<html><body><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px"><div dir="ltr" id="yui_3_16_0_1_1416894949617_49143"><span id="yui_3_16_0_1_1416894949617_49145">Hi, <br></span></div><div id="yui_3_16_0_1_1416894949617_49146" dir="ltr"><span style="" class="" id="yui_3_16_0_1_1416894949617_49145">I copied the all flags same as Nikita script.</span>Nothing has changed but in the rtp.log there are some lines :</div><div id="yui_3_16_0_1_1416894949617_49523" dir="ltr">Nov 24 07:36:55 spce rtpengine[6426]: Unknown flag encountered: 'symmetric'<br style="" class="">Nov 24 07:36:55 spce rtpengine[6426]: Unknown 'rtcp-mux' flag encountered: 'demuxSRTP'<br style="" class=""><br></div><div id="yui_3_16_0_1_1416894949617_49509" dir="ltr">Nov 24 07:37:00 spce rtpengine[6426]: [f5008c55-8329-f08e-e024-81d8260b1708 port 30865] SRTCP output wanted, but no crypto suite was negotiated</div><div id="yui_3_16_0_1_1416894949617_51662" dir="ltr">.</div><div id="yui_3_16_0_1_1416894949617_51661" dir="ltr">.</div><div id="yui_3_16_0_1_1416894949617_50947" dir="ltr">.</div><div id="yui_3_16_0_1_1416894949617_51660" dir="ltr">.</div><div id="yui_3_16_0_1_1416894949617_50942" dir="ltr">Nov 24 07:37:32 spce rtpengine[6426]: [f5008c55-8329-f08e-e024-81d8260b1708] Scheduling deletion of call branch 'R7t3SMDI7STFq4A53a9w' in 30 seconds<br style="" class=""></div><div id="yui_3_16_0_1_1416894949617_51659" dir="ltr"><br></div><div id="yui_3_16_0_1_1416894949617_51658" dir="ltr">this flags not supported by rtpengine now? (3.6.1) <br></div><div id="yui_3_16_0_1_1416894949617_51657" dir="ltr">how suite crypto will be negotiated?</div><div id="yui_3_16_0_1_1416894949617_51654" dir="ltr"><br></div><div id="yui_3_16_0_1_1416894949617_51656" dir="ltr"><br></div><div id="yui_3_16_0_1_1416894949617_51655" dir="ltr">Regards,</div><div id="yui_3_16_0_1_1416894949617_51806" dir="ltr">H. Yavari<br></div><div id="yui_3_16_0_1_1416894949617_50943" dir="ltr"><br></div><br> <div id="yui_3_16_0_1_1416894949617_49078" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 13px;"> <div id="yui_3_16_0_1_1416894949617_49077" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px;"> <div id="yui_3_16_0_1_1416894949617_49135" dir="ltr"> <hr id="yui_3_16_0_1_1416894949617_49142" size="1"> <font id="yui_3_16_0_1_1416894949617_49134" face="Arial" size="2"> <b id="yui_3_16_0_1_1416894949617_49141"><span id="yui_3_16_0_1_1416894949617_49140" style="font-weight:bold;">From:</span></b> Andreas Granig <agranig@sipwise.com><br> <br> </font> </div> <div id="yui_3_16_0_1_1416894949617_49076" class="y_msg_container"><br><div id="yui_3_16_0_1_1416894949617_49147" dir="ltr">You don't need stun/turn with rtpengine, because it puts itself into the<br clear="none">SDP as ICE candidate (if you set the according preferences), so it can<br clear="none">act as turn server. stun is really only needed if you want to enforce<br clear="none">peer-to-peer communication without rtpengine in between.<br clear="none"><br clear="none">Andreas<br clear="none"><br clear="none">On 11/25/2014 08:48 AM, H Yavari wrote:<br clear="none">> Hi,<br clear="none">> Thanks for helps. I know that you did all for free. I have a question,<br clear="none">> Are you using ICE server or STUN? I did all of my test in the local<br clear="none">> domain and with private IP's.<br clear="none">> SPCE team, have you any idea for this issue?<br clear="none">> <br clear="none">> <br clear="none">> Regards,<br clear="none">> H.Yavari<br clear="none">> <br clear="none">> <br clear="none">> ------------------------------------------------------------------------<br clear="none">> *From:* Nikita Stashkov <<a shape="rect" ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>><br clear="none">> **<br clear="none">> Sorry, I have done all I can do for free. You can test new versions with<br clear="none">> my site. I think they are working.<br clear="none">> If you need more help, it can be only commercial support.