Nice.<br><br>
<div class="gmail_quote">On Mon, Nov 14, 2011 at 7:29 PM, Schneur Rosenberg <span dir="ltr"><<a href="mailto:rosenberg11219@gmail.com">rosenberg11219@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">I added record_route() before calling route(20) (incoming did route)<br>and now the BYE does hit the phone, I think I might of fixed the<br>
problem, I will test it tomorrow I need to get moving, thanks Duane<br>and Khamis.<br>
<div class="HOEnZb">
<div class="h5"><br><br><br>On Tue, Nov 15, 2011 at 3:25 AM, Duane Larson <<a href="mailto:duane.larson@gmail.com">duane.larson@gmail.com</a>> wrote:<br>> The OpenSIPS setup I usually work with doesn't proxy that much with Asterisk<br>
> doing all the work so take what I say sparingly.<br>><br>> 404 Not Here means that OpenSIPS is saying no user account exists. So in<br>> your Asterisk BYE the user is<br>><br>> U asterisk2IP:5060 -> opensipsIP:5060<br>
> BYE sip:solhome7@93.172.0.116:5060;nat=yes SIP/2.0.<br>><br>> Does OpenSIPS know of a user named <a href="mailto:solhome7@93.172.0.116">solhome7@93.172.0.116</a>? Since that is all<br>> that is in the SIP message that is all I have to go by. I also see that<br>
> there are devices called solhome7, solhome3 and solhome5<br>><br>><br>> On Mon, Nov 14, 2011 at 7:00 PM, Schneur Rosenberg<br>> <<a href="mailto:rosenberg11219@gmail.com">rosenberg11219@gmail.com</a>> wrote:<br>
>><br>>> I see asterisk is sending the BYE to the phone, but opensips sends a<br>>> not here, bellow is the sip strace<br>>><br>>> U <a href="http://93.172.0.116:1047/" target="_blank">93.172.0.116:1047</a> -> opensipsip:5060INVITE<br>
>> sip:1917398XXXX@opensipsip SIP/2.0.Via: SIP/2.0/UDP<br>>> 192.168.1.8:5060;branch=z9hG4bK-b5ec4068.From:<br>>> <sip:solhome3@opensipsip>;tag=9c059eac8018b3c8o0.To:<br>>> <sip:<a href="tel:19173985000" value="+19173985000">19173985000</a>@opensipsip>.Remote-Party-ID:<br>
>> <sip:solhome3@opensipsip>;screen=yes;party=calling.Call-ID:<br>>> 82537c-a80f0538@192.168.1.8.CSeq: 101 INVITE.Max-Forwards: 70.Contact:<br>>> <<a href="http://sip:solhome3@192.168.1.8:5060" target="_blank">sip:solhome3@192.168.1.8:5060</a>>.Expires: 240.User-Agent:<br>
>> Linksys/SPA2102-5.2.12.Content-Length: 444.Allow: ACK, BYE, CANCEL,<br>>> INFO, INVITE, NOTIFY, OPTIONS, REFER.Supported: x-sipura,<br>>> replaces.Content-Type: application/sdp.<br>>><br>>><br>
>> U opensipsip:5060 -> <a href="http://93.172.0.116:1047/" target="_blank">93.172.0.116:1047</a><br>>> SIP/2.0 407 Proxy Authentication Required.<br>>> Via: SIP/2.0/UDP<br>>> 192.168.1.8:5060;branch=z9hG4bK-b5ec4068;rport=1047;received=93.172.0.116.<br>
>> From: <sip:solhome3@opensipsip>;tag=9c059eac8018b3c8o0.<br>>> To:<br>>> <sip:1917398XXXX@sopensipsip>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.ef95.<br>>> Call-ID: <a href="mailto:82537c-a80f0538@192.168.1.8">82537c-a80f0538@192.168.1.8</a>.<br>
>> CSeq: 101 INVITE.<br>>> Proxy-Authenticate: Digest realm="opensipsip",<br>>> nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee".<br>>> Server: OpenSIPS (1.6.4-2-notls (x86_64/linux)).<br>
