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Aastra 6731i incoming call problem 1

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gok

gok

Technical User
Joined
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I have an IPO 8.018 that we have registered a Aastra 6731i as a SIP extension on the system. We can call from the Aastra to other extensions but when we try to call the Aastra it will go busy as soon as we try to answer. This is my first attempt at SIP phones and I got polycoms connected to it just fine.
I've googled and checked here but the Aastra info seems scarce, any help would be appreciated.
 
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Maybe a codec problem, use monitor to see what happens ( if you know how to interpret the logs ).

If it ain't dutch it ain't much
 
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When watching monitor, there is not codec information presented.
 
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Try these:

Screen Name: Extension name
Screen Name2:
Phone number: Extension number
Caller ID: Extension number
Auth Name: Extension name
Password: 1234
BLA:
Line: Generic


Outbound Proxy: IP of IPO
Outbound Proxy port: 5060

Registrar server: IP of IPO
Registrar port: 5060

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___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool )
______________________________________
 
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We had all that set except it was the proxy server set not the outbound proxy set. We made that change and we now lost the ability to call out from the phone.
 
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You also need the proxy server, just pointed the chenges you might need to try.

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___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool )
______________________________________
 
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Start the monitor tool, hit Ctrl+T the "clear all" the goto SIP tab and tick all options. Then press Ctrl+U enter the IPO's IP addr and the password should be password so leave it as it is.

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___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool )
______________________________________
 
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Ok we put the outbound proxy back info back in and we can now make calls from the phone but still cannot make calls to the phone. As soon as i get out of this meeting i'll try to check my monitor settings and copy that info here.
 
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In the IPO remove the extension and the user, use the password 0000 in the Aastra the let it connect. The IPO should auto create the extension and the number.

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___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool )
______________________________________
 
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If it doesn't auto create the extension try 1234 as the password in the Aastra.


Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool )
______________________________________
 
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We had already tried deleting the ext and lettig it auto create with the same results.

Here are the monitor logs when we tried to call from a digital station to the aastra.

8889996mS SIP Call Tx: phone
INVITE sip:4999@192.168.42.4;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.42.1:5060;rport;branch=z9hG4bK9921b2229798a2948b8b025bddc97193
From: "Extn201" <sip:201@192.168.42.1>;tag=51eb124a68965bd8
To: <sip:4999@192.168.42.1;transport=udp>
Call-ID: e721053f89567400a886308fddaaf6a3@192.168.42.1
CSeq: 587699267 INVITE
Contact: "Extn201" <sip:201@192.168.42.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, SUBSCRIBE, REGISTER, PUBLISH, UPDATE
Content-Type: application/sdp
Supported: timer
P-Asserted-Identity: "Extn201" <sip:201@192.168.42.1:5060>
Content-Length: 298

v=0
o=UserA 480495283 3995838770 IN IP4 192.168.42.1
s=Session SDP
c=IN IP4 192.168.42.1
t=0 0
m=audio 49152 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
8889996mS SIP Tx: UDP 192.168.42.1:5060 -> 192.168.42.4:5060
INVITE sip:4999@192.168.42.4;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.42.1:5060;rport;branch=z9hG4bK9921b2229798a2948b8b025bddc97193
From: "Extn201" <sip:201@192.168.42.1>;tag=51eb124a68965bd8
To: <sip:4999@192.168.42.1;transport=udp>
Call-ID: e721053f89567400a886308fddaaf6a3@192.168.42.1
CSeq: 587699267 INVITE
Contact: "Extn201" <sip:201@192.168.42.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, SUBSCRIBE, REGISTER, PUBLISH, UPDATE
Content-Type: application/sdp
Supported: timer
P-Asserted-Identity: "Extn201" <sip:201@192.168.42.1:5060>
Content-Length: 298

v=0
o=UserA 480495283 3995838770 IN IP4 192.168.42.1
s=Session SDP
c=IN IP4 192.168.42.1
t=0 0
m=audio 49152 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
8890064mS SIP Rx: UDP 192.168.42.4:5060 -> 192.168.42.1:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.42.1:5060;rport=5060;branch=z9hG4bK9921b2229798a2948b8b025bddc97193;received=192.168.42.1
From: "Extn201" <sip:201@192.168.42.1>;tag=51eb124a68965bd8
To: <sip:4999@192.168.42.1;transport=udp>;tag=2981392499
Call-ID: e721053f89567400a886308fddaaf6a3@192.168.42.1
CSeq: 587699267 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "4999" <sip:4999@192.168.42.4:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D31A674>"
Server: Aastra 6731i/2.6.0.2010
Supported: gruu, path
Content-Length: 0

