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t.mahe at telemaque.fr Guest
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Posted: Mon Oct 19, 2009 10:19 am Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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Hi Helmut,
Just to add my 2 cents to the discussion, I have the same behaviour there...
Regards,
Gled.
Helmut Kuper a écrit :
Quote: | Hello Anthony,
I updated and restarted my test FS to "FreeSWITCH Version 1.0.trunk
(15174M)". Callee's experience didn't change:
Quote: | 1. Phone rings: caller's displayname
2. Callee picks up: switching from dislayname to unknown
3. Switching from unknown to displayname
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I used the two chvars you mentioned, set them via "set" and as well via
"export" but no change (neither on caller's nor in callee's display nor
in SIP INFO messages)
My dialplan portion for this is:
<extension name="Local_Extension">
<condition field="${ET_is_local}" expression="^true$">
<action application="set"
data="dialed_extension=${destination_number}"/>
<action application="set" data="sip_callee_id_number=1111"/>
<action application="set" data="sip_callee_id_name=hubu"/>
<action application="export" data="sip_callee_id_number=1111"/>
<action application="export" data="sip_callee_id_name=hubu"/>
<action application="info"/>
[...]
Here is the output of the info app after setting those chvars:
[INFO] mod_dptools.c:961 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/1001@85.16.246.12:5061]
Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [ringing]
Caller-Username: [1001]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [1001 an PBX1]
Caller-Caller-ID-Number: [1001]
Caller-Network-Addr: [85.16.245.206]
Caller-Destination-Number: [1000]
Caller-Unique-ID: [4b143750-bcbd-11de-9f91-c9cd82739033]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-RDNIS: [1000]
Caller-Channel-Name: [sofia/internal/1001@85.16.246.12:5061]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1255963206242587]
Caller-Channel-Created-Time: [1255963206214959]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_sip_received_ip: [85.16.245.206]
variable_sip_received_port: [1024]
variable_sip_via_protocol: [udp]
variable_sip_authorized: [true]
variable_sip_from_user: [1001]
variable_sip_from_port: [5061]
variable_sip_from_uri: [1001@85.16.246.12:5061]
variable_sip_from_host: [85.16.246.12]
variable_sip_from_user_stripped: [1001]
variable_sip_from_tag: [snfuiue6ga]
variable_sofia_profile_name: [internal]
variable_sip_req_params: [user=phone]
variable_sip_req_user: [1000]
variable_sip_req_port: [5061]
variable_sip_req_uri: [1000@85.16.246.12:5061]
variable_sip_req_host: [85.16.246.12]
variable_sip_to_params: [user=phone]
variable_sip_to_user: [1000]
variable_sip_to_port: [5061]
variable_sip_to_uri: [1000@85.16.246.12:5061]
variable_sip_to_host: [85.16.246.12]
variable_sip_contact_params: [line=eg3wp69a]
variable_sip_contact_user: [1001]
variable_sip_contact_port: [1024]
variable_sip_contact_uri: [1001@85.16.245.206:1024]
variable_sip_contact_host: [85.16.245.206]
variable_channel_name: [sofia/internal/1001@85.16.246.12:5061]
variable_sip_call_id: [3c2d2d8f9a49-edzr2i2iezjp]
variable_sip_user_agent: [snom820/8.2.16]
variable_sip_via_host: [85.16.245.206]
variable_sip_via_port: [1024]
variable_sip_via_rport: [1024]
variable_presence_id: [1001@85.16.246.12]
variable_sip_h_X-Serialnumber: [0004134002CB]
variable_sip_h_P-Key-Flags: [resolution="31x13", keys="4"]
variable_switch_r_sdp: [v=0
o=root 1331667919 1331667919 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 62882 RTP/SAVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:m6fas/KsLF57r9RnU7X0WEWeJw9Y6+a66YUIf9Dc
a=ptime:20
m=audio 62882 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
]
variable_ep_codec_string: [G722@8000h@20i,PCMA@8000h@20i]
variable_endpoint_disposition: [DELAYED NEGOTIATION]
variable_ET_is_local: [true]
variable_max_forwards: [69]
variable_domain_name: [85.16.246.12]
variable_dialed_extension: [1000]
variable_sip_callee_id_number: [1111]
variable_sip_callee_id_name: [hubu]
variable_export_vars: [sip_callee_id_number,sip_callee_id_name]
variable_current_application: [info]
On 16.10.2009 18:18, Anthony Minessale wrote:
Quote: | 1) you should update again there were a few issues.
2) you can set the variable sip_callee_id_name and sip_callee_id number
on the inbound leg before you answer to control what it says.
|
regards
Helmut
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helmut.kuper at ewetel.de Guest
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Posted: Mon Oct 19, 2009 11:09 am Post subject: [Freeswitch-users] Qustion about INFO messages after Connect |
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Hash: SHA1
Hi Anthony,
just to make it clear: my goal is to avoid to see something (newly
introduced since 1 or 2 weeks) like "1/a/890327" for outgoing in the
caller's display after answering the call (for openzap calls). I want
simply e.g. "89327". I don't want to put the call into "answer" state
before the called side demands it. So I'm not sure if sip_callee_* is
the right way to follow as long as it needs the answer state.
regards
helmut
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