[Freeswitch-users] FreeSwitch originate

Fri Jun 11 07:00:26 PDT 2010

Hi All,

I'm making slow and steady progress through this Freeswitch world.
I have a simple IVR running and I can listen to it when I call say 
extension 5000.
I did this by testing with X-lite by registering as extension 1001.

OS: Windows
Version: 1.0.6

The question now is I can't make FreeSwitch make a call. What is the 
command?
I tried:

originate sofia/internal/1001 at 10.1.1.4 1000

This is the output I get:

2010-06-11 21:54:30.682927 [NOTICE] switch_channel.c:669 New Channel 
sofia/inter
nal/1001 at 10.1.1.4 [e36dd3f5-a1ff-4c78-8e70-aca74533c068]
2010-06-11 21:54:30.686927 [NOTICE] switch_channel.c:669 New Channel 
sofia/exter
nal/0000000000 at 10.1.1.4 [db5c5175-da5c-4568-bbad-59343222b399]
2010-06-11 21:54:30.688927 [INFO] mod_dialplan_xml.c:418 Processing 
->1001 in co
ntext public
2010-06-11 21:54:30.690928 [NOTICE] switch_ivr.c:1447 Transfer 
sofia/external/00
00000000 at 10.1.1.4 to XML[1001 at default]
2010-06-11 21:54:30.691928 [INFO] mod_dialplan_xml.c:418 Processing 
->1001 in co
ntext default
2010-06-11 21:54:30.693928 [INFO] mod_dptools.c:965 CHANNEL_DATA:
Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [sofia/external/0000000000 at 10.1.1.4]
Unique-ID: [db5c5175-da5c-4568-bbad-59343222b399]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [ringing]
Channel-Read-Codec-Name: [G7221]
Channel-Read-Codec-Rate: [32000]
Channel-Write-Codec-Name: [G7221]
Channel-Write-Codec-Rate: [32000]
Caller-Username: [0000000000]
Caller-Dialplan: [XML]
Caller-Caller-ID-Number: [0000000000]
Caller-Network-Addr: [10.1.1.4]
Caller-ANI: [0000000000]
Caller-Destination-Number: [1001]
Caller-Unique-ID: [db5c5175-da5c-4568-bbad-59343222b399]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-RDNIS: [1001]
Caller-Channel-Name: [sofia/external/0000000000 at 10.1.1.4]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1276264470690928]
Caller-Channel-Created-Time: [1276264470686927]
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_uuid: [db5c5175-da5c-4568-bbad-59343222b399]
variable_sip_local_network_addr: [10.1.1.4]
variable_sip_network_ip: [10.1.1.4]
variable_sip_network_port: [5070]
variable_sip_received_ip: [10.1.1.4]
variable_sip_received_port: [5070]
variable_sip_via_protocol: [udp]
variable_sip_from_user: [0000000000]
variable_sip_from_uri: [0000000000 at 10.1.1.4]
variable_sip_from_host: [10.1.1.4]
variable_sip_from_user_stripped: [0000000000]
variable_sip_from_tag: [1XeZvDa011mBe]
variable_sofia_profile_name: [external]
variable_sip_Remote-Party-ID: 
[<sip:0000000000 at 10.1.1.4>;party=calling;screen=ye
s;privacy=off]
variable_sip_cid_type: [rpid]
variable_sip_full_via: [SIP/2.0/UDP 
10.1.1.4:5070;rport=5070;branch=z9hG4bKv0QyQ
4SSm612S]
variable_sip_full_from: ["" <sip:0000000000 at 10.1.1.4>;tag=1XeZvDa011mBe]
variable_sip_full_to: [<sip:1001 at 10.1.1.4>]
variable_sip_req_user: [1001]
variable_sip_req_uri: [1001 at 10.1.1.4]
variable_sip_req_host: [10.1.1.4]
variable_sip_to_user: [1001]
variable_sip_to_uri: [1001 at 10.1.1.4]
variable_sip_to_host: [10.1.1.4]
variable_sip_contact_user: [mod_sofia]
variable_sip_contact_port: [5070]
variable_sip_contact_uri: [mod_sofia at 10.1.1.4:5070]
variable_sip_contact_host: [10.1.1.4]
variable_channel_name: [sofia/external/0000000000 at 10.1.1.4]
variable_sip_call_id: [b304f1fd-f003-122d-c789-5d32389d672f]
variable_sip_user_agent: [FreeSWITCH-mod_sofia/1.0.6-exported]
variable_sip_via_host: [10.1.1.4]
variable_sip_via_port: [5070]
variable_sip_via_rport: [5070]
variable_switch_r_sdp: [v=0
o=FreeSWITCH 1276237680 1276237681 IN IP4 10.1.1.4
s=FreeSWITCH
c=IN IP4 10.1.1.4
t=0 0
m=audio 26790 RTP/AVP 115 107 9 0 8 3 101 13
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:107 G7221/16000
a=fmtp:107 bitrate=32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
]
variable_remote_media_ip: [10.1.1.4]
variable_remote_media_port: [26790]
variable_sip_use_codec_name: [G7221]
variable_sip_use_codec_fmtp: [bitrate=48000]
variable_sip_use_codec_rate: [32000]
variable_sip_use_codec_ptime: [20]
variable_read_codec: [G7221]
variable_read_rate: [32000]
variable_write_codec: [G7221]
variable_write_rate: [32000]
variable_endpoint_disposition: [RECEIVED]
variable_outside_call: [true]
variable_max_forwards: [69]
variable_current_application: [info]

-ERR NO_USER_RESPONSE

2010-06-11 21:54:30.693928 [NOTICE] switch_core_state_machine.c:185 
sofia/extern
al/0000000000 at 10.1.1.4 has executed the last dialplan instruction, 
hanging up.
2010-06-11 21:54:30.693928 [NOTICE] switch_core_state_machine.c:187 
Hangup sofia
/external/0000000000 at 10.1.1.4 [CS_EXECUTE] [NORMAL_CLEARING]
2010-06-11 21:54:30.695928 [NOTICE] sofia.c:4789 Hangup 
sofia/internal/1001 at 10.1
.1.4 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE]
2010-06-11 21:54:30.696928 [NOTICE] switch_core_session.c:1182 Session 6 
(sofia/
external/0000000000 at 10.1.1.4) Ended
2010-06-11 21:54:30.696928 [NOTICE] switch_core_session.c:1184 Close 
Channel sof
ia/external/0000000000 at 10.1.1.4 [CS_DESTROY]
2010-06-11 21:54:30.704928 [NOTICE] switch_core_session.c:1182 Session 5 
(sofia/
internal/1001 at 10.1.1.4) Ended
2010-06-11 21:54:30.704928 [NOTICE] switch_core_session.c:1184 Close 
Channel sof
ia/internal/1001 at 10.1.1.4 [CS_DESTROY]

I somehow don't think I'm doing the right thing....
Pretty much all Freeswitch XML is still default except the password.

Is Xlite allowed to be installed on the same computer as the freeswitch 
server?

At this stage I'm not testing with any external gateways, just trying to 
get something working in my internal network.
So should by domain be the IP of the server (10.1.1.4) or localhost?

Do I need "mod_dialplan_directory? or just mod_dialplan_xml?

**Confused**

Thanks,

Oliver