>> Content-Length: 0.<br>>><br>>><br>>> U <a href="http://93.172.0.116:1047/" target="_blank">93.172.0.116:1047</a> -> opensipsIP:5060<br>>> INVITE sip:1917398XXXX@opensipsIP SIP/2.0.<br>>> Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK-ec946528.<br>
>> From: <sip:solhome3@opensipsIP>;tag=9c059eac8018b3c8o0.<br>>> To: <sip:1917398XXXX@opensipsIP>.<br>>> Remote-Party-ID: <sip:solhome3@opensipsIP>;screen=yes;party=calling.<br>>> Call-ID: <a href="mailto:82537c-a80f0538@192.168.1.8">82537c-a80f0538@192.168.1.8</a>.<br>
>> CSeq: 102 INVITE.<br>>> Max-Forwards: 70.<br>>> Proxy-Authorization: Digest<br>>><br>>> username="solhome3",realm="opensipsIP",nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee",uri="sip:1917398XXXX@opensipsIP",algorithm=MD5,response="db2640507b2e9824235649f51629ceee".<br>
>> Contact: <<a href="http://sip:solhome3@192.168.1.8:5060" target="_blank">sip:solhome3@192.168.1.8:5060</a>>.<br>>> Expires: 240.<br>>> User-Agent: Linksys/SPA2102-5.2.12.<br>>> Content-Length: 444.<br>
>> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.<br>>> Supported: x-sipura, replaces.<br>>> Content-Type: application/sdp.<br>>><br>>><br>>> U opensipsIP:5060 -> <a href="http://93.172.0.116:1047/" target="_blank">93.172.0.116:1047</a><br>
>> SIP/2.0 100 Giving a try.<br>>> Via: SIP/2.0/UDP<br>>> 192.168.1.8:5060;branch=z9hG4bK-ec946528;rport=1047;received=93.172.0.116.<br>>> From: <sip:solhome3@opensipsIP>;tag=9c059eac8018b3c8o0.<br>
>> To: <sip:1917398xxxx@opensipsIP>.<br>>> Call-ID: <a href="mailto:82537c-a80f0538@192.168.1.8">82537c-a80f0538@192.168.1.8</a>.<br>>> CSeq: 102 INVITE.<br>>> Server: OpenSIPS (1.6.4-2-notls (x86_64/linux)).<br>
>> Content-Length: 0.<br>>><br>>> U opensipsIP:5060 -> asteriskIP:5060<br>>> INVITE sip:1917398XXXX@opensipsIP SIP/2.0.<br>>> Record-Route: <sip:opensipsIP;lr=on;did=935.e9420777>.<br>
>> Via: SIP/2.0/UDP opensipsIP;branch=z9hG4bK9049.19290602.0.<br>>> Via: SIP/2.0/UDP<br>>> 192.168.1.8:5060;rport=1047;received=93.172.0.116;branch=z9hG4bK-ec946528.<br>>> From: <sip:solhome3@opensipsIP>;tag=9c059eac8018b3c8o0.<br>
>> To: <sip:<a href="tel:19173985000" value="+19173985000">19173985000</a>@opensipsIP>.<br>>> Remote-Party-ID: <sip:solhome3@opensipsIP>;screen=yes;party=calling.<br>>> Call-ID: <a href="mailto:82537c-a80f0538@192.168.1.8">82537c-a80f0538@192.168.1.8</a>.<br>
>> CSeq: 102 INVITE.<br>>> Max-Forwards: 69.<br>>> Contact: <<a href="http://sip:solhome3@93.172.0.116:1047" target="_blank">sip:solhome3@93.172.0.116:1047</a>>.<br>>> Expires: 240.<br>>> User-Agent: Linksys/SPA2102-5.2.12.<br>
>> Content-Length: 444.<br>>> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.<br>>> Supported: x-sipura, replaces.<br>>> Content-Type: application/sdp.<br>>><br>>> U asteriskIP:5060 -> opensipsIP:5060<br>
>> SIP/2.0 100 Trying.<br>>> Via:<br>>> SIP/2.0/UDPopensipsIP;branch=z9hG4bK9049.19290602.0;received=opensipsIP;rport=5060.<br>>> Via: SIP/2.0/UDP<br>>> 192.168.1.8:5060;rport=1047;received=93.172.0.116;branch=z9hG4bK-ec946528.<br>