8890066mS SIP Call Rx: phone
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.42.1:5060;rport=5060;branch=z9hG4bK9921b2229798a2948b8b025bddc97193;received=192.168.42.1
From: "Extn201" <sip:201@192.168.42.1>;tag=51eb124a68965bd8
To: <sip:4999@192.168.42.1;transport=udp>;tag=2981392499
Call-ID: e721053f89567400a886308fddaaf6a3@192.168.42.1
CSeq: 587699267 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "4999" <sip:4999@192.168.42.4:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D31A674>"
Server: Aastra 6731i/2.6.0.2010
Supported: gruu, path
Content-Length: 0

8895149mS RES: Mon 23/4/2012 16:22:18 FreeMem=61248976(1) CMMsg=3 (6) Buff=5200 959 998 7423 5 Links=6737
8895149mS RES2: IP 500 V2 8.0(18) Tasks=46 RTEngine=0 CMRTEngine=0 ExRTEngine=0 Timer=61 Poll=0 Ready=0 CMReady=0 CMQueue=0 VPNNQueue=0 Monitor=1 SSA=2 ASC=1 SYS=MNTD OPT=UMNT SDSPD=2034
8897779mS SIP Rx: UDP 192.168.42.4:5060 -> 192.168.42.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.42.1:5060;rport=5060;branch=z9hG4bK9921b2229798a2948b8b025bddc97193;received=192.168.42.1
From: "Extn201" <sip:201@192.168.42.1>;tag=51eb124a68965bd8
To: <sip:4999@192.168.42.1;transport=udp>;tag=2981392499
Call-ID: e721053f89567400a886308fddaaf6a3@192.168.42.1
CSeq: 587699267 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "4999" <sip:4999@192.168.42.4:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D31A674>"
Server: Aastra 6731i/2.6.0.2010
Supported: gruu, path, timer, replaces
Content-Type: application/sdp
Content-Length: 325

v=0
o=MxSIP 0 0 IN IP4 192.168.42.4
s=SIP Call
c=IN IP4 192.168.42.4
t=0 0
m=audio 3000 RTP/AVP 0 8 18 4 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
8897783mS SIP Call Rx: phone
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.42.1:5060;rport=5060;branch=z9hG4bK9921b2229798a2948b8b025bddc97193;received=192.168.42.1
From: "Extn201" <sip:201@192.168.42.1>;tag=51eb124a68965bd8
To: <sip:4999@192.168.42.1;transport=udp>;tag=2981392499
Call-ID: e721053f89567400a886308fddaaf6a3@192.168.42.1
CSeq: 587699267 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "4999" <sip:4999@192.168.42.4:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D31A674>"
Server: Aastra 6731i/2.6.0.2010
Supported: gruu, path, timer, replaces
Content-Type: application/sdp
Content-Length: 325

v=0
o=MxSIP 0 0 IN IP4 192.168.42.4
s=SIP Call
c=IN IP4 192.168.42.4
t=0 0
m=audio 3000 RTP/AVP 0 8 18 4 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
8897785mS SIP Call Tx: phone
ACK sip:4999@192.168.42.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.42.1:5060;rport;branch=z9hG4bKb971b69b33533eb6508fe9ea4d4b5c6e
From: "Extn201" <sip:201@192.168.42.1>;tag=51eb124a68965bd8
To: "Extn4999" <sip:4999@192.168.42.1;transport=udp>;tag=2981392499
Call-ID: e721053f89567400a886308fddaaf6a3@192.168.42.1
CSeq: 587699267 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, SUBSCRIBE, REGISTER, PUBLISH, UPDATE
Content-Length: 0