<br clear="none">> <br clear="none">> Regards,<br clear="none">> Nikita Stashkov<br clear="none">> <br clear="none">>> 24 нояб. 2014 г., в 17:53, H Yavari <<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br clear="none">>> <mailto:<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>> написал(а):<br clear="none">>><br clear="none">>><br clear="none">>><br clear="none">>> Hi,<br clear="none">>> Very thanks for sharing the script. I'm very confused. I checked the<br clear="none">>> script line by line and differences are some lines that I think added<br clear="none">>> in the 3.6.1 and they are not related to the media. I added<br clear="none">>> "rtcp-mux-demux" flags like your script too. but nothing has changed<br clear="none">>> and issues not solved.<br clear="none">>> So I lost my way. maybe the all problems is from client side. Your<br clear="none">>> script working with current version of jssip and sipml5? and latest<br clear="none">>> Chrome and Firefox versions?<br clear="none">>><br clear="none">>> Regards,<br clear="none">>> H.Yavari<br clear="none">>><br clear="none">>> ------------------------------------------------------------------------<br clear="none">>> *From:* Nikita Stashkov <<a shape="rect" ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a> <mailto:<a shape="rect" ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>>><br clear="none">>><br clear="none">>><br clear="none">>> Ok, if it will help you.<br clear="none">>> Attached is my script (without push), and domain settings.<br clear="none">>> Should not be understood literally all. I have many changes in config.<br clear="none">>><br clear="none">>><br clear="none">>><br clear="none">>><br clear="none">>><br clear="none">>><br clear="none">>>> 24 нояб. 2014 г., в 13:07, H Yavari <<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br clear="none">>>> <mailto:<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>> написал(а):<br clear="none">>>><br clear="none">>>> Hi,<br clear="none">>>> Yes, with sipml5 calls have been terminated. I changed all ws to ws<br clear="none">>>> || wss. I did this too :<br clear="none">>>> if(isbflagset(FLB_SAVP_CALLER_SRTP))<br clear="none">>>> {<br clear="none">>>> xlog("L_INFO", "Try SRTP for<br clear="none">>>> caller - [% logreq -%]\n");<br clear="none">>>> $var(rtpp_flags) =<br clear="none">>>> $var(rtpp_flags) + "SRTP rtcp-mux-demux ";<br clear="none">>>> }<br clear="none">>>> but did not any changes.<br clear="none">>>><br clear="none">>>> Can you share with me? and you media settings?<br clear="none">>>><br clear="none">>>> I want only use this solution in our website for support calls to our<br clear="none">>>> IP-PBX.<br clear="none">>>><br clear="none">>>> Thanks.<br clear="none">>>><br clear="none">>>> Regards,<br clear="none">>>> H.YAvari<br clear="none">>>> ------------------------------------------------------------------------<br clear="none">>>> *From:* Nikita Stashkov <<a shape="rect" ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a> <mailto:<a shape="rect" ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>>><br clear="none">>>><br clear="none">>>><br clear="none">>>> And the first one is sipml5?<br clear="none">>>> In my config both are working.<br clear="none">>>> Check again your script. There is not one place, where automatic<br clear="none">>>> detection is done.<br clear="none">>>> I don’t exactly remember. It was about 4-5 month ago. But I think,<br clear="none">>>> difference is between ws and wss.<br clear="none">>>> Sorry, I can not publish my script. There are many other things,<br clear="none">>>> including push notifications.<br clear="none">>>> I think Sipwise will be not happy, if I publish it.<br clear="none">>>><br clear="none">>>> Regards,<br clear="none">>>> Nikita Stashkov<br clear="none">>>>> 24 нояб. 2014 г., в 11:45, H Yavari <<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br clear="none">>>>> <mailto:<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>> написал(а):<br clear="none">>>>><br clear="none">>>>><br clear="none">>>>><br clear="none">>>>> Hi,<br clear="none">>>>><br clear="none">>>>> I noticed a new thing that when I using jssip, calls not terminated.<br clear="none">>>>> in the logs and I didn't see any rtcp-mux. so this two webRTC client<br clear="none">>>>> is different in using SDP params?