[cisco-voip] Call manager 7 SIP trunk

[Nick Matthews]

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<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<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<sip%3A349609813 at 64.85.162.136>
> >;tag=4DDA620-1BCB
>
> To: <sip:13025476274 at sip.callwithus.com<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 <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<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 <sip%3A349609813 at 64.85.162.136>
> >;tag=4F0B32C-639
>
> To: <sip:13025476274 at sip.callwithus.com<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
>
>  _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20100912/7108d9db/attachment.html>