HI,
I have an interconexion with a proxy-server
registrar IPTEL to a Cisco GATEWAY AS5300 for the PSTN.
When I make a call by a Cisco
IP phone, I have no problem but when I make a call with a call-box(Same as
Cisco ATA) or a softphone Xpro (XTEN), I can speak 20 s and after the call is
cut.
I have analysed this problem,
the problem is after the SIP ringing frame, the softphone send a SIP frame ACK
with a bad request-line.
IP ADDRESS 80.118.128.1 : CISCO
AS5300 GATEWAY
IP ADDRESS 80.118.128.5 :
SERVER IPTEL
The RED parameter in the first DEBUG
is BAD, he as to be as the same of the second debug.
That's my debug of the SOFT
PHONE: BAD ACKNOWLEDGE
#########################################################
U 80.118.128.1:5060 ->
80.118.128.5:5060
SIP/2.0 200 OK..Via:
SIP/2.0/UDP 80.118.128.5;branch=z9hG4bKcce9.7a6be6f3.0,SIP/2.0/UDP
62.39.69.2:21791;rport=21791;branch=z9hG4bKB0A9B5A4D70A4FF8A745CCDD512FED41..Fr
om:
ID:
475E7EC7-5CA5-457D-8F0D-BCF6B8FA6209@192.168.18.21..Server:
Cisco-SIPGateway/IOS-12.x..CSeq: 24531 INVITE..Allow: INVITE, OPTIONS, BYE,
CANCEL, ACK, PRACK, COMET,
REFER, SUBSCRIBE,
NOTIFY, INFO..Allow-Events: telephone-event..Contact:
<sip:0156384185@80.118.128.1:5060>..Record-Route:
<sip:0156384185@80.118.128.5;ftag=1756293558;
lr=on>..Content-Type:
application/sdp..Content-Length: 256....v=0..o=CiscoSystemsSIP-GW-UserAgent 16
..m=audio 18194 RTP/AVP
18 101..c=IN IP4 80.118.128.1..a=rtpmap:18 G729/8000..a=fmtp:18
annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..
#
U 80.118.128.5:5060 ->
62.39.69.2:21791
SIP/2.0 200 OK..Via:
SIP/2.0/UDP
62.39.69.2:21791;rport=21791;branch=z9hG4bKB0A9B5A4D70A4FF8A745CCDD512FED41..From:
=1756293558..To:
<sip:0156384185@sip.vivaction.net>;tag=E855BD4-163A..Date: Thu, 02 Sep 2004
12:39:19 GMT..Call-ID: 475E7EC7-5CA5-457D-8F0D-BCF6B8FA6209@192.168.18.21.
.Server:
Cisco-SIPGateway/IOS-12.x..CSeq: 24531 INVITE..Allow: INVITE, OPTIONS, BYE,
CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO..Allow-Events:
telephone
-event..Contact:
<sip:0156384185@80.118.128.1:5060>..Record-Route:
<sip:0156384185@80.118.128.5;ftag=1756293558;lr=on>..Content-Type:
application/sdp..Content-Length:
256....v=0..o=CiscoSystemsSIP-GW-UserAgent 16
a=rtpmap:18
G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-15..
####
U 62.39.69.2:21791 ->
80.118.128.5:5060
ACK sip:0156384185@80.118.128.1:5060 SIP/2.0..Via:
SIP/2.0/UDP
62.39.69.2:21791;rport;branch=z9hG4bKDDE6A61442CF4B3985D47F7A633E9A9F..From:
1@sip.vivaction.net>;tag=1756293558..To:
<sip:0156384185@sip.vivaction.net>;tag=E855BD4-163A..Contact:
<sip:0170708661@62.39.69.2:21791>..Route: <sip:0156384185@80.118
.128.5;ftag=1756293558;lr=on>..Call-ID:
475E7EC7-5CA5-457D-8F0D-BCF6B8FA6209@192.168.18.21..CSeq: 24531
ACK..Max-Forwards: 70..Content-Length: 0....
