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

Wed Mar 6 11:28:59 EST 2013

Tried with some other phones and it worked, so it seems specific to the
SNOM phones. Of course we have over a thousand SNOM phones out there so for
us to start using this we'd have to find a fix. My guess is that SNOM needs
to fix it, but don't have the required ammo nor knowledge to take them to
task on it (unless I am completely wrong to start off with). Aside from
that, I doubt whether they would fix that in a matter of days and I had
hoped to start implementing this as a solution.

Anything that can be done on the scpe to help with this? Any words of
wisdom I can share with SNOM?

On Wed, Mar 6, 2013 at 12:45 PM, Theo <axessofficetheo at gmail.com> wrote:

> And the SIP trace from the SNOM logs. I don't see any refer-to being sent
> from the phone....
>
>
> Sent to udp:196.41.123.113:5060 at 6/3/2013 12:18:02:571 (1617 bytes):
>
> INVITE sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080' SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.62:2050;branch=z9hG4bK-ttlvkk77nm7u;rport
> Route: <sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Route: <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Route: <sip:127.0.0.1:5062
> ;lr=on;ftag=v0z1vszklt;did=da1.e171;mpd=ii;rtpprx=yes;vsf=V3V3ZwIILyJUOU1ORmtBYW50Z2JXdXdnRUx4YVQ5TU5mS2E->
> From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
> To: <sip:400 at domain.domain.com
> ;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
> Call-ID: 513717c8ba36-755kmq45j2i5
> CSeq: 3 INVITE
> Max-Forwards: 70
> Contact: <sip:user403 at 192.168.10.62:2050;line=pxk7nlcg>;reg-id=1
> X-Serialnumber: 0004132FAC49
> P-Key-Flags: keys="3"
> User-Agent: snom300/8.7.3.19
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
> MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, 100rel, replaces, from-change
> Session-Expires: 3600;refresher=uas
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 489
>
> v=0
> o=root 1854415615 1854415616 IN IP4 192.168.10.62
> s=call
> c=IN IP4 192.168.10.62
> t=0 0
> m=audio 64268 RTP/AVP 9 0 8 3 99 108 18 101
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:fhqyKsE9KbQmap0NxeGKIFRZ3Yj2so3mw/32wC5U
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:99 G726-32/8000
> a=rtpmap:108 AAL2-G726-32/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendonly
> Received from udp:196.41.123.113:5060 at 6/3/2013 12:18:02:719 (386
> bytes):
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.10.62:2050
> ;received=41.177.66.139;branch=z9hG4bK-ttlvkk77nm7u;rport=14297
> From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
> To: <sip:400 at domain.domain.com
> ;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
> Call-ID: 513717c8ba36-755kmq45j2i5
> CSeq: 3 INVITE
> Server: Sipwise NGCP Proxy 2.X
> Content-Length: 0
>
> Received from udp:196.41.123.113:5060 at 6/3/2013 12:18:03:231 (1253
> bytes):
>
> SIP/2.0 200 OK
> Record-Route: <sip:127.0.0.1:5062;lr=on;ftag=v0z1vszklt;rtpprx=yes;mpd=ii>
> Record-Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Record-Route:
> <sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Via: SIP/2.0/UDP 192.168.10.62:2050
> ;received=41.177.66.139;branch=z9hG4bK-ttlvkk77nm7u;rport=14297
> From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
> To: <sip:400 at domain.domain.com
> ;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
> Call-ID: 513717c8ba36-755kmq45j2i5
> CSeq: 3 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE
> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
> User-Agent: Zoiper rev.11619
> Allow-Events: presence, kpml
> P-NGCP-Src-Ip: 41.177.66.116
> P-NGCP-Src-Port: 22106
> P-NGCP-Src-Proto: udp
> P-NGCP-Src-Af: 4
> Content-Type: application/sdp
> Content-Length: 254
> Contact: <sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080'>
>
> v=0
> o=Z 0 3 IN IP4 196.41.123.113
> s=Z
> c=IN IP4 196.41.123.113
> t=0 0
> m=audio 37342 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
> a=nortpproxy:yes
> Sent to udp:196.41.123.113:5060 at 6/3/2013 12:18:03:255 (741 bytes):
>
> ACK sip:ngcp-lb at 196.41.123.113:5060;ngcpct='sip:127.0.0.1:5080' SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.62:2050;branch=z9hG4bK-cntszxfllm77;rport
> Route: <sip:196.41.123.113;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Route: <sip:127.0.0.1;r2=on;lr=on;ftag=v0z1vszklt;nat=yes;ngcplb=yes>
> Route: <sip:127.0.0.1:5062
> ;lr=on;ftag=v0z1vszklt;did=da1.e171;mpd=ii;rtpprx=yes;vsf=V3V3ZwIILyJUOU1ORmtBYW50Z2JXdXdnRUx4YVQ5TU5mS2E->
> From: "AA Office" <sip:user403 at domain.domain.com>;tag=v0z1vszklt
> To: <sip:400 at domain.domain.com
> ;user=phone>;tag=6E486C78-513717CB00030190-2E40F700
> Call-ID: 513717c8ba36-755kmq45j2i5
> CSeq: 3 ACK
> Max-Forwards: 70
> Contact: <sip:user403 at 192.168.10.62:2050;line=pxk7nlcg>;reg-id=1
> Content-Length: 0
>
>
> On Wed, Mar 6, 2013 at 11:41 AM, Theo <axessofficetheo at gmail.com> wrote:
>
>> 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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