<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"><base href="x-msg://17600/"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">All of this is handled in mod_sofia, look for a function with i_invite in the name.<div><br></div><div>Mike</div><div><br><div><div>On Dec 3, 2012, at 4:30 AM, sparklezou <<a href="mailto:sparklezou@163.com">sparklezou@163.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div marginheight="0" marginwidth="0" style="font-size: 12.1pt; font-family: 宋体, serif; color: rgb(0, 0, 17); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; line-height: 1.3; border-width: 0px; margin: 12px; "><div><font size="3" face="Times"><div><font size="3" face="Times">Hi Brian,</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">I'm trying to find where the session start.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Just found it start from switch_channel.c:951, the new channel. There is NO more debug info.</font></div><div> </div><div>My idea, when get the INVITE SIP message from the sofia stack. check the called number detail info in FS. add the dispaly name in the "to" field of the following SIP message.</div><div> </div><div>It should finish before the new channel.</div><div> </div><div>where could add such code?</div><div> </div><div>Thanks!</div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">2012-12-02 16:15:36.917452 [NOTICE] switch_channel.c:951 New Channel<span class="Apple-converted-space"> </span><a href="mailto:sofia/internal/1001@172.0.0.10">sofia/internal/1001@172.0.0.10</a><span class="Apple-converted-space"> </span>[b51ac8ed-69d4-409a-95ba-58b420c77bfc]<br>2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal<span class="Apple-converted-space"> </span><a href="mailto:sofia/internal/1001@172.0.0.10">sofia/internal/1001@172.0.0.10</a><span class="Apple-converted-space"> </span>[BREAK]<br>2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:415 (<a href="mailto:sofia/internal/1001@172.0.0.10">sofia/internal/1001@172.0.0.10</a>) Running State Change CS_NEW<br>2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:433 (<a href="mailto:sofia/internal/1001@172.0.0.10">sofia/internal/1001@172.0.0.10</a>) State NEW<br>2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal<span class="Apple-converted-space"> </span><a href="mailto:sofia/internal/1001@172.0.0.10">sofia/internal/1001@172.0.0.10</a><span class="Apple-converted-space"> </span>[BREAK]<br>2012-12-02 16:15:36.937457 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.<br>2012-12-02 16:15:36.937457 [DEBUG] switch_core_session.c:905 Send signal<span class="Apple-converted-space"> </span><a href="mailto:sofia/internal/1001@172.0.0.10">sofia/internal/1001@172.0.0.10</a><span class="Apple-converted-space"> </span>[BREAK]<br>2012-12-02 16:15:36.937457 [DEBUG] sofia.c:1733 detaching session b51ac8ed-69d4-409a-95ba-58b420c77bfc<br>2012-12-02 16:15:37.077884 [DEBUG] sofia.c:1825 Re-attaching to session b51ac8ed-69d4-409a-95ba-58b420c77bfc<br>2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal<span class="Apple-converted-space"> </span><a href="mailto:sofia/internal/1001@172.0.0.10">sofia/internal/1001@172.0.0.10</a><span class="Apple-converted-space"> </span>[BREAK]<br>2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal<span class="Apple-converted-space"> </span><a href="mailto:sofia/internal/1001@172.0.0.10">sofia/internal/1001@172.0.0.10</a><span class="Apple-converted-space"> </span>[BREAK]<br>2012-12-02 16:15:37.077884 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.<br>2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6306 Channel<span class="Apple-converted-space"> </span><a href="mailto:sofia/internal/1001@172.0.0.10">sofia/internal/1001@172.0.0.10</a><span class="Apple-converted-space"> </span>entering state [received][100]<br>2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6317 Remote SDP:</font></div><div> </div></font></div><div align="left"><font color="#c0c0c0" size="2" face="Verdana">2012-12-03</font></div><div align="left"><font size="2" face="Verdana"><hr id="SignNameHR" align="left" size="2" style="width: 122px; height: 2px; "></font></div><div align="left"><font color="#c0c0c0" size="2" face="Verdana">sparklezou</font></div><div><font size="2" face="Verdana"><hr></font></div><div><font size="2" face="Verdana"><strong>发件人：</strong>sparklezou</font></div><div><font size="2" face="Verdana"><strong>发送时间：</strong>2012-12-03 09:53</font></div><div><font size="2" face="Verdana"><strong>主题：</strong>Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling</font></div><div><font size="2" face="Verdana"><strong>收件人：</strong>"brian"<<a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a>></font></div><div><font size="2" face="Verdana"><strong>抄送：</strong>"freeswitch-dev"<<a href="mailto:freeswitch-dev@lists.freeswitch.org">freeswitch-dev@lists.freeswitch.org</a>></font></div><div><font size="2" face="Verdana"></font> </div><div><font size="2" face="Verdana"><div><font size="3" face="Times">Hi Brian,</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">It's the default Dialplan.