[Freeswitch-users] Wrong IP on ACK?

Tue Nov 4 09:32:14 PST 2008

You need to set localnet and externip or externhost on Asterisk so it  
doesn't lie about its IP.

/b

On Nov 4, 2008, at 11:10 AM, David Aldworth wrote:

> For some reason when trunking with Asterisk PBX's (yes, I know) FS
> wants to send the ACK to the internal ip found in the Contact field of
> the 200 OK. We have the force rport setting on but it's still not
> responding to that IP. Register's work. Most of the sip signalling
> works, just when the customer specifies the Contact filed with an
> internal ip. Below is a packet capture and our external.xml conf file.
>
> U 2008/11/04 09:17:08.259672 64.74.188.23:5060 -> 68.188.189.202:5060
> INVITE sip:989607XXXX at 68.188.189.202:5060 SIP/2.0.
> Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bKc6y6pr80HyeaN.
> Max-Forwards: 68.
> From: "user" <sip:303452XXXX at 64.74.188.23>;tag=Dt6v81cDZXa3B.
> To: <sip:989607XXXX at 68.188.189.202:5060>.
> Call-ID: de7c471c-252e-122c-3cba-5f1bac93b621.
> CSeq: 106789378 INVITE.
> Contact: <sip:mod_sofia at 64.74.188.23:5060>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
> Supported: 100rel, timer, precondition, path, replaces.
> Allow-Events: talk, presence, dialog, call-info, sla, include-session-
> description, presence.winfo, message-summary.
> Min-SE: 120.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 370.
>
> U 2008/11/04 09:21:32.861720 68.188.189.202:5060 -> 64.74.188.23:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP
> 64.74.188.23
> ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060.
> From: "user" <sip:303452XXXX at 64.74.188.23>;tag=4UF788r8ct8aD.
> To: <sip:989607XXXX at 68.188.189.202:5060>.
> Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621.
> CSeq: 106789510 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces.
> Contact: <sip:989607XXXX at 192.168.0.5>.
> Content-Length: 0.
> .
>
> U 2008/11/04 09:21:32.861845 68.188.189.202:5060 -> 64.74.188.23:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 64.74.188.23
> ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060.
> From: "user" <sip:303452XXXX at 64.74.188.23>;tag=4UF788r8ct8aD.
> To: <sip:989607XXXX at 68.188.189.202:5060>;tag=as1da4b7aa.
> Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621.
> CSeq: 106789510 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces.
> Contact: <sip:989607XXXX at 192.168.0.5>.
> Content-Type: application/sdp.
> Content-Length: 285.
> .
> v=0.
> o=root 10970 10970 IN IP4 192.168.0.5.
> s=session.
> c=IN IP4 192.168.0.5.
> t=0 0.
> m=audio 15876 RTP/AVP 18 0 101.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> a=sendrecv.
>
>
> U 2008/11/04 09:21:32.862573 64.74.188.23:5060 -> 192.168.0.5:5060
> ACK sip:989607XXXX at 192.168.0.5 SIP/2.0.
> Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bK0r7c2a501SQvD.
> Max-Forwards: 70.
> From: "user" <sip:303452XXXX at 64.74.188.23>;tag=4UF788r8ct8aD.
> To: <sip:989607XXXX at 68.188.189.202:5060>;tag=as1da4b7aa.
> Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621.
> CSeq: 106789510 ACK.
> Contact: <sip:mod_sofia at 64.74.188.23:5060>.
> Content-Length: 0.
> .
>
>
>
>
>
> external.xml
>
>
>   <settings>
>     <param name="debug" value="0"/>
>     <param name="sip-trace" value="no"/>
>     <param name="rfc2833-pt" value="101"/>
>     <param name="sip-port" value="5060"/>
>     <param name="dialplan" value="XML"/>
>     <param name="context" value="public"/>
>     <param name="dtmf-duration" value="100"/>
>     <param name="codec-prefs" value="$${outbound_codec_prefs}"/>
>     <param name="hold-music" value="$${hold_music}"/>
>     <param name="use-rtp-timer" value="true"/>
>     <param name="rtp-timer-name" value="soft"/>
>     <param name="multiple-registrations" value="true"/>
>     <param name="manage-presence" value="true"/>
>     <param name="aggressive-nat-detection" value="true"/>
>     <param name="NDLB-force-rport" value="true"/>
>     <param name="inbound-codec-negotiation" value="generous"/>
>     <param name="nonce-ttl" value="60"/>
>     <param name="auth-calls" value="true"/>
>     <param name="rtp-timeout-sec" value="1800"/>
>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>     <param name="sip-ip" value="$${local_ip_v4}"/>
>     <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>     <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>     <param name="rtp-timeout-sec" value="300"/>
>     <param name="rtp-hold-timeout-sec" value="1800"/>
>   </settings>
>
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