I found nat=yes was set for the first reINVITE to set hold but not the second. All the nat tests (tried all the modes for client_nat_test and nat_uac_test) seem to fail picking up that the second reINVITE (take off hold INVITE) is being nated, thus mediaproxy isn't used. From what I can see in the packet captures and what the nat tests test for, they would fail to pick up the fact that it is NAt'ed. (Packet Capture at the bottom) At the monment I'm trying to figure a way to compare the owner field with source ip address. These two IP address are different if NAT'ed. Owner is the private IP address while the source IP has the NAT public IP address. Been trying something like: if(!search("o=[0-9]* [0-9]* [0-9]* IN IP4 \Q$src_ip\E$")) { setflag(6); use_media_proxy(); } The problem I'm facing is it seems that search doesn't take in variables ($src_ip). The \Q and \E are used to quote thus avoid escaping the dots. Is there way for search to take in variables ? Or better way to compare these values? Another idea that came to me: Would checking the ruri or Route for ser's server ip address be better way, or is this abit dangerous to be routing/mediaproxying according to this? -Shaun The Following is a Packet Capture of a packet sent to SER to take the phones off hold (x.x.x.x3 = sip.xyz.com): Session Initiation Protocol Request-Line: INVITE sip:88009@x.x.x.x6:5516;user=phone SIP/2.0 Method: INVITE Resent Packet: False Message Header Via: SIP/2.0/UDP x.x.x.x70:1025;branch=z9hG4bK2d93cdb6357ddb75 Route: <sip:x.x.x.x3;ftag=10f10cec586cbf45;lr=on> From: <sip:88008@sip.xyz.com;user=phone>;tag=10f10cec586cbf45 SIP from address: sip:88008@sip.xyz.com SIP tag: 10f10cec586cbf45 To: "test-gxp2000" <sip:88009@sip.xyz.com;user=phone>;tag=c6bad1edef48137f SIP Display info: "test-gxp2000" SIP to address: sip:88009@sip.xyz.com SIP tag: c6bad1edef48137f Contact: <sip:88008@x.x.x.x70:1025> Contact Binding: <sip:88008@x.x.x.x70:1025> URI: <sip:88008@x.x.x.x70:1025> SIP contact address: sip:88008@x.x.x.x70:1025 Supported: replaces, timer Call-ID: 98176a3a3b1280d6(a)192.168.1.3 CSeq: 22284 INVITE User-Agent: Grandstream GXP2000 1.1.0.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 267 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 88008 8000 8002 IN IP4 10.0.10.111 Owner Username: 88008 Session ID: 8000 Session Version: 8002 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 10.0.10.111 Session Name (s): SIP Call Connection Information (c): IN IP4 x.x.x.x70 Connection Network Type: IN Connection Address Type: IP4 Connection Address: x.x.x.x70 Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 ---snip---- On Wednesday 25 October 2006 17:14, Greger V. Teigre wrote:

Hi, * Greger V. Teigre &lt;greger(a)teigre.com&gt; [061030 12:44]: This could cause intresting other issues. If a user has 2 phones registerd. the reinvite might get forked to both phones, wich (in a case I saw) made one phone ring, while the other picked up the call.