8897785mS SIP Tx: UDP 192.168.42.1:5060 -> 192.168.42.4:5060
ACK sip:4999@192.168.42.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.42.1:5060;rport;branch=z9hG4bKb971b69b33533eb6508fe9ea4d4b5c6e
From: "Extn201" <sip:201@192.168.42.1>;tag=51eb124a68965bd8
To: "Extn4999" <sip:4999@192.168.42.1;transport=udp>;tag=2981392499
Call-ID: e721053f89567400a886308fddaaf6a3@192.168.42.1
CSeq: 587699267 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, SUBSCRIBE, REGISTER, PUBLISH, UPDATE
Content-Length: 0

8897792mS SIP Call Tx: phone
BYE sip:4999@192.168.42.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.42.1:5060;rport;branch=z9hG4bK9a68c1deb2bc0eb9e72df17c646ab92f
From: "Extn201" <sip:201@192.168.42.1>;tag=51eb124a68965bd8
To: "Extn4999" <sip:4999@192.168.42.1;transport=udp>;tag=2981392499
Call-ID: e721053f89567400a886308fddaaf6a3@192.168.42.1
CSeq: 587699268 BYE
Contact: "Extn201" <sip:201@192.168.42.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, SUBSCRIBE, REGISTER, PUBLISH, UPDATE
Supported: timer
Content-Length: 0

8897793mS SIP Tx: UDP 192.168.42.1:5060 -> 192.168.42.4:5060
BYE sip:4999@192.168.42.4:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.42.1:5060;rport;branch=z9hG4bK9a68c1deb2bc0eb9e72df17c646ab92f
From: "Extn201" <sip:201@192.168.42.1>;tag=51eb124a68965bd8
To: "Extn4999" <sip:4999@192.168.42.1;transport=udp>;tag=2981392499
Call-ID: e721053f89567400a886308fddaaf6a3@192.168.42.1
CSeq: 587699268 BYE
Contact: "Extn201" <sip:201@192.168.42.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, SUBSCRIBE, REGISTER, PUBLISH, UPDATE
Supported: timer
Content-Length: 0

8897796mS PRN: Config Write Wake Up
8897942mS SIP Rx: UDP 192.168.42.4:5060 -> 192.168.42.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.42.1:5060;rport=5060;branch=z9hG4bK9a68c1deb2bc0eb9e72df17c646ab92f;received=192.168.42.1
From: "Extn201" <sip:201@192.168.42.1>;tag=51eb124a68965bd8
To: "Extn4999" <sip:4999@192.168.42.1;transport=udp>;tag=2981392499
Call-ID: e721053f89567400a886308fddaaf6a3@192.168.42.1
CSeq: 587699268 BYE
Server: Aastra 6731i/2.6.0.2010
Content-Length: 0

8897943mS SIP Call Rx: phone
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.42.1:5060;rport=5060;branch=z9hG4bK9a68c1deb2bc0eb9e72df17c646ab92f;received=192.168.42.1
From: "Extn201" <sip:201@192.168.42.1>;tag=51eb124a68965bd8
To: "Extn4999" <sip:4999@192.168.42.1;transport=udp>;tag=2981392499
Call-ID: e721053f89567400a886308fddaaf6a3@192.168.42.1
CSeq: 587699268 BYE
Server: Aastra 6731i/2.6.0.2010
Content-Length: 0

 
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Turn off SDP in the Aastra.

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool )
______________________________________
 
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what is SDP?
 
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Change the RTP settings.

Set the "Force RFC2833 Out-of-Band DTMF" field to "Enabled"
Set the "DTMF Method" field to RTP
And set the STP encryption to "SRTP disabled"

The Autentication name cannot start with a number it needs to start with at least 1 letter.

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool )
______________________________________
 
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RTP settings are already set that way.
The authentication name does begin with a letter as well.
 
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Try a different sip client like 3CX or X-Lite, use that with the settings from the extension you like to register to if that makes a difference. I could be a firmware issue in the Aastra or in the IPO.

What firmware do you have in the IPO?

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool )
______________________________________
 
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8.018 in the IPO. We have a Polycom Soundpoint IP650 and it is registering and can make and receive calls just fine.

We just defaulted the IP config to try again fresh and it does the same thing.
 
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before doing that we downgraded the ipo to 7.0.27 and it worked. We then upgraded to 8.0.42 and it still works. Looks like there was a fix in the maintenance release that says "No PAI added to outgoing SIP INVITE when the PAI option is set to [Use Internal Data]"

thank you Bas1234 for all your assistance.
 
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