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">I have tried the fresh last version also. The same result. It should NOT be configuration problem.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">You could review the wireshark log.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Thanks!</font></div><div><font size="3" face="Times"></font> </div><div align="left"><font color="#c0c0c0" size="2" face="Verdana">2012-12-03</font></div><div align="left"><font size="2" face="Verdana"><hr id="SignNameHR" align="left" size="2" style="width: 122px; height: 2px; "></font></div><div align="left"><font color="#c0c0c0" size="2" face="Verdana"><span id="_FlashSignName">sparklezou</span></font></div><div><font size="2" face="Verdana"><hr></font></div><div><font size="2" face="Verdana"><strong>发件人：</strong>sparklezou</font></div><div><font size="2" face="Verdana"><strong>发送时间：</strong>2012-12-02 17:03</font></div><div><font size="2" face="Verdana"><strong>主题：</strong>Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling</font></div><div><font size="2" face="Verdana"><strong>收件人：</strong>"freeswitch-dev"<<a href="mailto:freeswitch-dev@lists.freeswitch.org">freeswitch-dev@lists.freeswitch.org</a>></font></div><div><font size="2" face="Verdana"><strong>抄送：</strong></font></div><div><font size="2" face="Verdana"></font> </div><div><font size="2" face="Verdana"><div><font size="3" face="Times New Roman">Hi Develop team,</font></div><div><font size="3" face="Times New Roman"></font> </div><div><font size="3" face="Times New Roman">In these days, I reviewed the "DEBUG" info, and the sip info log. summarize as following:</font></div><div><font size="3" face="Times New Roman"></font> </div><div><font size="3" face="Times New Roman">Caller <---> FS, in the "from" with dispaly neam. The display is set from the Caller Phone. in the "to" without the display name. As the caller only dial the number.</font></div><div><font face="Times New Roman"></font> </div><div><font face="Times New Roman"></font> </div><div><font face="Times New Roman"><font size="3">FS -----> Called, in the "from" with display name.It's transfer from "Caller <---> FS". in the "to"</font> without the display neme. As the caller only dial the number.</font></div><div><font face="Times New Roman"></font> </div><div><font face="Times New Roman">Called -----> FS. in the "to" with display name. Here the Called Phone add the Display name into the SIP message.</font></div><div><font face="Times New Roman"></font> </div><div><font face="Times New Roman">Here conside the "Caller <----> FS" and "FS<---->Called" are two sessions. I think the best way, is check the Display Name of Called in FS system when get the INVITE message. Add the Display Name for the called in the whole process.</font></div><div><font face="Times New Roman"></font> </div><div><font face="Times New Roman">Then in the following whole process, there will be display name both for "from" & "to".</font></div><div><font face="Times New Roman"></font> </div><div><font size="3" face="Times">How about your idea?</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Thanks in advance!</font></div><div> </div><div align="left"><font color="#c0c0c0" size="2" face="Verdana">2012-12-02</font></div><div align="left"><font size="2" face="Verdana"><hr id="SignNameHR" align="left" size="2" style="width: 122px; height: 2px; "></font></div><div align="left"><font color="#c0c0c0" size="2" face="Verdana"><span id="_FlashSignName">sparklezou</span></font></div><div><font size="2" face="Verdana"><hr></font></div><div><font size="2" face="Verdana"><strong>发件人：</strong>sparklezou</font></div><div><font size="2" face="Verdana"><strong>发送时间：</strong>2012-11-30 21:13</font></div><div><font size="2" face="Verdana"><strong>主题：</strong>Re: Re: [Freeswitch-users] About the display problem during calling</font></div><div><font size="2" face="Verdana"><strong>收件人：</strong>"freeswitch-dev"<<a href="mailto:freeswitch-dev@lists.freeswitch.org">freeswitch-dev@lists.freeswitch.org</a>></font></div><div><font size="2" face="Verdana"><strong>抄送：</strong></font></div><div><font size="2" face="Verdana"></font> </div><div><font size="2" face="Verdana"><div><font size="3" face="Times">Hi Develop team,</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Just simlify this question.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">In the message 100, 180,183, 200 response to the caller side from Freeswitch side, the called part Name should be include in the sip address of "To" & SDP part. It should be "ABC"<sip: 1234@a.b.c.d>, NOT only <sip: 1234@a.b.c.d>.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Currently, at the called part, the caller info is correct with Name. So the display at called side is OK.</font></div><div><font size="3" face="Times"></font> </div><div>Could you please let me know where is the source code for such process? I also would like to review it.</div><div> </div><div>Thanks!