>> Record-Route: <sip:opensipsIP;lr=on;did=935.e9420777>.<br>>> From: <sip:solhome3@opensipsIP>;tag=9c059eac8018b3c8o0.<br>>> To: <sip:1917398xxxx@opensipsIP>.<br>>> Call-ID: <a href="mailto:82537c-a80f0538@192.168.1.8">82537c-a80f0538@192.168.1.8</a>.<br>
>> CSeq: 102 INVITE.<br>>> Server: Asterisk PBX 1.8.7.1.<br>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,<br>>> INFO, PUBLISH.<br>>> Supported: replaces, timer.<br>>> Contact: <<a href="http://sip:19173985000@64.69.47.109:5060" target="_blank">sip:19173985000@64.69.47.109:5060</a>>.<br>
>> Content-Length: 0.<br>>><br>>> U DIDProviderIP:5060 -> opensipsIP:5060<br>>> INVITE sip:917398xxxx@opensipsIP SIP/2.0.<br>>> Via: SIP/2.0/UDP DIDProviderIP:5060;branch=z9hG4bK0b523109;rport.<br>
>> Max-Forwards: 70.<br>>> From: "ROSENBERG S" <sip:9173985xxxx@DIDproviderIP>;tag=as09899a91.<br>>> To: <sip:917398xxxx@opensipsIP>.<br>>> Contact: <sip:917398xxxx@DIDProviderip>.<br>
>> Call-ID: 66d0ba94185dba0430f45f195772e31a@DIDProvidorIP.<br>>> CSeq: 102 INVITE.<br>>> User-Agent: Linksys/SPA2100-3.3.6(0911s).<br>>> Remote-Party-ID: "ROSENBERG S"<br>>> <sip:917398xxxx@DIDProviderIP>;privacy=off;screen=no.<br>
>> Date: Mon, 14 Nov 2011 23:35:28 GMT.<br>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<br>>> Supported: replaces, timer.<br>>> Content-Type: application/sdp.<br>>> Content-Length: 340.<br>
>><br>>> U opensipsIP:5060 -> asterisk2ip:5060<br>>> INVITE sip:did917398xxxx@opensipsIP SIP/2.0.<br>>> Via: SIP/2.0/UDP opensipsIP;branch=z9hG4bKf77f.f5d40393.0.<br>>> Via:<br>>> SIP/2.0/UDPDIDProviderIP:5060;received=DIDProviderIP;branch=z9hG4bK0b523109;rport=5060.<br>
>> Max-Forwards: 69.<br>>> From: "ROSENBERG S" <sip:917398xxxx@DIDProviderIP>;tag=as09899a91.<br>>> To: <sip:<a href="tel:9173985000" value="+19173985000">9173985000</a>@opensipsIP>.<br>
>> Contact: <sip:917398xxxx@DIDProviderIP>.<br>>> Call-ID: 66d0ba94185dba0430f45f195772e31a@DIDProviderIP.<br>>> CSeq: 102 INVITE.<br>>> User-Agent: Linksys/SPA2100-3.3.6(0911s).<br>>> Remote-Party-ID: "ROSENBERG S"<br>
>> <sip:917398xxxx@DIDProviderIP>;privacy=off;screen=no.<br>>> Date: Mon, 14 Nov 2011 23:35:28 GMT.<br>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<br>>> Supported: replaces, timer.<br>
>> Content-Type: application/sdp.<br>>> Content-Length: 340.<br>>> P-hint: Unathenticated from outside ie did.<br>>><br>>> U asterisk2IP:5060 -> opensipsIP:5060<br>>> SIP/2.0 100 Trying<br>
>> Truncated because of length<br>>><br>>> U asterisk2IP:5060 -> opensipsIP:5060<br>>> INVITE sip:solhome7@opensipsIP SIP/2.0.<br>>> Via: SIP/2.0/UDP asterisk2IP:5060;branch=z9hG4bK39459435;rport.<br>
>> Max-Forwards: 70.<br>>> From: "ROSENBERG S" <sip:917398xxxx@asterisk2IP>;tag=as5ec8d074.<br>>> To: <sip:solhome5@opensipsIP>.<br>>> Contact: <sip:917398xxxx@asterisk2IP:5060>.<br>
>> Call-ID: 73f977bc448143a26b68be5d38de196e@asterisk2IP:5060.<br>>> CSeq: 102 INVITE.<br>>> User-Agent: Asterisk PBX 1.8.7.1.<br>>> Date: Mon, 14 Nov 2011 23:35:19 GMT.<br>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,<br>
