Am I Cursed ?

[talltrees]

Am I Cursed ? Update… Resolved see post at bottom

I have one extension at a remote site, it fails to log in to the PIAF server properly, you can call it but it does not call out…
It does some times but then it fails….
I have even set it up on a one to one IP mapping so it uses port 5060 to log into the server…
Still no joy, other remote extensions are OK..
What can be happening ?
I have filled out
Proxy Server
Outbound Proxy Server
Registrar Server
with the external IP of the server

I have put in
Phone Number
Caller ID
Authentication Name
as the asterisk extension number..

Unit is 6757i Aastra
I am going insane… this should just work…
Any idea's ?
Maybe a Stun Server ? Any suggestions for Stun servers ?
Thanks…

----------------------
Solution…
http://pbxinaflash.com/community/threads/aastra-6757i-57i-sip-bug-discovered.6041/#post-36698
-------------------

 

[jmullinix]

Have you manually set the port used on the phone to 5060? I don't remember the name of the field, but it is blank by default.

 

[madberry]

Check the DTMF mode make sure it is set to the same setting. My phone where using DTMF and the PIAF was using rfc2833. When I set the phone to rfc2833 I was able to make calls. Before that I could only receive them.

 

[talltrees]

Yes 5060, it shows up in sip info on the server as port 15752 due to NAT I suppose..
on the 6757i you can set DTMF to RTP SIP INFO or BOTH I set to RTP
RTP encryption is disabled…
the call just fails.

here is a sip trace…

pbx*CLI> sip set debug peer 502
SIP Debugging Enabled for IP: 88.***.***.140:15752
pbx*CLI>
<--- SIP read from 88.***.***.140:15752 --->
INVITE sip:07*******75@***.***.***.53:5060 SIP/2.0
Via: SIP/2.0/UDP 192.68.1.40:5060;branch=z9hG4bK581335de56ec5d7e6.b5add2fc4a9dcb9e9
Max-Forwards: 70
From: "502" <sip:502@***.***.***.53:5060>;tag=75b1459841
To: "07*******75" <sip:07*******75@***.***.***.53:5060>
Call-ID: b65d02092a9f4fad
CSeq: 17221 INVITE
Allow: *INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "502" <sip:[email protected]:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D10B769>"
Supported: gruu, path, timer, 100rel, replaces
User-Agent: Aastra 57i/2.5.2.1010
Content-Type: application/sdp
Content-Length: 593

v=0
o=MxSIP 0 0 IN IP4 192.68.1.40
s=SIP Call
c=IN IP4 192.68.1.40
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSuppff - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (14 headers 25 lines) ---
Using INVITE request as basis request - b65d02092a9f4fad

<--- Reliably Transmitting (NAT) to 88.***.***.140:15752 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.68.1.40:5060;branch=z9hG4bK581335de56ec5d7e6.b5add2fc4a9dcb9e9;received=88.***.***.140
From: "502" <sip:502@***.***.***.53:5060>;tag=75b1459841
To: "07*******75" <sip:07*******75@***.***.***.53:5060>;tag=as30bd5ad1
Call-ID: b65d02092a9f4fad
CSeq: 17221 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3385701b"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'b65d02092a9f4fad' in 32000 ms (Method: INVITE)
Found user '502'

