you only need to "set" it on the inbound leg and you must answer and bridge it somewhere.<br><br><br><div class="gmail_quote">On Mon, Oct 19, 2009 at 9:47 AM, Helmut Kuper <span dir="ltr"><<a href="mailto:helmut.kuper@ewetel.de">helmut.kuper@ewetel.de</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class="im">-----BEGIN PGP SIGNED MESSAGE-----<br>
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</div>Hello Anthony,<br>
<br>
I updated and restarted my test FS to "FreeSWITCH Version 1.0.trunk<br>
(15174M)". Callee's experience didn't change:<br>
<div class="im"><br>
> 1. Phone rings: caller's displayname<br>
> 2. Callee picks up: switching from dislayname to unknown<br>
> 3. Switching from unknown to displayname<br>
<br>
</div>I used the two chvars you mentioned, set them via "set" and as well via<br>
"export" but no change (neither on caller's nor in callee's display nor<br>
in SIP INFO messages)<br>
<br>
<br>
My dialplan portion for this is:<br>
<extension name="Local_Extension"><br>
<condition field="${ET_is_local}" expression="^true$"><br>
<action application="set"<br>
data="dialed_extension=${destination_number}"/><br>
<action application="set" data="sip_callee_id_number=1111"/><br>
<action application="set" data="sip_callee_id_name=hubu"/><br>
<action application="export" data="sip_callee_id_number=1111"/><br>
<action application="export" data="sip_callee_id_name=hubu"/><br>
<action application="info"/><br>
[...]<br>
<br>
<br>
Here is the output of the info app after setting those chvars:<br>
<br>
[INFO] mod_dptools.c:961 CHANNEL_DATA:<br>
Channel-State: [CS_EXECUTE]<br>
Channel-State-Number: [4]<br>
Channel-Name: [sofia/internal/<a href="http://1001@85.16.246.12:5061" target="_blank">1001@85.16.246.12:5061</a>]<br>
Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]<br>
Call-Direction: [inbound]<br>
Presence-Call-Direction: [inbound]<br>
Answer-State: [ringing]<br>
Caller-Username: [1001]<br>
Caller-Dialplan: [XML]<br>
Caller-Caller-ID-Name: [1001 an PBX1]<br>
Caller-Caller-ID-Number: [1001]<br>
Caller-Network-Addr: [85.16.245.206]<br>
Caller-Destination-Number: [1000]<br>
Caller-Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]<br>
Caller-Source: [mod_sofia]<br>
Caller-Context: [default]<br>
Caller-RDNIS: [1000]<br>
Caller-Channel-Name: [sofia/internal/<a href="http://1001@85.16.246.12:5061" target="_blank">1001@85.16.246.12:5061</a>]<br>
Caller-Profile-Index: [2]<br>
Caller-Profile-Created-Time: [1255963206242587]<br>
Caller-Channel-Created-Time: [1255963206214959]<br>
Caller-Channel-Answered-Time: [0]<br>
Caller-Channel-Progress-Time: [0]<br>
Caller-Channel-Progress-Media-Time: [0]<br>
Caller-Channel-Hangup-Time: [0]<br>
Caller-Channel-Transfer-Time: [0]<br>
Caller-Screen-Bit: [true]<br>
Caller-Privacy-Hide-Name: [false]<br>
Caller-Privacy-Hide-Number: [false]<br>
variable_sip_received_ip: [85.16.245.206]<br>
variable_sip_received_port: [1024]<br>
variable_sip_via_protocol: [udp]<br>
variable_sip_authorized: [true]<br>
variable_sip_from_user: [1001]<br>
variable_sip_from_port: [5061]<br>
variable_sip_from_uri: [<a href="http://1001@85.16.246.12:5061" target="_blank">1001@85.16.246.12:5061</a>]<br>