>> INFO, PUBLISH.<br>>> Supported: replaces, timer.<br>>> P-Asserted-Identity: "ROSENBERG S" <sip:917398xxxx@asterisk2IP>.<br>>> Content-Type: application/sdp.<br>>> Content-Length: 282.<br>
>><br>>> RINGING<br>>><br>>> U <a href="http://93.172.0.116:5060/" target="_blank">93.172.0.116:5060</a> -> opensipsIP:5060<br>>> SIP/2.0 200 OK.<br>>> Via: SIP/2.0/UDP opensipsIP;branch=z9hG4bKa96f.8afc2a77.0.<br>
>> Via: SIP/2.0/UDP<br>>> asterisk2IP:5060;received=asterisk2IP;branch=z9hG4bK727d493c;rport=5060.<br>>> From: "ROSENBERG S" <sip:917398xxxx@asterisk2IP>;tag=as605029e0.<br>>> To: <sip:solhome7@sopensipsIP>;tag=6A174081-8FE8464C.<br>
>> CSeq: 102 INVITE.<br>>> Call-ID: 09fdaad65a393c1751acd56e150d50a9@asterisk2IP:5060.<br>>> Contact: <<a href="mailto:sip%3Asolhome7@192.168.1.2">sip:solhome7@192.168.1.2</a>>.<br>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,<br>
>> NOTIFY, PRACK, UPDATE, REFER.<br>>> User-Agent: PolycomSoundPointIP-SPIP_601-UA/<a href="http://3.1.7.92/" target="_blank">3.1.7.0134</a>.<br>>> Accept-Language: en.<br>>> Content-Type: application/sdp.<br>
>> Content-Length: 197.<br>>><br>>> U opensipsIP:5060 -> asterisk2IP:5060<br>>> SIP/2.0 200 OK.<br>>><br>>> U asterisk2IP:5060 -> opensipsIP:5060<br>>> ACK sip:solhome7@93.172.0.116:5060;nat=yes SIP/2.0.<br>
>><br>>> U <a href="http://93.172.0.116:1047/" target="_blank">93.172.0.116:1047</a> -> opensipsIP:5060<br>>> BYE sip:1917398xxxx@asteriskIP:5060;nat=yes SIP/2.0.<br>>> Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK-5f187bca.<br>
>> From: <sip:solhome3@opensipsIP>;tag=9c059eac8018b3c8o0.<br>>> To: <sip:1917398xxxx@opensipsIP>;tag=as5852d19d.<br>>> Call-ID: <a href="mailto:82537c-a80f0538@192.168.1.8">82537c-a80f0538@192.168.1.8</a>.<br>
>> CSeq: 103 BYE.<br>>> Max-Forwards: 70.<br>>> Route: <sip:opensipsIP;lr=on;did=935.e9420777>.<br>>> Proxy-Authorization: Digest<br>>><br>>> username="solhome3",realm="opensipsIP",nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee",uri="sip:1917398xxxx@asteriskIP:5060",algorithm=MD5,response="3bc688c27090bca344187bef1a5e4eee".<br>
>> User-Agent: Linksys/SPA2102-5.2.12.<br>>> Content-Length: 0.<br>>> .<br>>><br>>><br>>> U opensipsIP:5060 -> asteriskIP:5060<br>>> BYE sip:1917398xxxx@asteriskIP:5060 SIP/2.0.<br>
>> Via: SIP/2.0/UDP opensipsIP;branch=z9hG4bKa049.76464162.0.<br>>> Via: SIP/2.0/UDP<br>>> 192.168.1.8:5060;rport=1047;received=93.172.0.116;branch=z9hG4bK-5f187bca.<br>>> From: <sip:solhome3@opensikpsIP>;tag=9c059eac8018b3c8o0.<br>
>> To: <sip:1917398xxxx@opensipsIP>;tag=as5852d19d.<br>>> Call-ID: <a href="mailto:82537c-a80f0538@192.168.1.8">82537c-a80f0538@192.168.1.8</a>.<br>>> CSeq: 103 BYE.<br>>> Max-Forwards: 69.<br>
>> Proxy-Authorization: Digest<br>>><br>>> username="solhome3",realm="opensipsIP",nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee",uri="sip:1917398xxxx@asteriskIP:5060",algorithm=MD5,response="3bc688c27090bca344187bef1a5e49d8".<br>
