the phone
is prompted to auth, it does send back with auth and will register
locally AND natted...

 

Thanks for the help!

 

Joshua Tressler

Network Engineer

Enhanced Telecommunications Corporation

Office: (812) 222-1020

Cell: (812) 593-0314

Email: jtressler at etc1.net 

 

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