<br clear="none">>>>><br clear="none">>>>><br clear="none">>>>> Regards,<br clear="none">>>>> H.Yavari<br clear="none">>>>> ------------------------------------------------------------------------<br clear="none">>>>> *From:* Nikita Stashkov <<a shape="rect" ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a> <mailto:<a shape="rect" ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>>><br clear="none">>>>><br clear="none">>>>><br clear="none">>>>> There may be different logic. I am doing it if caller is WebRTC, and<br clear="none">>>>> callee is SIP. <br clear="none">>>>> Simply add this flag, when calling Rtpengine, like all other flags.<br clear="none">>>>> You can do nothing if both are WebRTC.<br clear="none">>>>><br clear="none">>>>> Regards,<br clear="none">>>>> Nikita Stashkov<br clear="none">>>>><br clear="none">>>>><br clear="none">>>>><br clear="none">>>>> 24 нояб. 2014 г., в 11:13, H Yavari <<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br clear="none">>>>> <mailto:<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>> написал(а):<br clear="none">>>>><br clear="none">>>>><br clear="none">>>>><br clear="none">>>>>> Hi,<br clear="none">>>>>> I checked it. the client (webRTC browser-sipml5) send rtp-mux. and<br clear="none">>>>>> is in the rtp.log too. so how can I disable this? or how can I add<br clear="none">>>>>> rtcp-mux-demux ? I should do this for all calls? or only for webRTC<br clear="none">>>>>> client?<br clear="none">>>>>><br clear="none">>>>>><br clear="none">>>>>> Thanks a lot.<br clear="none">>>>>><br clear="none">>>>>> Regards,<br clear="none">>>>>> H.Yavari<br clear="none">>>>>><br clear="none">>>>>> ------------------------------------------------------------------------<br clear="none">>>>>> *From:* Nikita Stashkov <<a shape="rect" ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a> <mailto:<a shape="rect" ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>>><br clear="none">>>>>> **<br clear="none">>>>>><br clear="none">>>>>> You need to look logs from WebRTC client and Pcap from SIP.<br clear="none">>>>>> Of cource, if SIP client recives SDP with rtp-mux, he will not<br clear="none">>>>>> understand it. And after 30 sec call will be terminated. But you<br clear="none">>>>>> must see logs. My system is based on SPCE 3.2, and manually<br clear="none">>>>>> compiled rtpengine. And I don't know was changed in current<br clear="none">>>>>> version. Also, you can look rtp.log. Sometimes it helps.<br clear="none">>>>>><br clear="none">>>>>> Regards,<br clear="none">>>>>> Nikita Stashkov<br clear="none">>>>>><br clear="none">>>>>><br clear="none">>>>>> 23. nov. 2014, в 13.40, H Yavari <<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br clear="none">>>>>> <mailto:<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>> написал(а):<br clear="none">>>>>><br clear="none">>>>>><br clear="none">>>>>><br clear="none">>>>>>> Hi,<br clear="none">>>>>>><br clear="none">>>>>>> Dear I did this before that I changed "ws" with "wss" but now<br clear="none">>>>>>> after your reply I did "ws" || "wss". but not any changes.<br clear="none">>>>>>> As I told before, now my main problem is calls hangup after 30<br clear="none">>>>>>> sec. In your opinion the rtcp-mux-demux flags adding will solve this?<br clear="none">>>>>>> another point is that before 30 sec, If any call parties (caller:<br clear="none">>>>>>> browser and callee: soft phone) hangs up, the call not terminate<br clear="none">>>>>>> until 30 sec timeout. I think that the dialog of a call not<br clear="none">>>>>>> recognized.<br clear="none">>>>>>><br clear="none">>>>>>> So situation is complicated :)<br clear="none">>>>>>> SPCE specialist plz help!<br clear="none">>>>>>><br clear="none">>>>>>><br clear="none">>>>>>> Regards,<br clear="none">>>>>>> H. Yavari<br clear="none">>>>>>><br clear="none">>>>>>> ------------------------------------------------------------------------<br clear="none">>>>>>> *From:* Nikita Stashkov <<a shape="rect" ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a> <mailto:<a shape="rect" ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>>><br clear="none">>>>>>> **<br clear="none">>>>>>> Sorry, I can not share my script.<br clear="none">>>>>>> What can you do.<br clear="none">>>>>>> Look the<br clear="none">>>>>>> script /etc/ngcp-config/templates/etc/kamailio/proxy/proxy.cfg.tt2<br clear="none">>>>>>> Of course, before modifying copy it to proxy.cfg.customtt.tt2<br clear="none">>>>>>> I think, webrtc endpoint automatic detection is not working for you.