_________________________________________________________________________ThanksIf so, which profiles do they need to be set in? If they are set in both the internal and external SIP profiles, does that matter?Should sip-ip ALWAYS be set to your internal IP address and ext-sip-ip to your external IP address?Have to say I'm getting a little confused as to when and where to use sip-ip and ext-sip-ipHi Brian.I did have the IP set in the external profile, but after removing them it didn't seem to make any difference, ie: the SIP ports were listening on the correct IP address.
PeterOn Thu, Aug 8, 2013 at 11:41 AM, Brian Foster <bdfoster@davri.com> wrote: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.
Thank you,
Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster@davri.com
Indianapolis, IndianaSent from a mobile device.
On Aug 7, 2013 9:36 PM, "Peter" <eidevm5@gmail.com> wrote:shows:The output from:My internal sip profile has:My dialplan for calls to CME is:Hi AnthonyReally appreciate the taking your time to look at this. It's starting to drive me nuts.
<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>
<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"/>
sofia statusis:
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
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
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
PeterOn 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 10.10.10.206Paste up your logs from the sbc including sip trace.AnthonyOn 7 Aug 2013, at 08:12, Peter <eidevm5@gmail.com> wrote:_________________________________________________________________________Peter10.10.10.203 - CME10.10.10.206 - Internal side of Freeswitch SBC10.1.1.206 - External side of Freeswitch SBC10.1.1.204 - Freeswitch where SIP clients register towhereThe call is coming from 1001@10.1.1.204 to 3000@10.10.10.203You can see the SIP debug from CME at:Note that CME only has one interface, so binding the source interface doesn't really make much sense.Hi Anthony.Yes, the SIP profiles are the same for calls going to Kamailio and to CME/CUBE.
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.
http://pastebin.freeswitch.org/21274
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'
AnthonyIs my understanding of the sip bind media command correct?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 just used to bind the SIP or media stream to the appropriate interface on the CUBE.voice-class sip bind media source-interface ....As far as I can seeThanks for replying Anthony.Keep in mind that I have very little experience with Cisco products, so I may be missing something fundamental.
Thanks
PeterOn 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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_________________________________________________________________________
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
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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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http://www.freeswitch.org
_________________________________________________________________________
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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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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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