######
That's my debug of the IP
PHONE: GOOD ACKNOWLEDGE
U 80.118.128.1:5060 ->
80.118.128.5:5060
SIP/2.0 200 OK..Via:
SIP/2.0/UDP 80.118.128.5;branch=z9hG4bK119a.2eb3c633.0,SIP/2.0/UDP
62.39.70.245:20105;branch=z9hG4bK21610a53..From: "ip.phone" <sip:0170708662@sip
.vivaction.net>;tag=000f8f58915f002e0e896df1-21abefc7..To:
<sip:0156384185@sip.vivaction.net>;tag=EB213D4-
915f0025-7e34225d-1cdf9dae@192.168.254.12..Server: Cisco-SIPGateway/IOS-12.x..CSeq:
102 INVITE..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIB
E, NOTIFY,
INFO..Allow-Events: telephone-event..Contact:
<sip:0156384185@80.118.128.1:5060>..Record-Route:
<sip:0156384185@80.118.128.5;ftag=000f8f58915f002e0e896df1-2
1abefc7;lr=on>..Content-Type: application/sdp..Content-Length:
258....v=0..o=CiscoSystemsSIP-GW-UserAgent 1735
8.1..t=0 0..m=audio
18760 RTP/AVP 18 101..c=IN IP4 80.118.128.1..a=rtpmap:18 G729/8000..a=fmtp:18
annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..
#
U 80.118.128.5:5060 ->
62.39.70.245:20105
SIP/2.0 200 OK..Via:
SIP/2.0/UDP 62.39.70.245:20105;branch=z9hG4bK21610a53..From:
"ip.phone" <sip:0170708662@sip.vivaction.net>;tag=000f8f58915f002e0e896df1-21abefc7..
To:
<sip:0156384185@sip.vivaction.net>;tag=EB213D4-
o-SIPGateway/IOS-12.x..CSeq: 102 INVITE..Allow: INVITE, OPTIONS, BYE, CANCEL,
ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO..Allow-Events:
telephone-event..Contact
:
<sip:0156384185@80.118.128.1:5060>..Record-Route:
<sip:0156384185@80.118.128.5;ftag=000f8f58915f002e0e896df1-21abefc7;lr=on>..Content-Type:
application/sdp..Content-
Length:
258....v=0..o=CiscoSystemsSIP-GW-UserAgent 1735
18.128.1..a=rtpmap:18
G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-15..
####
U 62.39.70.245:20600 ->
80.118.128.5:5060
ACK sip:0156384185@80.118.128.5:5060;ftag=000f8f58915f002e0e896df1-21abefc7;lr=on
SIP/2.0..Via: SIP/2.0/UDP 62.39.70.245:20105;branch=z9hG4bK47003517..From:
"ip.phone"
<sip:0170708662@sip.vivaction.net>;tag=000f8f58915f002e0e896df1-21abefc7..To:
<sip:0156384185@sip.vivaction.net>;tag=EB213D4-
-1cdf9dae@192.168.254.12..CSeq: 102 ACK..User-Agent: CSCO/6..Route:
<sip:0156384185@80.118.128.1:5060>..Content-Length: 0....
#
U 80.118.128.5:5060 ->
80.118.128.1:5060
ACK
sip:0156384185@80.118.128.1:5060 SIP/2.0..Max-Forwards: 10..Record-Route:
<sip:0156384185@80.118.128.5;ftag=000f8f58915f002e0e896df1-21abefc7;lr=on>..Via:
SIP/2.0/
UDP
80.118.128.5;branch=0..Via: SIP/2.0/UDP
62.39.70.245:20105;branch=z9hG4bK47003517..From: "ip.phone"
<sip:0170708662@sip.vivaction.net>;tag=000f8f58915f002e0e896df1
-21abefc7..To:
<sip:0156384185@sip.vivaction.net>;tag=EB213D4-
tent-Length: 0....
Thans for your help