All of this is handled in mod_sofia, look for a function with i_invite in the name. Mike On Dec 3, 2012, at 4:30 AM, sparklezou <sparklezou@163.com> wrote:
Hi Brian,
I'm trying to find where the session start.
Just found it start from switch_channel.c:951, the new channel. There is NO more debug info.
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.
It should finish before the new channel.
where could add such code?
Thanks!
2012-12-02 16:15:36.917452 [NOTICE] switch_channel.c:951 New Channel sofia/internal/1001@172.0.0.10 [b51ac8ed-69d4-409a-95ba-58b420c77bfc] 2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001@172.0.0.10 [BREAK] 2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1001@172.0.0.10) Running State Change CS_NEW 2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/1001@172.0.0.10) State NEW 2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001@172.0.0.10 [BREAK] 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. 2012-12-02 16:15:36.937457 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001@172.0.0.10 [BREAK] 2012-12-02 16:15:36.937457 [DEBUG] sofia.c:1733 detaching session b51ac8ed-69d4-409a-95ba-58b420c77bfc 2012-12-02 16:15:37.077884 [DEBUG] sofia.c:1825 Re-attaching to session b51ac8ed-69d4-409a-95ba-58b420c77bfc 2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001@172.0.0.10 [BREAK] 2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001@172.0.0.10 [BREAK] 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. 2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6306 Channel sofia/internal/1001@172.0.0.10 entering state [received][100] 2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6317 Remote SDP:
2012-12-03 sparklezou 发件人：sparklezou 发送时间：2012-12-03 09:53 主题：Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling 收件人："brian"<brian@freeswitch.org> 抄送："freeswitch-dev"<freeswitch-dev@lists.freeswitch.org>
Hi Brian,
It's the default Dialplan.
I have tried the fresh last version also. The same result. It should NOT be configuration problem.
You could review the wireshark log.
Thanks!
2012-12-03 sparklezou 发件人：sparklezou 发送时间：2012-12-02 17:03 主题：Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling 收件人："freeswitch-dev"<freeswitch-dev@lists.freeswitch.org> 抄送：
Hi Develop team,
In these days, I reviewed the "DEBUG" info, and the sip info log. summarize as following:
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.
FS -----> Called, in the "from" with display name.It's transfer from "Caller <---> FS". in the "to" without the display neme. As the caller only dial the number.
Called -----> FS. in the "to" with display name. Here the Called Phone add the Display name into the SIP message.
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.
Then in the following whole process, there will be display name both for "from" & "to".
How about your idea?
Thanks in advance!
2012-12-02 sparklezou 发件人：sparklezou 发送时间：2012-11-30 21:13 主题：Re: Re: [Freeswitch-users] About the display problem during calling 收件人："freeswitch-dev"<freeswitch-dev@lists.freeswitch.org> 抄送：
Hi Develop team,
Just simlify this question.
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>.
Currently, at the called part, the caller info is correct with Name. So the display at called side is OK.
Could you please let me know where is the source code for such process? I also would like to review it.
Thanks!
2012-11-30 sparklezou 发件人：sparklezou 发送时间：2012-11-30 15:55 主题：Re: [Freeswitch-users] About the display problem during calling 收件人："freeswitch-dev"<freeswitch-dev@lists.freeswitch.org> 抄送："freeswitch-users"<freeswitch-users@lists.freeswitch.org>
Hi All,
Before, I saved the user name in the phone. So it will show the name when ringing.
From the FreeSwitch side, there is NO difference.
@ Develop Team,
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".
Usually the phone will update the display name from the sip response message.
So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234@a.b.c.d>.
Currently in the inital 100,180,183 message, only <sip: 1234@a.b.c.d>. In the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234@a.b.c.d>; ....... "
If it could be updated, then the called name could be displayed on the caller phone.
Thanks!
2012-11-30 sparklezou 发件人：sparklezou 发送时间：2012-11-30 11:19 主题：[Freeswitch-users] About the display problem during calling 收件人："freeswitch-users"<freeswitch-users@lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev@lists.freeswitch.org> 抄送：
Hi All,
I facing a strange problem.
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.
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.
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.
I checked the sip info.
The "Outbound Call" is in the 200 SDP message. the called is set to
""Remote-Part-ID: "Outbound Call" <sip: 1234@a.b.c.d>; ....... "
Not the correct called name.
Is there anything wrong?
Please help me.
Thanks!
2012-11-30 sparklezou _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org