>> User-Agent: Linksys/SPA2102-5.2.12.<br>>> Content-Length: 0.<br>>><br>>> U asteriskIP:5060 -> opensipsIP:5060<br>>> SIP/2.0 200 OK.<br>>><br>>> U opensipsIP:5060 -> <a href="http://93.172.0.116:1047/" target="_blank">93.172.0.116:1047</a><br>
>> SIP/2.0 200 OK.<br>>><br>>> U asterisk2IP:5060 -> opensipsIP:5060<br>>> BYE sip:solhome7@93.172.0.116:5060;nat=yes SIP/2.0.<br>>><br>>> .<br>>> U opensipsIP:5060 -> asteriskIP:5060<br>
>> SIP/2.0 404 Not here.<br>>><br>>><br>>><br>>><br>>> On Tue, Nov 15, 2011 at 2:19 AM, <<a href="mailto:duane.larson@gmail.com">duane.larson@gmail.com</a>> wrote:<br>>> > Could you provide a sip trace of a call from INVITE to BYE? Also in your<br>
>> > opensips config look and see where you have "404 Not here" configured.<br>>> ><br>>> ><br>>> ><br>>> > On , Schneur Rosenberg <<a href="mailto:rosenberg11219@gmail.com">rosenberg11219@gmail.com</a>> wrote:<br>
>> >> In my case this is not relevant, because I'm calling the other phone<br>>> >><br>>> >> through a DID and the did needs to go to asterisk to decide what to do<br>>> >><br>
>> >> with it, it can send it to a IVR which can later send it to Opensips<br>>> >><br>>> >> etc. in any case I need to know why asterisk is not sending the BYE to<br>>> >><br>
>> >> the phone, and why opensips sends a not here when the BYE comes from a<br>>> >><br>>> >> phone not on the system, in that case asterisk sends the BYE to<br>>> >><br>>> >> opensips which sends a not here instead of sending it to the phone<br>
>> >><br>>> >><br>>> >><br>>> >> On Tue, Nov 15, 2011 at 2:06 AM, <a href="mailto:duane.larson@gmail.com">duane.larson@gmail.com</a>> wrote:<br>>> >><br>>> >> > If you want VM then you send to Asterks when the call times out (AKA<br>
>> >> > the<br>>> >><br>>> >> > callee doesn't pick up). We weren't talking about VM here. If you<br>>> >> > want<br>>> >> > MOH<br>>> >><br>
>> >> > then that is a totally different beast. You would always have to send<br>>> >> > the<br>>> >><br>>> >> > calls to Asterisk and Asterisk would stay in the flow of the call.<br>
>> >> > From<br>>> >> > what<br>>> >><br>>> >> > I read above it sounded like the following<br>>> >><br>>> >> ><br>>> >><br>>> >> > When I call from one phone on the system to another phone on the<br>
>> >><br>>> >> > same opensips, the phone sends a BYE to opensips which sends it to<br>>> >> > the<br>>> >><br>>> >> > asterisk but the BYE never gets sent to the called phone.<br>
>> >><br>>> >> ><br>>> >><br>>> >> > Sounds like Asterisk is not sending the BYE back to OpenSIPS because<br>>> >> > its<br>>> >><br>>> >> > stated " opensips which sends it to the asterisk but the BYE never<br>
>> >> > gets<br>>> >> > sent<br>>> >><br>>> >> > to the called phone."<br>>> >><br>>> >> ><br>>> >><br>>> >> ><br>
>> >><br>>> >> ><br>>> >><br>>> >> ><br>>> >><br>>> >> > On , Nick Khamis <a href="mailto:symack@gmail.com">symack@gmail.com</a>> wrote:<br>
>> >><br>>> >> >> On Mon, Nov 14, 2011 at 6:50 PM, <a href="mailto:duane.larson@gmail.com">duane.larson@gmail.com</a>> wrote:<br>>> >><br>>> >> >><br>>> >><br>