-A > g-) > > Shaun Hofer wrote: > >I found nat=yes was set for the first reINVITE to set hold but not the > > second. All the nat tests (tried all the modes for client_nat_test and > > nat_uac_test) seem to fail picking up that the second reINVITE (take > > off hold INVITE) is being nated, thus mediaproxy isn't used. From what > > I can see in the packet captures and what the nat tests test for, they > > would fail to pick up the fact that it is NAt'ed. (Packet Capture at > > the bottom) > > > >At the monment I'm trying to figure a way to compare the owner field > > with source ip address. These two IP address are different if NAT'ed. > > Owner is the private IP address while the source IP has the NAT public > > IP address. > > > >Been trying something like: if(!search("o=[0-9]* [0-9]* [0-9]* IN IP4 > > \Q$src_ip\E$")) { setflag(6); > > use_media_proxy(); > >} > >The problem I'm facing is it seems that search doesn't take in variables > > ($src_ip). The \Q and \E are used to quote thus avoid escaping the > > dots. > >Is there way for search to take in variables ? Or better way to compare > > these values? > > > >Another idea that came to me: Would checking the ruri or Route for ser's > > server ip address be better way, or is this abit dangerous to be > > routing/mediaproxying according to this? > >-Shaun > > > > > >The Following is a Packet Capture of a packet sent to SER to take the > > phones off hold (x.x.x.x3 = sip.xyz.com): > > > >Session Initiation Protocol > > Request-Line: INVITE sip:88009@x.x.x.x6:5516;user=phone SIP/2.0 > > Method: INVITE > > Resent Packet: False > > Message Header > > Via: SIP/2.0/UDP x.x.x.x70:1025;branch=z9hG4bK2d93cdb6357ddb75 > > Route: <sip:x.x.x.x3;ftag=10f10cec586cbf45;lr=on> > > From: <sip:88008@sip.xyz.com;user=phone>;tag=10f10cec586cbf45 > > SIP from address: sip:88008@sip.xyz.com > > SIP tag: 10f10cec586cbf45 > > To: "test-gxp2000" > > <sip:88009@sip.xyz.com;user=phone>;tag=c6bad1edef48137f SIP Display > > info: "test-gxp2000" SIP to address: sip:88009@sip.xyz.com > > SIP tag: c6bad1edef48137f > > Contact: <sip:88008@x.x.x.x70:1025> > > Contact Binding: <sip:88008@x.x.x.x70:1025> > > URI: <sip:88008@x.x.x.x70:1025> > > SIP contact address: sip:88008@x.x.x.x70:1025 > > Supported: replaces, timer > > Call-ID: 98176a3a3b1280d6(a)192.168.1.3 > > CSeq: 22284 INVITE > > User-Agent: Grandstream GXP2000 1.1.0.16 > > Max-Forwards: 70 > > Allow: > > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > > Content-Type: application/sdp > > Content-Length: 267 > > Message body > > Session Description Protocol > > Session Description Protocol Version (v): 0 > > Owner/Creator, Session Id (o): 88008 8000 8002 IN IP4 > > 10.0.10.111 Owner Username: 88008 > > Session ID: 8000 > > Session Version: 8002 > > Owner Network Type: IN > > Owner Address Type: IP4 > > Owner Address: 10.0.10.111 > > Session Name (s): SIP Call > > Connection Information (c): IN IP4 x.x.x.x70 > > Connection Network Type: IN > > Connection Address Type: IP4 > > Connection Address: x.x.x.x70 > > Time Description, active time (t): 0 0 > > Session Start Time: 0 > > Session Stop Time: 0 > >---snip---- > > > >On Wednesday 25 October 2006 17:14, Greger V. Teigre wrote: > >>- Make sure nat=yes is found in the Route set of the reINVITE. > >>- Look at the mediaproxy messages in /var/log/messages, you should get > >>one for the INVITE and then one for the OK, but not '' empty response. > >>g-) > >> > >>Shaun Hofer wrote: > >>>Hi, > >>> > >>>I've been having a problem, where audio is lost either in one or both > >>>directions when conversaion is taken off 'hold'. The parties are both > >>>behind NAT and depending UA as whether one or both loose audio. From > >>> what I can tell its to do with my loose route and nathelper, and how > >>> my ser.cfg deals with the take off hold INVITE from the phones. When > >>> the call is taken off hold the rtp streams aren't setup properly > >>> again (eg not using mediaproxy correctly). What is the best way to > >>> solve this problem? > >>> > >>>I've seen similarly posts to the mailing list about this problem with > >>> no solution posted. > >>>http://lists.iptel.org/pipermail/serusers/2006-March/027424.html > >>>http://lists.iptel.org/pipermail/serusers/2006-April/027885.html > >>>http://lists.iptel.org/pipermail/serusers/2006-May/028407.html > >>> > >>>Thanks > >>>Shaun > >>> > >>> > >>>I have a similarly config to getting started guides ser.cfg > >>> > >>># ----------------------------------------------------------------- > >>># Loose Route Section > >>># ----------------------------------------------------------------- > >>> > >>> if (loose_route()) { > >>> if (!has_totag()) { > >>> sl_send_reply("403", "Forbidden"); > >>> break; > >>> }; > >>> if (method=="INVITE") { > >>> if ((method=="INVITE" || method=="REFER") > >>>&& !has_totag()) { > >>> if (!proxy_authorize("","subscriber")) > >>> { proxy_challenge("","0"); break; > >>> } else if (!check_from()) { > >>> sl_send_reply("403", "Use > >>>From=ID"); break; > >>> }; > >>> consume_credentials(); > >>> }; > >>> if > >>> (client_nat_test("3")||search("^Route:.*;nat=yes")) { > >>> setflag(6); > >>> use_media_proxy(); > >>> }; > >>> }; > >>> route(1); > >>> break; > >>> }; > >>>_______________________________________________ > >>>Serusers mailing list > &gt;&gt;&gt;Serusers(a)lists.iptel.org > >>>http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

I'm not sure what the search for a from tag really does for you? if(loose_route() && method=="INVITE" && has_totag()) will catch all in-dialog INVITEs in general. When does the in-dialog INVITE not have a from tag? g-) Shaun Hofer wrote: