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@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@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@lists.freeswitch.org>
抄送：
All of this is handled in mod_sofia, look for a
function with i_invite in the name.
Mike
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
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
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
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
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
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
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