>> >> >> > If two phones are registered with OpenSIPS and they call each<br>>> >> >> > other<br>>> >> >> > why<br>>> >><br>>> >> >><br>
>> >><br>>> >> >> > would you send the SIP messages to Asterisk?<br>>> >><br>>> >> >><br>>> >><br>>> >> >><br>>> >><br>
>> >> >><br>>> >><br>>> >> >> Because "<a href="http://www.opensips.org/Resources/DocsTutAsterisk" target="_blank">http://www.opensips.org/Resources/DocsTutAsterisk</a>" said so!<br>
>> >> >> ;)<br>>> >><br>>> >> >><br>>> >><br>>> >> >><br>>> >><br>>> >> >><br>>> >><br>>> >> >> > You need to set up route logic so that if two local users call<br>
>> >> >> > each<br>>> >><br>>> >> >> > other then<br>>> >><br>>> >> >><br>>> >><br>>> >> >> > the asterisk boxes are kept out of the equation.<br>
>> >><br>>> >> >><br>>> >><br>>> >> >><br>>> >><br>>> >> >><br>>> >><br>>> >> >> Amazing idea! But what would happen to MOH, and VM?<br>
>> >><br>>> >> >><br>>> >><br>>> >> >><br>>> >><br>>> >> >><br>>> >><br>>> >> >> Nick.<br>>> >><br>
>> >> >><br>>> >><br>>> >> >><br>>> >><br>>> >> >><br>>> >><br>>> >> >> _______________________________________________<br>
>> >><br>>> >> >><br>>> >><br>>> >> >> Users mailing list<br>>> >><br>>> >> >><br>>> >><br>>> >> >> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
>> >><br>>> >> >><br>>> >><br>>> >> >> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
>> >><br>>> >> >><br>>> >><br>>> >> >><br>>> >><br>>> >> > _______________________________________________<br>>> >><br>>> >> > Users mailing list<br>
>> >><br>>> >> > <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>>> >><br>>> >> > <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
>> >><br>>> >> ><br>>> >><br>>> >> ><br>>> >><br>>> >><br>>> >><br>>> >> _______________________________________________<br>
>> >><br>>> >> Users mailing list<br>>> >><br>>> >> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>>> >><br>>> >> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
>> >><br>>> >><br>>> > _______________________________________________<br>>> > Users mailing list<br>>> > <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
>> > <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>>> ><br>>> ><br>>><br>>> _______________________________________________<br>
>> Users mailing list<br>>> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>>> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
><br>><br>><br>> --<br>> --<br>> *--*--*--*--*--*<br>> Duane<br>> *--*--*--*--*--*<br>> --<br>><br>> _______________________________________________<br>> Users mailing list<br>> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>><br>><br><br>_______________________________________________<br>
Users mailing list<br><a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br><a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
</div></div></blockquote></div><br><br clear="all"><br>-- <br>--<br>*--*--*--*--*--*<br>Duane<br>*--*--*--*--*--*<br>--<br>