[Spce-user] Upper Registration to Asterisk - transfers not working

[Theo]

A little more now - I upgraded the firmware of a SNOM phone. Following the
same process, upon pressing the transfer button I now get this:

U 2013/03/06 11:36:22.236542 41.177.66.116:22106 -> 196.41.123.113:5060
SIP/2.0 200 OK'
Via: SIP/2.0/UDP
196.41.123.113;branch=z9hG4bKaa2.bc57c1db3d0853f982c750b38e5926b2.0'
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
Record-Route:
<sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Contact: <sip:user400 at 192.168.10.88:5060
;rinstance=7e064df24c43ccf0;transport=UDP>'
To: <sip:user400 at domain.domain.com>;tag=d37f6717'
From: <sip:2721403 at domain.domain.com
>;tag=304E3AA0-51370E09000C6324-2EE19700'
Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
CSeq: 11 INVITE'
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE'
Content-Type: application/sdp'
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
User-Agent: Zoiper rev.11619'
Allow-Events: presence, kpml'
Content-Length: 185'
'
v=0'
o=Z 0 3 IN IP4 192.168.10.88'
s=Z'
c=IN IP4 192.168.10.88'
t=0 0'
m=audio 8000 RTP/AVP 8 101'
a=rtpmap:8 PCMA/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'
a=inactive'

#
U 2013/03/06 11:36:22.236896 127.0.0.1:5060 -> 127.0.0.1:5080
SIP/2.0 200 OK'
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK~IoIoamR;rport=5080'
Record-Route:
<sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Contact: <sip:user400 at 192.168.10.88:5060
;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP>'
To: <sip:user400 at domain.domain.com>;tag=d37f6717'
From: <sip:2721403 at domain.domain.com
>;tag=304E3AA0-51370E09000C6324-2EE19700'
Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
CSeq: 11 INVITE'
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE'
Content-Type: application/sdp'
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri'
User-Agent: Zoiper rev.11619'
Allow-Events: presence, kpml'
Content-Length: 233'
P-NGCP-Src-Ip: 41.177.66.116'
P-NGCP-Src-Port: 22106'
P-NGCP-Src-Proto: udp'
P-NGCP-Src-Af: 4'
'
v=0'
o=Z 0 3 IN IP4 192.168.10.88'
s=Z'
c=IN IP4 41.177.66.116'
t=0 0'
m=audio 8000 RTP/AVP 8 101'
a=rtpmap:8 PCMA/8000'
a=rtpmap:101 telephone-event/8000'
a=fmtp:101 0-15'
a=inactive'
a=direction:active'
a=oldmediaip:192.168.10.88'

#
U 2013/03/06 11:36:22.236968 127.0.0.1:5080 -> 127.0.0.1:5060
ACK sip:user400 at 192.168.10.88:5060;alias=41.177.66.116~22106~1;rinstance=7e064df24c43ccf0;transport=UDP
SIP/2.0'
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport'
From: <sip:2721403 at domain.domain.com
>;tag=304E3AA0-51370E09000C6324-2EE19700'
To: <sip:user400 at domain.domain.com>;tag=d37f6717'
CSeq: 11 ACK'
Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
Contact: <sip:127.0.0.1:5080>'
Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>,
<sip:196.41.123.113;lr;r2=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Max-Forwards: 68'
Content-Length: 0'
'

##
U 2013/03/06 11:36:22.237048 196.41.123.113:5060 -> 41.177.66.116:22106
ACK sip:user400 at 192.168.10.88:5060;rinstance=7e064df24c43ccf0;transport=UDP
SIP/2.0'
Record-Route:
<sip:196.41.123.113;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=304E3AA0-51370E09000C6324-2EE19700;ngcplb=yes>'
Via: SIP/2.0/UDP
196.41.123.113;branch=z9hG4bKaa2.04ec964b43561d47cbdd4d22812c93a8.0'
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKTbHrca0H;rport=5080'
From: <sip:2721403 at domain.domain.com
>;tag=304E3AA0-51370E09000C6324-2EE19700'
To: <sip:user400 at domain.domain.com>;tag=d37f6717'
CSeq: 11 ACK'
Call-ID: 51370e076928-0af4wphlmirc_b2b-1'
Max-Forwards: 67'
Content-Length: 0'
Contact: <sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080'>'


On Wed, Mar 6, 2013 at 9:41 AM, Theo <axessofficetheo at gmail.com> wrote:

> Hi
>
> ngrep-sip gives nothing - Here's the process on a SNOM phone.
>
> Call comes in. You put it on hold. You dial another extension and announce
> the call. All that shows up with ngrep-sip. Once the call is announced you
> press the transfer button. Nothing is recorded for that with ngrep sip
> whatsoever, which is obviously why it doesn't work :-(.
>
> Any ideas on that one?
>
>
> On Tue, Mar 5, 2013 at 8:59 AM, Jon Bonilla <jbonilla at sipwise.com> wrote:
>
>> El Mon, 4 Mar 2013 08:59:03 +0200
>> Theo <axessofficetheo at gmail.com> escribió:
>>
>> > Hi
>> >
>> > Not sure if I am breaking protocol by bumping - if I did - apologies. I
>> am
>> > almost ready to put this in as our SBC upper registration solution
>> (more to
>> > follow after that I hope :-) - but need to resolve the attended transfer
>> > problem. Any ideas on this? Sip Header filtering is off altogether (does
>> > that have any caveats?). Transfer working fine now for all phones
>> tested,
>> > but attended transfer not.
>> >
>>
>> ngrep-sip?
>>
>>
>
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