<br clear="none">>>>>>> It must look like this:<br clear="none">>>>>>><br clear="none">>>>>>> if($(ru{uri.param,transport}) == "ws" ||<br clear="none">>>>>>> $(ru{uri.param,transport}) == "wss»)<br clear="none">>>>>>><br clear="none">>>>>>> Then check flags you are sending to rtpengine.<br clear="none">>>>>>> To call SIP clients you must use flag rtcp-mux-demux<br clear="none">>>>>>><br clear="none">>>>>>> Regards,<br clear="none">>>>>>> Nikita Stashkov<br clear="none">>>>>>><br clear="none">>>>>>><br clear="none">>>>>>>> 22 нояб. 2014 г., в 20:28, H Yavari <<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br clear="none">>>>>>>> <mailto:<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>> написал(а):<br clear="none">>>>>>>><br clear="none">>>>>>>><br clear="none">>>>>>>><br clear="none">>>>>>>> Hi,<br clear="none">>>>>>>> I checked you site. it seems that is a good webRTC solution.<br clear="none">>>>>>>> Can you share with us your experience to solve our problem? or<br clear="none">>>>>>>> any script modifications?<br clear="none">>>>>>>><br clear="none">>>>>>>><br clear="none">>>>>>>> Regards,<br clear="none">>>>>>>> H.Yavari<br clear="none">>>>>>>><br clear="none">>>>>>>> ------------------------------------------------------------------------<br clear="none">>>>>>>> *From:* Nikita Stashkov <<a shape="rect" ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a><br clear="none">>>>>>>> <mailto:<a shape="rect" ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>>><br clear="none">>>>>>>><br clear="none">>>>>>>> You can try with my site - www.sipmobile.org<br clear="none">>>>>>>> <<a shape="rect" href="http://www.sipmobile.org/" target="_blank">http://www.sipmobile.org/</a>>.<br clear="none">>>>>>>> Create accounts: <a shape="rect" href="https://www.sipmobile.org/register/" target="_blank">https://www.sipmobile.org/register/</a><br clear="none">>>>>>>> And try to call with webRTC client and SIP.<br clear="none">>>>>>>> I have modified some Kamailio SPCE scripts.<br clear="none">>>>>>>><br clear="none">>>>>>>> Regards,<br clear="none">>>>>>>> Nikita Stashkov<br clear="none">>>>>>>><br clear="none">>>>>>>><br clear="none">>>>>>>>> 22 нояб. 2014 г., в 16:04, Thomas Odorfer <<a shape="rect" ymailto="mailto:odotom@gmail.com" href="mailto:odotom@gmail.com">odotom@gmail.com</a><br clear="none">>>>>>>>> <mailto:<a shape="rect" ymailto="mailto:odotom@gmail.com" href="mailto:odotom@gmail.com">odotom@gmail.com</a>>> написал(а):<br clear="none">>>>>>>>><br clear="none">>>>>>>><br clear="none">>>>>>>><br clear="none">>>>>>>> Hi,<br clear="none">>>>>>>> not sure if I understood correctly which scenario works and which<br clear="none">>>>>>>> not. <br clear="none">>>>>>>> So browser to soft phone is now working, but what is the meaning<br clear="none">>>>>>>> of browser to client? Which client?<br clear="none">>>>>>>><br clear="none">>>>>>>> I tested myself and I have to confess that I had to do some<br clear="none">>>>>>>> changes in the account configs for soft phones where I am not<br clear="none">>>>>>>> happy about.<br clear="none">>>>>>>> It only worked between browser-webrtc and soft phone when the<br clear="none">>>>>>>> corresponding account for the soft phone - nat & media flow<br clear="none">>>>>>>> control had been changed to "force avp"“ and "force rtp“ ie. no<br clear="none">>>>>>>> encryption.<br clear="none">>>>>>>> (I have to investigate that one - could be related to an upgrade<br clear="none">>>>>>>> I had performed last week - usually srtp should also work with<br clear="none">>>>>>>> softphones, within the ftp.log there was „SRTP output wanted but<br clear="none">>>>>>>> no crypto suite was negotiated“).<br clear="none">>>>>>>> However, after my changes the following tests had been successful:<br clear="none">>>>>>>> browser webrtc to softphone (eg. jitsi, counterpath x-lite -<br clear="none">>>>>>>> should be software compatible with eyebeam)<br clear="none">>>>>>>> browser webrtc to another browser webrtc (jssip-0.50)<br clear="none">>>>>>>> browser webrtc to pstn via sip trunking (standard sip trunk,<br clear="none">>>>>>>> peer settings for media force „rtp“, „force rtp“, „always with<br clear="none">>>>>>>> plain SDP“)<br clear="none">>>>>>>><br clear="none">>>>>>>> That is based on the latest SPCE version 3.6.1.