[Freeswitch-users] Outbound calls to external domain

[Michael Collins]

Sandeep,

 

In addition to the stuff Brian and Anthony asked you about, please do
these steps:

#1 - if you didn't make any modifications to your
conf/dialplan/default.xml file then delete it

#2 - in the fs source directory do "make current" which will update
everything

#3 - when the update is done then do "make samples"

 

After you have these steps done you'll be 100% updated, and you'll have
a new extension defined in the default configs.  Brian and I added
extension 9992, which simply does an "info" and then hangs up.  If you
have a phone reg'd to fs then just pickup and dial 9992 and watch the
CLI.  You should see something like this:

 

2008-06-23 14:02:32 [INFO] mod_dialplan_xml.c:222 dialplan_hunt()
Processing Mikey->9992 at default

2008-06-23 14:02:32 [INFO] mod_dptools.c:909 info_function()
CHANNEL_DATA:

Channel-State: [CS_EXECUTE]

Channel-State-Number: [4]

Channel-Name: [sofia/internal/1160 at 192.168.50.229]

Unique-ID: [99f379ca-c06b-4a8e-b73c-3fa19c03dca8]

Call-Direction: [inbound]

Answer-State: [answered]

Channel-Read-Codec-Name: [PCMU]

Channel-Read-Codec-Rate: [8000]

Channel-Write-Codec-Name: [PCMU]

Channel-Write-Codec-Rate: [8000]

Caller-Username: [1160]

Caller-Dialplan: [XML]

Caller-Caller-ID-Name: [Mikey]

Caller-Caller-ID-Number: [1160]

Caller-Network-Addr: [192.168.50.21]

Caller-Destination-Number: [9992]

Caller-Unique-ID: [99f379ca-c06b-4a8e-b73c-3fa19c03dca8]

Caller-Source: [mod_sofia]

Caller-Context: [default]

Caller-Channel-Name: [sofia/internal/1160 at 192.168.50.229]

Caller-Profile-Index: [1]

Caller-Channel-Created-Time: [1214254952218289]

Caller-Channel-Answered-Time: [1214254952330188]

Caller-Channel-Hangup-Time: [0]

Caller-Channel-Transfer-Time: [0]

Caller-Screen-Bit: [yes]

Caller-Privacy-Hide-Name: [no]

Caller-Privacy-Hide-Number: [no]

variable_sip_received_ip: [192.168.50.21]

variable_sip_received_port: [18394]

variable_sip_authorized: [true]

variable_sip_mailbox: [1160]

variable_sip_auth_username: [1160]

variable_sip_auth_realm: [192.168.50.229]

variable_mailbox: [1160]

variable_record_stereo: [true]

variable_accountcode: [1160]

variable_user_context: [default]

variable_effective_caller_id_name: [Michael S Collins]

variable_effective_caller_id_number: [5597993757]

variable_sip_from_user: [1160]

variable_sip_from_uri: [1160 at 192.168.50.229]

variable_sip_from_host: [192.168.50.229]

variable_sip_from_user_stripped: [1160]

variable_sip_from_tag: [3678fa75]

variable_sofia_profile_name: [internal]

variable_sofia_profile_domain_name: [192.168.50.229]

variable_sip_req_user: [9992]

variable_sip_req_uri: [9992 at 192.168.50.229]

variable_sip_req_host: [192.168.50.229]

variable_sip_to_user: [9992]

variable_sip_to_uri: [9992 at 192.168.50.229]

variable_sip_to_host: [192.168.50.229]

variable_sip_contact_user: [1160]

variable_sip_contact_port: [18394]

variable_sip_contact_uri: [1160 at 192.168.50.21:18394]

variable_sip_contact_host: [192.168.50.21]

variable_channel_name: [sofia/internal/1160 at 192.168.50.229]

variable_sip_call_id: [YTA1NTY5ZWY5NWQ4Y2EzZGJiNmVlYTVjOWQxYzhiMWM.]

variable_sip_user_agent: [X-Lite release 1011s stamp 41150]

variable_sip_via_host: [192.168.50.21]

variable_sip_via_port: [18394]

variable_sip_via_rport: [18394]

variable_max_forwards: [70]

variable_presence_id: [1160 at 192.168.50.229]

variable_switch_r_sdp: [v=0

o=- 7 2 IN IP4 192.168.50.21

s=CounterPath X-Lite 3.0

c=IN IP4 192.168.50.21

t=0 0

m=audio 34100 RTP/AVP 107 119 100 106 0 105 98 8 101

a=rtpmap:107 BV32/16000

a=rtpmap:119 BV32-FEC/16000

a=rtpmap:100 SPEEX/16000

a=rtpmap:106 SPEEX-FEC/16000

a=rtpmap:105 SPEEX-FEC/8000

a=rtpmap:98 iLBC/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=alt:1 1 : SZaASTso Ck/IwSg9 192.168.50.21 34100

]

variable_remote_media_ip: [192.168.50.21]

variable_remote_media_port: [34100]

variable_read_codec: [PCMU]

variable_read_rate: [8000]

variable_write_codec: [PCMU]

variable_write_rate: [8000]

variable_open: [true]

variable_use_profile: [nat]

variable_numbering_plan: [US]

variable_default_gateway: [asterlink.com]

variable_default_area_code: [918]

variable_user_name: [default]

variable_domain_name: [192.168.50.229]

variable_local_media_ip: [192.168.50.229]

variable_local_media_port: [34100]

variable_sip_nat_detected: [true]

variable_endpoint_disposition: [ANSWER]

 

Anything prefixed with "variable_" is a channel variable.  For example,
"variable_default_area_code" tells you what's in the channel variable
named "default_area_code" - as you can see, there are lots and lots of
channel variables that are preset for your use.  Also, if you set your
own channel variables (e.g with the dialplan "set" application) then
you'll see those listed as well.

 

Write it down: the "info" app is your friend!  :-)

-MC

 

 

________________________________

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Sandeep Ammaladinna
Sent: Monday, June 23, 2008 3:36 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Outbound calls to external domain

 

Hi all,

 

  Is it a mandatory requirement that for all outbound calls to an
external domain (domains that freeswitch is not responsible for), an
entry in the dial plan is required. 

 To be more precise, is it a pre condition that the name of the external
domain need to be know before making a call. If YES, then I feel it is a
bad design, because each time a user wants to call a user from different
external domain a new entry in dial plan is required. If the answer is
NO, please tell me what is the configuration required apart from DNS
records, to be there at the freeswitch.

 

Thanks and Regards,

-Sandeep A 

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