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
Example: sofia/profile/user@remotedomain.com /b On Jun 23, 2008, at 5:36 AM, Sandeep Ammaladinna wrote:
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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
route your inbound call to the "info" app. You will see the dozens of things you can parse in the dialplan. You can make a single extension that determines if it is a local domain or not and formulate the dial string. You have a lot of reading to do before you are authorized to comment on if it's a bad design or not. http://wiki.freeswitch.org/wiki/Dialplan_XML On Mon, Jun 23, 2008 at 7:44 AM, Brian West <brian@freeswitch.org> wrote:
Example: sofia/profile/user@remotedomain.com
/b
On Jun 23, 2008, at 5:36 AM, Sandeep Ammaladinna wrote:
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 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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@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@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@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@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@192.168.50.229] variable_sip_req_host: [192.168.50.229] variable_sip_to_user: [9992] variable_sip_to_uri: [9992@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@192.168.50.21:18394] variable_sip_contact_host: [192.168.50.21] variable_channel_name: [sofia/internal/1160@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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sandeep Ammaladinna Sent: Monday, June 23, 2008 3:36 AM To: freeswitch-users@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
participants (4)
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Anthony Minessale
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Brian West
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Michael Collins
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Sandeep Ammaladinna