<br clear="none">>>>>>>> What does not seem to be achievable at the moment that you can<br clear="none">>>>>>>> have an account that supports „standard“ and webrtc<br clear="none">>>>>>>> simultaneously ( at least I haven’t succeeded with such a setup,<br clear="none">>>>>>>> maybe some sipwise/kamailio/rtpengine expert knows the trick).<br clear="none">>>>>>>> And I do not have a solution yet how to share one phone number<br clear="none">>>>>>>> between two accounts with different profiles.<br clear="none">>>>>>>> The only solution I have at the moment is that I put a webrtc<br clear="none">>>>>>>> gateway (similar to webrtc2sip from doubango) in front of SPCE<br clear="none">>>>>>>> for webrtc clients.<br clear="none">>>>>>>><br clear="none">>>>>>>> For your particular problem, maybe you have to check whether your<br clear="none">>>>>>>> domain settings allow „bypass rtp proxy“ behind the same NAT -<br clear="none">>>>>>>> assuming you are testing wthin your LAN - this should be set to<br clear="none">>>>>>>> never.<br clear="none">>>>>>>><br clear="none">>>>>>>> Good luck<br clear="none">>>>>>>> Thomas<br clear="none">>>>>>>><br clear="none">>>>>>>><br clear="none">>>>>>>> Am 22.11.2014 um 12:49 schrieb H Yavari <<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br clear="none">>>>>>>> <mailto:<a shape="rect" ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>>:<br clear="none">>>>>>>><br clear="none">>>>>>>>> Hi,<br clear="none">>>>>>>>><br clear="none">>>>>>>>> I did this configs:<br clear="none">>>>>>>>> use_rtpproxy: „Always with rtpptoxy as only ICE candidate“<br clear="none">>>>>>>>> rtcp_feedback: „Force AVP“<br clear="none">>>>>>>>> srtp_transcoding: „Force RTP“<br clear="none">>>>>>>>><br clear="none">>>>>>>>> now calls between browser to soft phone is ok, but browser to<br clear="none">>>>>>>>> client and browser to browser receive this error "Failed to get<br clear="none">>>>>>>>> local SDP"<br clear="none">>>>>>>>> and calls not be established. Have you any idea about this<br clear="none">>>>>>>>> situation?<br clear="none">>>>>>>>> Thanks for helps.<br clear="none">>>>>>>>><br clear="none">>>>>>>>> Regards,<br clear="none">>>>>>>>> H.Yavari<br clear="none">>>>>>>>> ------------------------------------------------------------------------<br clear="none">>>>>>>>><br clear="none">>>>>>>><br clear="none">>>>>>>> _______________________________________________<br clear="none">>>>>>>> Spce-user mailing list<br clear="none">>>>>>>> <a shape="rect" ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a> <mailto:<a shape="rect" ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a>><br clear="none">>>>>>>> <a shape="rect" href="https://lists.sipwise.com/listinfo/spce-user" target="_blank">https://lists.sipwise.com/listinfo/spce-user</a><div class="qtdSeparateBR"><br><br></div><div class="yqt4343808641" id="yqtfd05006"><br clear="none">>>>>>>><br clear="none">>>>>>>><br clear="none">>>>>>>><br clear="none">>>>>>><br clear="none">>>>>>><br clear="none">>>>>>><br clear="none">>>>>><br clear="none">>>>>><br clear="none">>>>><br clear="none">>>>><br clear="none">>>><br clear="none">>>><br clear="none">>>><br clear="none">>><br clear="none">>><br clear="none">>><br clear="none">> <br clear="none">> <br clear="none">> <br clear="none">> <br clear="none">> <br clear="none">> <br clear="none">> <br clear="none">> _______________________________________________<br clear="none">> Spce-user mailing list<br clear="none">> <a shape="rect" ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a><br clear="none">> <a shape="rect" href="https://lists.sipwise.com/listinfo/spce-user" target="_blank">https://lists.sipwise.com/listinfo/spce-user</a><br clear="none">> <br clear="none">_______________________________________________<br clear="none">Spce-user mailing list<br clear="none"><a shape="rect" ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a><br clear="none"><a shape="rect" href="https://lists.sipwise.com/listinfo/spce-user" target="_blank">https://lists.sipwise.com/listinfo/spce-user</a><br clear="none"></div></div><br><br></div> </div> </div> </div></body></html>