variable_sip_from_host: [85.16.246.12]<br>
variable_sip_from_user_stripped: [1001]<br>
variable_sip_from_tag: [snfuiue6ga]<br>
variable_sofia_profile_name: [internal]<br>
variable_sip_req_params: [user=phone]<br>
variable_sip_req_user: [1000]<br>
variable_sip_req_port: [5061]<br>
variable_sip_req_uri: [<a href="http://1000@85.16.246.12:5061" target="_blank">1000@85.16.246.12:5061</a>]<br>
variable_sip_req_host: [85.16.246.12]<br>
variable_sip_to_params: [user=phone]<br>
variable_sip_to_user: [1000]<br>
variable_sip_to_port: [5061]<br>
variable_sip_to_uri: [<a href="http://1000@85.16.246.12:5061" target="_blank">1000@85.16.246.12:5061</a>]<br>
variable_sip_to_host: [85.16.246.12]<br>
variable_sip_contact_params: [line=eg3wp69a]<br>
variable_sip_contact_user: [1001]<br>
variable_sip_contact_port: [1024]<br>
variable_sip_contact_uri: [<a href="http://1001@85.16.245.206:1024" target="_blank">1001@85.16.245.206:1024</a>]<br>
variable_sip_contact_host: [85.16.245.206]<br>
variable_channel_name: [sofia/internal/<a href="http://1001@85.16.246.12:5061" target="_blank">1001@85.16.246.12:5061</a>]<br>
variable_sip_call_id: [3c2d2d8f9a49-edzr2i2iezjp]<br>
variable_sip_user_agent: [snom820/8.2.16]<br>
variable_sip_via_host: [85.16.245.206]<br>
variable_sip_via_port: [1024]<br>
variable_sip_via_rport: [1024]<br>
variable_presence_id: [<a href="mailto:1001@85.16.246.12">1001@85.16.246.12</a>]<br>
variable_sip_h_X-Serialnumber: [0004134002CB]<br>
variable_sip_h_P-Key-Flags: [resolution="31x13", keys="4"]<br>
variable_switch_r_sdp: [v=0<br>
o=root 1331667919 1331667919 IN IP4 85.16.245.206<br>
s=call<br>
c=IN IP4 85.16.245.206<br>
t=0 0<br>
m=audio 62882 RTP/SAVP 0 8 9 99 3 18 4 101<br>
a=rtpmap:0 pcmu/8000<br>
a=rtpmap:8 pcma/8000<br>
a=rtpmap:9 g722/8000<br>
a=rtpmap:99 g726-32/8000<br>
a=rtpmap:3 gsm/8000<br>
a=rtpmap:18 g729/8000<br>
a=rtpmap:4 g723/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=crypto:1 AES_CM_128_HMAC_SHA1_32<br>
inline:m6fas/KsLF57r9RnU7X0WEWeJw9Y6+a66YUIf9Dc<br>
a=ptime:20<br>
m=audio 62882 RTP/AVP 0 8 9 99 3 18 4 101<br>
a=rtpmap:0 pcmu/8000<br>
a=rtpmap:8 pcma/8000<br>
a=rtpmap:9 g722/8000<br>
a=rtpmap:99 g726-32/8000<br>
a=rtpmap:3 gsm/8000<br>
a=rtpmap:18 g729/8000<br>
a=rtpmap:4 g723/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
]<br>
variable_ep_codec_string: [G722@8000h@20i,PCMA@8000h@20i]<br>
variable_endpoint_disposition: [DELAYED NEGOTIATION]<br>
variable_ET_is_local: [true]<br>
variable_max_forwards: [69]<br>
variable_domain_name: [85.16.246.12]<br>
variable_dialed_extension: [1000]<br>
variable_sip_callee_id_number: [1111]<br>
variable_sip_callee_id_name: [hubu]<br>
variable_export_vars: [sip_callee_id_number,sip_callee_id_name]<br>
variable_current_application: [info]<br>
<div class="im"><br>
<br>
On 16.10.2009 18:18, Anthony Minessale wrote:<br>
> 1) you should update again there were a few issues.<br>
> 2) you can set the variable sip_callee_id_name and sip_callee_id number<br>
> on the inbound leg before you answer to control what it says.<br>
<br>
</div><div class="im">regards<br>
Helmut<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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