</div><div> </div><div align="left"><font color="#c0c0c0" size="2" face="Verdana">2012-11-30</font></div><div align="left"><font size="2" face="Verdana"><hr id="SignNameHR" align="left" size="2" style="width: 122px; height: 2px; "></font></div><div align="left"><font color="#c0c0c0" size="2" face="Verdana"><span id="_FlashSignName">sparklezou</span></font></div><div><font size="2" face="Verdana"><hr></font></div><div><font size="2" face="Verdana"><strong>发件人：</strong>sparklezou</font></div><div><font size="2" face="Verdana"><strong>发送时间：</strong>2012-11-30 15:55</font></div><div><font size="2" face="Verdana"><strong>主题：</strong>Re: [Freeswitch-users] About the display problem during calling</font></div><div><font size="2" face="Verdana"><strong>收件人：</strong>"freeswitch-dev"<<a href="mailto:freeswitch-dev@lists.freeswitch.org">freeswitch-dev@lists.freeswitch.org</a>></font></div><div><font size="2" face="Verdana"><strong>抄送：</strong>"freeswitch-users"<<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>></font></div><div><font size="2" face="Verdana"></font> </div><div><font size="2" face="Verdana"><div><font size="3" face="Times">Hi All,</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Before, I saved the user name in the phone. So it will show the name when ringing.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">From the FreeSwitch side, there is NO difference.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">@ Develop Team,</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">For Internal call, or any sip call from gateway, it should be better show the Name during ring. And also keep show the name in the call, in stead of "Outbound Call".</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Usually the phone will update the display name from the sip response message.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234@a.b.c.d>.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Currently in the inital 100,180,183 message, only <sip: 1234@a.b.c.d>. In the 200 message<span class="Apple-converted-space"> </span><font size="3" face="Times">"Remote-Part-ID: "Outbound Call" <sip: 1234@a.b.c.d>; ....... "</font></font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">If it could be updated, then the called name could be displayed on the caller phone.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Thanks!</font></div><div> </div><div align="left"><font color="#c0c0c0" size="2" face="Verdana">2012-11-30</font></div><div align="left"><font size="2" face="Verdana"><hr id="SignNameHR" align="left" size="2" style="width: 122px; height: 2px; "></font></div><div align="left"><font color="#c0c0c0" size="2" face="Verdana"><span id="_FlashSignName">sparklezou</span></font></div><div><font size="2" face="Verdana"><hr></font></div><div><font size="2" face="Verdana"><strong>发件人：</strong>sparklezou</font></div><div><font size="2" face="Verdana"><strong>发送时间：</strong>2012-11-30 11:19</font></div><div><font size="2" face="Verdana"><strong>主题：</strong>[Freeswitch-users] About the display problem during calling</font></div><div><font size="2" face="Verdana"><strong>收件人：</strong>"freeswitch-users"<<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>>,"freeswitch-dev"<<a href="mailto:freeswitch-dev@lists.freeswitch.org">freeswitch-dev@lists.freeswitch.org</a>></font></div><div><font size="2" face="Verdana"><strong>抄送：</strong></font></div><div><font size="2" face="Verdana"></font> </div><div><font size="2" face="Verdana"><stationery><div><font size="3" face="Times">Hi All,</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">I facing a strange problem.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">I checked the sip info.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">The "Outbound Call" is in the 200 SDP message. the called is set to</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">""Remote-Part-ID: "Outbound Call" <sip: 1234@a.b.c.d>; ....... "</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Not the correct called name.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Is there anything wrong?</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Please help me.</font></div><div><font size="3" face="Times"></font> </div><div><font size="3" face="Times">Thanks!</font></div><div> </div><div> </div><div align="left"><font color="#c0c0c0" size="2" face="Verdana">2012-11-30</font></div><font size="2" face="Verdana"><hr id="SignNameHR" align="left" size="2" style="width: 122px; height: 2px; "></font><div><font color="#c0c0c0" size="2" face="Verdana"><span id="_FlashSignName">sparklezou</span></font></div></stationery></font></div></font></div></font></div></font></div></font></div>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>http://www.freeswitchsolutions.com<br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>http://www.cudatel.com<br><br>Official FreeSWITCH Sites<br>http://www.freeswitch.org<br>http://wiki.freeswitch.org<br>http://www.cluecon.com<br><br>FreeSWITCH-dev mailing list<br>FreeSWITCH-dev@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev<br>http://www.freeswitch.org</div></blockquote></div><br></div></body></html>