[Freeswitch-users] Qustion about INFO messages after Connect/Answer

Mon Oct 19 08:00:49 PDT 2009

you only need to "set" it on the inbound leg and you must answer and bridge
it somewhere.

On Mon, Oct 19, 2009 at 9:47 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
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> Hello Anthony,
>
> I updated and restarted my test FS to "FreeSWITCH Version 1.0.trunk
> (15174M)". Callee's experience didn't change:
>
> > 1. Phone rings: caller's displayname
> > 2. Callee picks up: switching from dislayname to unknown
> > 3. Switching from unknown to displayname
>
> 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 at 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 at 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 at 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 at 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 at 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 at 85.16.245.206:1024]
> variable_sip_contact_host: [85.16.245.206]
> variable_channel_name: [sofia/internal/1001 at 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 at 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 at 8000h@20i,PCMA at 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:
> > 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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-- 
Anthony Minessale II

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