Hi, NAT scenario, correct? https://confluence.freeswitch.org/display/FREESWITCH/NAT+Traversal try this: vars.xml <X-PRE-PROCESS cmd="set" data="bind_server_ip=my.domain.com"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/> sip_profiles: sipinterface_1.xml <param name="apply-nat-acl" value="nat.auto"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> Doug On Sat, Aug 16, 2014 at 7:26 AM, afshin afzali <a.afzali2003@gmail.com> wrote:
Hi Miguel,
I do not understand how the ACL (if you mean apply-nat-acl) can effects on using public IP in SDP when I specified both ext-sip-ip and ext-rtp-ip in sip profile! I changed the apply-nat-acl to a "always allow" and "rfc1918" but does not make difference.
Afshin
On Thu, Aug 14, 2014 at 3:18 PM, Miguel Oyarzo <miguelaustro@gmail.com> wrote:
Wrong ACL?
On Wed, Aug 13, 2014 at 10:06 PM, afshin afzali <a.afzali2003@gmail.com> wrote:
Hi FreeSWITCHers:
I configured a sip profile as bellow:
================================================================================================= Name sipinterface_1 Domain Name N/A Auto-NAT false DBName sofia_reg_sipinterface_1 Pres Hosts Dialplan XML Context context_1 Challenge Realm auto_to RTP-IP 192.168.10.59 Ext-RTP-IP X.Y.Z.W SIP-IP 192.168.10.59 Ext-SIP-IP X.Y.Z.W URL sip:mod_sofia@X.Y.Z.W:5060 BIND-URL sip:mod_sofia@X.Y.Z.W :5060;maddr=192.168.10.59;transport=udp,tcp HOLD-MUSIC N/A OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM CODECS OUT G722,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 true CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 1
for every INVITE that comes in via this profile, FS replys with this SDP:
v=0 o=FreeSWITCH 1407905843 1407905844 IN IP4 192.168.10.59 s=FreeSWITCH c=IN IP4 192.168.10.59 t=0 0 m=audio 25142 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
Although the Contact header of 200 OK is correct : Contact: <sip:4000@X.Y.Z.W:5060;transport=udp>
Appreciate all comments, -- Afshin
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Miguel Oyarzo DevOps & VoIP Engineer Linux User: # 483188 - counter.li.org http://au.linkedin.com/in/mikeaustralia Melbourne, Australia
_________________________________________________________________________ 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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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