<--- SIP read from 88.***.***.140:15752 --->
ACK sip:07*******75@***.***.***.53:5060 SIP/2.0
Via: SIP/2.0/UDP 192.68.1.40:5060;branch=z9hG4bK581335de56ec5d7e6.b5add2fc4a9dcb9e9
Max-Forwards: 70
From: "502" <sip:502@***.***.***.53:5060>;tag=75b1459841
To: "07*******75" <sip:07*******75@***.***.***.53:5060>;tag=as30bd5ad1
Call-ID: b65d02092a9f4fad
CSeq: 17221 ACK
User-Agent: Aastra 57i/2.5.2.1010
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Reliably Transmitting (NAT) to 88.***.***.140:15752:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP ***.***.***.53:5060;branch=z9hG4bK658399da;rport
From: "Unknown" <sip:Unknown@***.***.***.53>;tag=as5926bf92
To: <sip:[email protected]:5060;transport=udp>
Contact: <sip:Unknown@***.***.***.53>
Call-ID: 649e94d442d90d44255f206639f4e496@***.***.***.53
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 14 Nov 2009 10:49:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Reliably Transmitting (NAT) to 88.***.***.140:15752:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP ***.***.***.53:5060;branch=z9hG4bK29611e07;rport
From: "Unknown" <sip:Unknown@***.***.***.53>;tag=as3781a393
To: <sip:[email protected]:5060;transport=udp>
Contact: <sip:Unknown@***.***.***.53>
Call-ID: 495ae8284ce3a594093ee13611ccd156@***.***.***.53
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 14 Nov 2009 10:49:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

 

[talltrees]

in sip_nat.conf I have this set up in both the servers I have…
I am sure this is OK…

externip=88.***.***.***
localnet=192.168.1.0/255.255.255.0
bindaddr=0.0.0.0

second server

externip=88.***.***.***
localnet=192.68.1.0/255.255.255.0

 

[jmullinix]

Is the second server behind the same nat as the phone. If so, this could be a firewall problem. Some firewall/gateways don't handle sip to multiple internal devices very well.

 

[talltrees]

Yes but the server has direct one to one NAT, the phone is on one to many...

 

[talltrees]

I have removed all other sip connections for the router which is a Xyzel 661H-D1
If anyone knows of any issues with these routers I'd like to know...

 

[talltrees]

Here is a VOIP analysis from wireshark of the data from tcpdump
I'm not getting very far… any idea's anyone…
The call fails…
|Time | 88.**.***.140 |
| | | 192.168.1.121 |
|15.161 | INVITE SDP ( g711U g729 BV16 BV32 L16 PCMU PCM...16 G726-16 G) |SIP From: sip:501@87.***.***.53:5060 To:sip:07**********5@87.***.***.53:5060
| |(15752) ------------------> (5060) |
|15.161 | 407 Proxy Authentication Required |SIP Status
| |(15752) <------------------ (5060) |
|15.283 | ACK | |SIP Request
| |(15752) ------------------> (5060) |
|15.304 | INVITE SDP ( g711U g729 BV16 BV32 L16 PCMU PCM...16 G726-16 G) |SIP From: sip:501@87.***.***.53:5060 To:sip:07**********5@87.***.***.53:5060
| |(15752) ------------------> (5060) |
|15.791 | INVITE SDP ( g711U g729 BV16 BV32 L16 PCMU PCM...16 G726-16 G) |SIP From: sip:501@87.***.***.53:5060 To:sip:07**********5@87.***.***.53:5060
| |(15752) ------------------> (5060) |
|16.792 | INVITE SDP ( g711U g729 BV16 BV32 L16 PCMU PCM...16 G726-16 G) |SIP From: sip:501@87.***.***.53:5060 To:sip:07**********5@87.***.***.53:5060
| |(15752) ------------------> (5060) |
|18.793 | INVITE SDP ( g711U g729 BV16 BV32 L16 PCMU PCM...16 G726-16 G) |SIP From: sip:501@87.***.***.53:5060 To:sip:07**********5@87.***.***.53:5060
| |(15752) ------------------> (5060) |

 

[talltrees]

OK,
so in sip_nat I have
canreinvite=no
localnet
externip=
nat=route (tried yes)

external phone is natted
asterisk server has 121 NAT.
Snom phone can ring out no problem Aastra phone cannot… What can I do why do Snom work where Aastra do not, I think this is a nat issue but this is crazy…

Anyone help ?

 

[jmullinix]

Are you running the most recent firmware on the Aastra phone? I have had that cause problems with them.

 

[talltrees]

I am should I down grade ?

 

[talltrees]

and like all happy stories this has a happy end…
http://pbxinaflash.com/community/threads/aastra-6757i-57i-sip-bug-discovered.6041/#post-36698