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@sip.mydomain.org:5060 SIP/2.0.
Via: SIP/2.0/UDP E.F.G.H:5060;branch=z9hG4bK54A5503.
From: <sip:05751944945@E.F.G.H>;tag=8E2C3094-11F5.
To: <sip:0662293701@sip.mydomain.org>.
Date: Fri, 06 May 2005 11:35:27 GMT.
Call-ID: C73B2995-BD5911D9-971C9909-651E86BE(a)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@E.F.G.H>;party=calling;screen=yes;privacy=off.
Timestamp: 1115379327.
Contact: <sip:05751944945@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@E.F.G.H>;tag=8E2C3094-11F5.
To: <sip:0662293701@sip.mydomain.org>.
Call-ID: C73B2995-BD5911D9-971C9909-651E86BE(a)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@sip.mydomain.org:5060
out_uri=sip:0662293701@W.W.W.W:4848 via_cnt==1".
.
#
U A.B.C.D:5060 -> W.W.W.W:4848 33275@0:1480
........INVITE sip:0662293701@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@E.F.G.H>;tag=8E2C3094-11F5.
To: <sip:0662293701@sip.mydomain.org>.
Date: Fri, 06 May 2005 11:35:27 GMT.
Call-ID: C73B2995-BD5911D9-971C9909-651E86BE(a)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@E.F.G.H>;party=calling;screen=yes;privacy=off.
Timestamp: 1115379327.
Contact: <sip:05751944945@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@0:1480
........INVITE sip:0662293701@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@E.F.G.H>;tag=8E2C3094-11F5.
To: <sip:0662293701@sip.mydomain.org>.
Date: Fri, 06 May 2005 11:35:27 GMT.
Call-ID: C73B2995-BD5911D9-971C9909-651E86BE(a)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@E.F.G.H>;party=calling;screen=yes;privacy=off.
Timestamp: 1115379327.
Contact: <sip:05751944945@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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