[Freeswitch-dev] [Freeswitch-users] About the display problem during calling
Bing LI
enst.bupt at gmail.com
Tue Dec 4 15:12:02 MSK 2012
http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_name
http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number
http://wiki.freeswitch.org/wiki/Clarification:gateways#Caller_ID
On Tue, Dec 4, 2012 at 1:43 AM, sparklezou <sparklezou at 163.com> wrote:
> **
> Hi Mike & Brian,
>
> I find the way to correct the final 200 SDP response message.
>
> The "Outbound Call" is in the 200 SDP message. the called is set to
>
> ""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "
>
> In the switch_ivr_originate.c file, in the following code.
>
> new_profile->callee_id_name = switch_core_strdup(new_profile->pool,
> "Outbound Call");
>
> Before this sentence, there should be one function to find the display
> name from the number. Then replace the default "Outbound Call".
>
> Is there any function to find the name from the user xml files?
>
> In this way, just correct the final dispaly issue.
>
> And during the call, I still NOT find the corret location to add the
> dispaly name in to the sip "to" field.
>
> But I believe it should NOT add in the sofia sip stack side. It should add
> at the uplay application invote function side.
>
> I think, it should be in switch_channel.c switch_ivr_originate.c
> switch_event.c files.
>
> Please check and consider it.
>
> Thanks!
>
>
> 2012-12-04
> ------------------------------
> sparklezou
> ------------------------------
> *发件人：*sparklezou
> *发送时间：*2012-12-03 23:40
> *主题：*Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display
> problem during calling
> *收件人：*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> *抄送：*
>
> Hi Mike,
>
> Thank!
>
> In which functin should be the right place to add the Name for the "to"
> field?
>
> How about your idea?
>
> 2012-12-03
> ------------------------------
> sparklezou
> ------------------------------
> *发件人：*Michael Jerris
> *发送时间：*2012-12-03 23:20
> *主题：*Re: [Freeswitch-dev] [Freeswitch-users] About the display problem
> during calling
> *收件人：*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> *抄送：*
>
> 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 at 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 at 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 at 172.0.0.10 [BREAK]
> 2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:415 (
> sofia/internal/1001 at 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 at 172.0.0.10) State NEW
> 2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/1001 at 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 at 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 at 172.0.0.10 [BREAK]
> 2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/1001 at 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 at 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 at freeswitch.org>
> *抄送：*"freeswitch-dev"<freeswitch-dev at 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 at 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 at 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 at a.b.c.d>, NOT only
> <sip: 1234 at 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 at lists.freeswitch.org>
> *抄送：*"freeswitch-users"<freeswitch-users at 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 at a.b.c.d>.
>
> Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In
> the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at 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 at lists.freeswitch.org
> >,"freeswitch-dev"<freeswitch-dev at 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 at a.b.c.d>; ....... "
>
> Not the correct called name.
>
> Is there anything wrong?
>
> Please help me.
>
> Thanks!
>
>
> 2012-11-30
> ------------------------------
> sparklezou
> **
> _________________________________________________________________________
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>
>
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>
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