[Serusers] Possible bug in SER or Cisco GW?

Francesco Bottà francesco.botta at eutelia.it
Fri May 6 13:40:40 CEST 2005


Hi all,

I'm observing a very strange behaviour....when I'm calling from PSTN through Cisco GW to my SER Proxy, if in my voip dial-peer there's the CLI command 'dtmf-relay rtp-nte', and then in the SDP line appears this 'a' line 'a=fmtp:101 0-16', when arrives to my SER, SER itself is unable to process it...below the ngrep dump, where the actors are:

A.B.C.D: SER IP addr
E.F.G.H: GW Cisco SIP signalling addr
E.F.G.K: GW Cisco SIP media addr
W.W.W.W: my SIP ATA addr
U E.F.G.H:52007 -> A.B.C.D:5060

INVITE sip:0662293701 at sip.mydomain.org:5060 SIP/2.0.

Via: SIP/2.0/UDP E.F.G.H:5060;branch=z9hG4bK54A5503.

From: <sip:05751944945 at E.F.G.H>;tag=8E2C3094-11F5.

To: <sip:0662293701 at sip.mydomain.org>.

Date: Fri, 06 May 2005 11:35:27 GMT.

Call-ID: C73B2995-BD5911D9-971C9909-651E86BE at E.F.G.H.

Supported: rel1xx,timer.

Min-SE: 1800.

Cisco-Guid: 3342503125-3176731097-3023962122-4094721758.

User-Agent: Cisco-SIPGateway/IOS-12.x.

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

CSeq: 101 INVITE.

Max-Forwards: 10.

Remote-Party-ID: <sip:05751944945 at E.F.G.H>;party=calling;screen=yes;privacy=off.

Timestamp: 1115379327.

Contact: <sip:05751944945 at E.F.G.H:5060>.

Expires: 180.

Allow-Events: telephone-event.

Content-Type: application/sdp.

Content-Length: 536.

.

v=0.

o=CiscoSystemsSIP-GW-UserAgent 974 6362 IN IP4 E.F.G.H.

s=SIP Call.

c=IN IP4 E.F.G.K.

t=0 0.

m=audio 19340 RTP/AVP 18 8 0 4 3 98 99 2 125 101 19.

c=IN IP4 E.F.G.K.

a=rtpmap:18 G729/8000.

a=fmtp:18 annexb=yes.

a=rtpmap:8 PCMA/8000.

a=rtpmap:0 PCMU/8000.

a=rtpmap:4 G723/8000.

a=fmtp:4 annexa=yes.

a=rtpmap:3 GSM/8000.

a=rtpmap:98 G726-16/8000.

a=rtpmap:99 G726-24/8000.

a=rtpmap:2 G726-32/8000.

a=rtpmap:125 X-CCD/8000.

a=rtpmap:101 telephone-event/8000.

a=fmtp:101 0-16.

a=rtpmap:19 CN/8000.

a=direction:passive.

#

U A.B.C.D:5060 -> E.F.G.H:5060

SIP/2.0 100 trying -- your call is important to us.

Via: SIP/2.0/UDP E.F.G.H:5060;branch=z9hG4bK54A5503.

From: <sip:05751944945 at E.F.G.H>;tag=8E2C3094-11F5.

To: <sip:0662293701 at sip.mydomain.org>.

Call-ID: C73B2995-BD5911D9-971C9909-651E86BE at E.F.G.H.

CSeq: 101 INVITE.

Server: SPSexp (0.9.2 (i386/linux)).

Content-Length: 0.

Warning: 392 A.B.C.D:5060 "Noisy feedback tells: pid=9919 req_src_ip=E.F.G.H req_src_port=52007 in_uri=sip:0662293701 at sip.mydomain.org:5060 out_uri=sip:0662293701 at W.W.W.W:4848 via_cnt==1".

.

#

U A.B.C.D:5060 -> W.W.W.W:4848 33275 at 0:1480

........INVITE sip:0662293701 at W.W.W.W:4848 SIP/2.0.

Record-Route: <sip:A.B.C.D;ftag=8E2C3094-11F5;lr=on>.

Via: SIP/2.0/UDP A.B.C.D;branch=z9hG4bK60cd.c694de32.0.

Via: SIP/2.0/UDP E.F.G.H:5060;branch=z9hG4bK54A5503.

From: <sip:05751944945 at E.F.G.H>;tag=8E2C3094-11F5.

