[cisco-voip] Call manager 7 SIP trunk

Sun Sep 12 13:13:32 EDT 2010

MTP enabled with Enable Outbound FastStart did the trick.

Thanks!

From: matthn at gmail.com [mailto:matthn at gmail.com] On Behalf Of Nick Matthews
Sent: Sunday, September 12, 2010 12:39 PM
To: Mike Olivere
Cc: Ki Wi; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Call manager 7 SIP trunk

Two choices:

Switch your trunk to your CUBE from H.323 to SIP and use early-offer forced
(DO-EO conversion)
or
Check MTP enabled on your H.323 gateway.

The better design by far is the first option.  SIP-SIP is far preferable to
H.323-SIP, and MTP can be resource intensive and restricts you to a single
outgoing codec.

-nick

On Sun, Sep 12, 2010 at 11:09 AM, Mike Olivere <mikeeo at msn.com> wrote:

The debugs looks a little different:

Working debug:

INVITE sip:13025476274 at sip.callwithus.com:5060 SIP/2.0

Date: Sun, 12 Sep 2010 15:25:36 GMT

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

From: "3025563101" <sip:349609813 at 64.85.162.136
<mailto:sip%3A349609813 at 64.85.162.136> >;tag=4DDA620-1BCB

Allow-Events: telephone-event

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 3523226779-3182957023-2159588088-1442706662

Timestamp: 1284305136

Content-Length: 337

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

To: <sip:13025476274 at sip.callwithus.com
<mailto:sip%3A13025476274 at sip.callwithus.com> >

Contact: <sip:349609813 at 192.168.203.50:5060>

Expires: 180

Content-Disposition: session;handling=required

Content-Type: application/sdp

Call-ID: D352ECFB-BDB811DF-80BDB2F8-55FDF4E6 at 192.168.203.50

Via: SIP/2.0/UDP 192.168.203.50:5060;branch=z9hG4bK6322BC

CSeq: 101 INVITE

Max-Forwards: 70

v=0

o=CiscoSystemsSIP-GW-UserAgent 9327 7572 IN IP4 192.168.203.50

s=SIP Call

c=IN IP4 192.168.203.50

t=0 0

m=audio 18724 RTP/AVP 0 8 18 101 19

c=IN IP4 192.168.203.50

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

*Sep 12 15:25:36.258: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received: 

SIP/2.0 407 Proxy Authentication Required

From: "3025563101" <sip:349609813 at 64.85.162.136
<mailto:sip%3A349609813 at 64.85.162.136> >;tag=4DDA620-1BCB

To: <sip:13025476274 at sip.callwithus.com
<mailto:sip%3A13025476274 at sip.callwithus.com>
>;tag=5eeac0551a0e6b4db65e365785cb8817.fc09

Call-ID: D352ECFB-BDB811DF-80BDB2F8-55FDF4E6 at 192.168.203.50

Via: SIP/2.0/UDP
192.168.203.50:5060;branch=z9hG4bK6322BC;rport=48672;received=76.99.184.220

CSeq: 101 INVITE

Proxy-Authenticate: Digest realm="64.85.162.136",
nonce="4c8ce82800001675cc021ad52ef574de4d30f7b601654e07"

Server: CWU SIP GW

Content-Length: 0

Non-working debug:

Sent: 

INVITE sip:13025476274 at sip.callwithus.com:5060 SIP/2.0

Date: Sun, 12 Sep 2010 15:46:24 GMT

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

From: <sip:349609813 at 64.85.162.136 <mailto:sip%3A349609813 at 64.85.162.136>
>;tag=4F0B32C-639

Allow-Events: telephone-event

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 2163106036-3672883656-16791041-168428083

Timestamp: 1284306384

Content-Length: 0

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

To: <sip:13025476274 at sip.callwithus.com
<mailto:sip%3A13025476274 at sip.callwithus.com> >

Contact: <sip:349609813 at 192.168.203.50:5060>

Expires: 180

Call-ID: BB81C616-BDBB11DF-80CCB2F8-55FDF4E6 at 192.168.203.50

Via: SIP/2.0/UDP 192.168.203.50:5060;branch=z9hG4bK6D1934

CSeq: 101 INVITE

Max-Forwards: 70

*Sep 12 15:46:24.782: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received: 

SIP/2.0 407 Proxy Authentication Required

From: <sip:349609813 at 64.85.162.136 <mailto:sip%3A349609813 at 64.85.162.136>
>;tag=4F0B32C-639

To: <sip:13025476274 at sip.callwithus.com
<mailto:sip%3A13025476274 at sip.callwithus.com>
>;tag=5eeac0551a0e6b4db65e365785cb8817.fa75

Call-ID: BB81C616-BDBB11DF-80CCB2F8-55FDF4E6 at 192.168.203.50

Via: SIP/2.0/UDP
192.168.203.50:5060;branch=z9hG4bK6D1934;rport=65322;received=76.99.184.220

CSeq: 101 INVITE

Proxy-Authenticate: Digest realm="64.85.162.136",
nonce="4c8ced09000011853a7becbca99d846726544885b5beea9c"

Server: CWU SIP GW

Content-Length: 0

From: Ki Wi [mailto:kiwi.voice at gmail.com] 
Sent: Sunday, September 12, 2010 4:44 AM
To: Mike Olivere
Cc: <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Call manager 7 SIP trunk

Is there any differences with the number presentation during normal
operation and srst? 

Sent from my iPhone

Pls pardon my fat fingers.

On Sep 12, 2010, at 9:04 AM, "Mike Olivere" <mikeeo at msn.com> wrote:

Hey all I'm running call manger 7 with an H323 gateway that connects to the
PSTN via a sip trunk. I can't get calls to go out unless I go into SRST
mode.

Any ideas? I get a fast busy with the SIP error: SIP/2.0 488 Not acceptable
here

Once I go into SRST mode it works fine.

Thanks,

Mike

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