Yes internal should show 10.10.10.206 change ext-sip-ip and ext-rtp-ip to "auto-nat" Restart and check sofia status From the wiki ext-rtp-ip This is the IP behind which FreeSWITCH is seen from the Internet, so if FreeSWITCH is behind NAT, this is basically the public IP that should be used for RTP. On 8 Aug 2013, at 02:30, 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
_________________________________________________________________________
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.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 FreeSWITCH-users@lists.freeswitch.org 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 FreeSWITCH-users@lists.freeswitch.org 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 FreeSWITCH-users@lists.freeswitch.org 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 FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org