Your ip for the external profile will show whatever the ip is set to in the external profile configs. It doesn't matter what is set in the internal profile.

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster@davri.com
Indianapolis, Indiana

On Aug 7, 2013 9:36 PM, "Peter" <eidevm5@gmail.com> wrote:

Hi Anthony

Really appreciate the taking your time to look at this. It's starting to drive me nuts.

My dialplan for calls to CME is:

<extension name="incoming-cme">
        <condition field="destination_number" expression="^(3\d\d\d)$">
            <action application="bridge" data="sofia/internal/${destination_number}@10.10.10.203"/>
        </condition>
</extension>

My internal sip profile has:

    <param name="rtp-ip" value="10.10.10.206"/>
    <param name="sip-ip" value="10.10.10.206"/>
    <param name="ext-rtp-ip" value="10.1.1.206"/>
    <param name="ext-sip-ip" value="10.1.1.206"/>

The output from:

sofia status

is:

internal    profile                 sip:mod_sofia@10.1.1.206:5060    RUNNING (0)
external    profile                 sip:mod_sofia@10.1.1.206:5060    RUNNING (0)

Should internal show 10.10.10.206??

The output from:

sofia status profile internal

shows:

Name                 internal
Domain Name          N/A
Auto-NAT             false
DBName               sofia_reg_internal
Pres Hosts           10.10.10.206,10.1.1.206
Dialplan             XML
Context              public
Challenge Realm     auto_from
RTP-IP               10.10.10.206
Ext-RTP-IP           10.1.1.206
SIP-IP               10.10.10.206
Ext-SIP-IP           10.1.1.206
URL                  sip:mod_sofia@10.1.1.206:5060
BIND-URL             sip:mod_sofia@10.1.1.206:5060;maddr=10.10.10.206
HOLD-MUSIC           N/A
OUTBOUND-PROXY       N/A
CODECS IN            iLBC@30i,PCMU,PCMA,GSM
CODECS OUT           iLBC@30i,PCMU,PCMA,GSM
TEL-EVENT            101
DTMF-MODE            rfc2833
CNG                  13
SESSION-TO           0
MAX-DIALOG           0
NOMEDIA              false
LATE-NEG             false
PROXY-MEDIA          false
ZRTP-PASSTHRU        false
AGGRESSIVENAT        false
STUN-ENABLED         true
STUN-AUTO-DISABLE    false

The SIP trace from the Freeswitch SBC is at:

http://pastebin.freeswitch.org/21279

I've been playing around with all sorts of different combinations of SIP/RTP IP settings, but still no closer.

Hope you have some insight.

Thanks

Peter

On Wed, Aug 7, 2013 at 6:31 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote:

Hi Peter,

Your debug shows the invite with via/from/contact/rpid all coming from 10.1.1.206, your external side.

Check your bridge statement, is it using the correct sip profile?
Check your sip profile SBC internal params rtp-ip & sip-ip, make sure they are set correctly to 10.10.10.206
Paste up your logs from the sbc including sip trace.

Anthony

On 7 Aug 2013, at 08:12, Peter <eidevm5@gmail.com> wrote:

Hi Anthony.

Yes, the SIP profiles are the same for calls going to Kamailio and to CME/CUBE.

Note that CME only has one interface, so binding the source interface doesn't really make much sense.

Note that I've simplified my set up a little and the phone that was registered to CUCM is now registered to CME. However, the result is still the same, ie: one way audio to the Cisco phone.

You can see the SIP debug from CME at:

http://pastebin.freeswitch.org/21274

The call is coming from 1001@10.1.1.204 to 3000@10.10.10.203

where

10.1.1.204 - Freeswitch where SIP clients register to
10.1.1.206 - External side of Freeswitch SBC
10.10.10.206 - Internal side of Freeswitch SBC
10.10.10.203 - CME

Peter

On Tue, Aug 6, 2013 at 5:13 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote:

Hi Peter,

Because the calls are fine when using Kamailio I'm assuming your sip profiles are fine and you FS SBC config is fine. Are you using the same profiles?

Yes you are correct. Have you added the commands? Add them as a first step.
Send on a 'debug ccsip messages'

Anthony

On 6 Aug 2013, at 05:35, Peter <eidevm5@gmail.com> wrote:

Thanks for replying Anthony.

Keep in mind that I have very little experience with Cisco products, so I may be missing something fundamental.

As far as I can see

voice-class sip bind media source-interface ....

is just used to bind the SIP or media stream to the appropriate interface on the CUBE.

My issue is that the CUBE is trying to initiate the return RTP stream to the external interface (instead of the internal interface) on the Freeswitch SBC.

Is my understanding of the sip bind media command correct?

Thanks

Peter

On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote:

On cube make sure you specify the source address on your dial-peers
voice-class sip bind media|control
to the correct side. I have seen one way audio when not set.

On 5 Aug 2013, at 06:29, Peter <eidevm5@gmail.com> wrote:

>
>
> I currently have successful two way calls (signalling and media) in the following setup
>
>
> External Linphone --> Freeswitch --> Freeswitch SBC -> Router -> Kamailio --> Internal Linphone
>
> However, when I try to call a Cisco handset that is registered to CUCM9 via CME in the following config:
>
> External Linphone --> Freeswitch --> Freeswitch SBC -> Router -> CME -> CUCM9 --> Cisco handset
>
> The call signalling appears to be working fine and I can successfully initiate a call from each end, but the only RTP stream that is working is from the external Linphone client to the Cisco handset.
>
> Note that CME is being used as a CUBE device, so all SIP and RTP goes via it.
>
> Looking at the RTP debugs on CME I can see the problem is that the "Media Dest Addr" is getting set to the external side of the FS SBC rather than the internal IP address.
>
>
> I tried setting adding:
>
> <action application="set" data="disable_rtp_auto_adjust="true" />
>
> to the dialplan on the SBC, but it made no difference.
>
>
> Any suggestions as to what to check next?
>
> Thanks
>
> Peter
>

> _________________________________________________________________________

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_________________________________________________________________________
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http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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_________________________________________________________________________
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http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
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