[cisco-voip] cisco ccm 4.0 and ser?
Dinesh
dinesh at imcb.a-star.edu.sg
Tue Jun 28 07:05:05 EDT 2005
Yes teodor,
I am experimenting with SIP Express Router to call manager. I have made a
sip trunk on my ccm 4.0. It was talking fine to my asterisk server. There
was a howto
http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration.
The problem comes when I change the ip address of the sip device name on ccm
4.0 to my SIP Express Router for SER.
In my sip config I have it was
# send out 222 prefix to
if (uri=~"^sip:222[0-9].*@.*") {
if (!is_user_in("From", "ld")) {
sl_send_reply("403", "LD required");
break;
};
setflag(1);
strip(3);
rewritehostport("10.217.84.11:5060");
if (!t_relay()) {
sl_reply_error();
};
break;
};
So I restart my SER and I do a ngrep
./ngrep -p port 5060
And I get this in my ngrep output.
#
U 137.132.43.121:5060 -> 203.125.17.228:5060
INVITE sip:2229804 at imcb.a-star.edu.sg SIP/2.0..Via: SIP/2.0/UDP
137.132.43.121:5060;rport;branch=z9hG4bK12A15B3E34B04EA58C1E9B3
FFA79C2A9..From: Dinesh Birlasekaran
<sip:65869804 at imcb.a-star.edu.sg>;tag=522275532..To:
<sip:2229804 at imcb.a-star.edu.sg>..Con
tact: <sip:65869804 at 137.132.43.121:5060>..Call-ID:
A3D83DEB-D6EA-4784-A84B-0460E7CCCFFA at 137.132.43.121..CSeq: 46335 INVITE..Max
-Forwards: 70..Content-Type: application/sdp..User-Agent: X-Lite release
1103m..Content-Length: 301....v=0..o=65869804 39381197
39381227 IN IP4 137.132.43.121..s=X-Lite..c=IN IP4 137.132.43.121..t=0
0..m=audio 8000 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0 pcm
u/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98
iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/
8000..a=fmtp:101 0-15..
#
U 203.125.17.228:5060 -> 137.132.43.121:5060
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
137.132.43.121:5060;rport=5060;branch=z9hG4bK12A15B3E34B04
EA58C1E9B3FFA79C2A9..From: Dinesh Birlasekaran
<sip:65869804 at imcb.a-star.edu.sg>;tag=522275532..To:
<sip:2229804 at imcb.a-star.ed
u.sg>..Call-ID: A3D83DEB-D6EA-4784-A84B-0460E7CCCFFA at 137.132.43.121..CSeq:
46335 INVITE..Server: Sip EXpress router (0.8.12 (i3
86/linux))..Content-Length: 0..Warning: 392 203.125.17.228:5060 "Noisy
feedback tells: pid=21513 req_src_ip=137.132.43.121 req
_src_port=5060 in_uri=sip:2229804 at imcb.a-star.edu.sg
out_uri=sip:9804 at 10.217.84.11:5060 via_cnt==1"....
#
U 137.132.43.116:5060 -> 203.125.17.228:5060
SIP/2.0 400 Bad Request - 'Malformed/Missing Record Route'..Via:
SIP/2.0/UDP 203.125.17.228;branch=z9hG4bK6cd1.17544ff1.0,SIP/2
.0/UDP
137.132.43.121:5060;rport=5060;branch=z9hG4bK12A15B3E34B04EA58C1E9B3FFA79C2A
9..From: Dinesh Birlasekaran <sip:65869804 at i
mcb.a-star.edu.sg>;tag=522275532..To:
<sip:2229804 at imcb.a-star.edu.sg>;tag=16898095..Call-ID:
A3D83DEB-D6EA-4784-A84B-0460E7CCC
FFA at 137.132.43.121..CSeq: 46335 INVITE..Content-Length: 0....
#
U 203.125.17.228:5060 -> 137.132.43.121:5060
SIP/2.0 400 Bad Request - 'Malformed/Missing Record Route'..Via:
SIP/2.0/UDP 137.132.43.121:5060;rport=5060;branch=z9hG4bK12A15
B3E34B04EA58C1E9B3FFA79C2A9..From: Dinesh Birlasekaran
<sip:65869804 at imcb.a-star.edu.sg>;tag=522275532..To: <sip:2229804 at imcb.a
-star.edu.sg>;tag=16898095..Call-ID:
A3D83DEB-D6EA-4784-A84B-0460E7CCCFFA at 137.132.43.121..CSeq: 46335
INVITE..Content-Length: 0
....
#
U 137.132.43.121:5060 -> 203.125.17.228:5060
ACK sip:2229804 at imcb.a-star.edu.sg SIP/2.0..Via: SIP/2.0/UDP
137.132.43.121:5060;rport;branch=z9hG4bK12A15B3E34B04EA58C1E9B3FFA
79C2A9..From: Dinesh Birlasekaran
<sip:65869804 at imcb.a-star.edu.sg>;tag=522275532..To:
<sip:2229804 at imcb.a-star.edu.sg>;tag=168
98095..Contact: <sip:65869804 at 137.132.43.121:5060>..Call-ID:
A3D83DEB-D6EA-4784-A84B-0460E7CCCFFA at 137.132.43.121..CSeq: 46335 A
CK..Max-Forwards: 70..Content-Length: 0....
#
U 137.132.43.121:5060 -> 203.125.17.228:5060
Any help is appreciated.
When I change my SER config to send to my Asterisk server, I also get some
error message from my asterisk when I call the number 2223001, which is a
sip phone inside the asterisk network.
owl*CLI>
-- Executing AbsoluteTimeout("SIP/imcb.a-star.edu.sg-09827d88", "15") in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/imcb.a-star.edu.sg-09827d88", "") in new
stack
== Spawn extension (from-sip-external, 3001, 2) exited non-zero on
'SIP/imcb.a-star.edu.sg-09827d88'
-- Executing AbsoluteTimeout("SIP/imcb.a-star.edu.sg-09827d88", "15") in
new stack
-- Set Absolute Timeout to 15
-- Executing Congestion("SIP/imcb.a-star.edu.sg-09827d88", "") in new
stack
== Spawn extension (from-sip-external, h, 2) exited non-zero on
'SIP/imcb.a-star.edu.sg-09827d88'
Dinesh.
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Teodor Georgiev
Sent: Tuesday, June 28, 2005 6:52 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] cisco ccm 4.0 and ser?
Do you mean SIP Express Router for SER?
If so, then - yes.
On Tuesday 28 June 2005 12:55, Dinesh wrote:
> Hello,
>
>
>
> Is there anyone here that has made ccm 4.0 and ser talk with one another?
>
>
>
> Regards,
>
>
>
> Dinesh Birlasekaran
> Network Engineer,
> ComIT, Institute of Molecular and Cell Biology
> 61 Biopolis Drive, Singapore 138673
> HP : 92962676 DID : 65869804 Fax : 67791117 Email :
> dinesh at imcb.a-star.edu.sg
> WWW: www.imcb.a-star.edu.sg
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