To: <sip:0662293701 at sip.mydomain.org>.

Date: Fri, 06 May 2005 11:35:27 GMT.

Call-ID: C73B2995-BD5911D9-971C9909-651E86BE at E.F.G.H.

Supported: rel1xx,timer.

Min-SE: 1800.

Cisco-Guid: 3342503125-3176731097-3023962122-4094721758.

User-Agent: Cisco-SIPGateway/IOS-12.x.

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

CSeq: 101 INVITE.

Max-Forwards: 9.

Remote-Party-ID: <sip:05751944945 at E.F.G.H>;party=calling;screen=yes;privacy=off.

Timestamp: 1115379327.

Contact: <sip:05751944945 at E.F.G.H:5060>.

Expires: 180.

Allow-Events: telephone-event.

Content-Type: application/sdp.

Content-Length: 536.

.

v=0.

o=CiscoSystemsSIP-GW-UserAgent 974 6362 IN IP4 E.F.G.H.

s=SIP Call.

c=IN IP4 E.F.G.K.

t=0 0.

m=audio 19340 RTP/AVP 18 8 0 4 3 98 99 2 125 101 19.

c=IN IP4 E.F.G.K.

a=rtpmap:18 G729/8000.

a=fmtp:18 annexb=yes.

a=rtpmap:8 PCMA/8000.

a=rtpmap:0 PCMU/8000.

a=rtpmap:4 G723/8000.

a=fmtp:4 annexa=yes.

a=rtpmap:3 GSM/8000.

a=rtpmap:98 G726-16/8000.

a=rtpmap:99 G726-24/8000.

a=rtpmap:2 G726-32/8000.

a=rtpmap:125 X-CCD/8000.

a=rtpmap:101 telephone-event/8000.

a=fmtp:101 0-16.

a=rtpmap:19 CN/80

#

U A.B.C.D:5060 -> W.W.W.W:4848 33276 at 0:1480

........INVITE sip:0662293701 at W.W.W.W:4848 SIP/2.0.

Record-Route: <sip:A.B.C.D;ftag=8E2C3094-11F5;lr=on>.

Via: SIP/2.0/UDP A.B.C.D;branch=z9hG4bK60cd.c694de32.0.

Via: SIP/2.0/UDP E.F.G.H:5060;branch=z9hG4bK54A5503.

From: <sip:05751944945 at E.F.G.H>;tag=8E2C3094-11F5.

To: <sip:0662293701 at sip.mydomain.org>.

Date: Fri, 06 May 2005 11:35:27 GMT.

Call-ID: C73B2995-BD5911D9-971C9909-651E86BE at E.F.G.H.

Supported: rel1xx,timer.

Min-SE: 1800.

Cisco-Guid: 3342503125-3176731097-3023962122-4094721758.

User-Agent: Cisco-SIPGateway/IOS-12.x.

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

CSeq: 101 INVITE.

Max-Forwards: 9.

Remote-Party-ID: <sip:05751944945 at E.F.G.H>;party=calling;screen=yes;privacy=off.

Timestamp: 1115379327.

Contact: <sip:05751944945 at E.F.G.H:5060>.

Expires: 180.

Allow-Events: telephone-event.

Content-Type: application/sdp.

Content-Length: 536.

.

v=0.

o=CiscoSystemsSIP-GW-UserAgent 974 6362 IN IP4 E.F.G.H.

s=SIP Call.

c=IN IP4 E.F.G.K.

t=0 0.

m=audio 19340 RTP/AVP 18 8 0 4 3 98 99 2 125 101 19.

c=IN IP4 E.F.G.K.

a=rtpmap:18 G729/8000.

a=fmtp:18 annexb=yes.

a=rtpmap:8 PCMA/8000.

a=rtpmap:0 PCMU/8000.

a=rtpmap:4 G723/8000.

a=fmtp:4 annexa=yes.

a=rtpmap:3 GSM/8000.

a=rtpmap:98 G726-16/8000.

a=rtpmap:99 G726-24/8000.

a=rtpmap:2 G726-32/8000.

a=rtpmap:125 X-CCD/8000.

a=rtpmap:101 telephone-event/8000.

a=fmtp:101 0-16.

a=rtpmap:19 CN/80





Did ever seen this strangeness??

Regards,



Verbal

Francesco Bottà
Service Engineering
Eutelia SpA
Arezzo
0575-1944945


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