Added PJSIP support for Asterisk 16
git-svn-id: svn://192.168.202.10@3505 3d104415-ff17-0410-8863-d5cf3c621b8a
This commit is contained in:
mattf
@@ -1,4 +1,4 @@
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Sending email after agent dispositions a call Started: 2015-08-06 Updated: 2021-08-23
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Sending email after agent dispositions a call Started: 2015-08-06 Updated: 2021-08-25
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These instructions will show you how to set up automatic emails to be sent out when an agent dispositions a call with a specific defined "sale status".
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@@ -132,5 +132,58 @@ We hope you enjoy your stuff, and happy roadrunner chasing!
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ACME Support Department
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312-555-1212
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email_body_end
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HERE IS AN EXAMPLE NO AGENT CALL URL ENTRY FROM AN IN-GROUP FOR THIS FEATURE:
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VARhttp://server/agc/dispo_send_email.php?container_id=INGROUP_NA_EMAIL-test&lead_id=--A--lead_id--B--&call_id=--A--call_id--B--&dispo=--A--dispo--B--&user=NOAGENTURL&pass=--A--call_id--B--&sale_status=TIMEOT---NANQUE---DROP---AFTHRS&log_to_file=1
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HERE IS AN EXAMPLE SETTINGS CONTAINER TO SEND OUT AN EMAIL FROM THE ABOVE EXAMPLE:
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; destination email
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email_to => vicidialemailtesting@gmail.com
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; sending email address
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email_from => customertesting@vicidial.com
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; subject of the email
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email_subject => --A--dispo--B-- - 001 - --A--phone_number--B--
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; body of the email
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email_body_begin =>
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Hello,
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This email is to let you know a call from --A--phone_number--B-- has gone unanswered from in-group test.
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Here is a summary call information:
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In group: test In-Group
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phone_number: --A--phone_number--B--
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lead_id: --A--lead_id--B--
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list_id: --A--list_id--B--
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list_name: --A--list_name--B--
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first_name: --A--first_name--B--
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last_name: --A--last_name--B--
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address1: --A--address1--B--
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address2: --A--address2--B--
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city: --A--city--B--
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state: --A--state--B--
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postal_code: --A--postal_code--B--
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gender: --A--gender--B--
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date_of_birth: --A--date_of_birth--B--
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alt_phone: --A--alt_phone--B--
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email: --A--email--B--
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comments: --A--comments--B--
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user: --A--user--B--
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closer: --A--user--B--
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call_id: --A--call_id--B--
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ACME Call Center
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email_body_end
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@@ -0,0 +1,45 @@
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[global]
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type = global
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contact_expiration_check_interval = 30
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debug = no
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default_from_user = Asterisk
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default_outbound_endpoint = default_outbound_endpoint
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default_realm = Asterisk
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default_voicemail_extension = 8500
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disable_multi_domain = false
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endpoint_identifier_order = ip,username,anonymous
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ignore_uri_user_options = false
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keep_alive_interval = 90
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max_forwards = 70
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max_initial_qualify_time = 0
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mwi_disable_initial_unsolicited = false
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mwi_tps_queue_high = 500
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mwi_tps_queue_low = -1
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norefersub = yes
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;regcontext = myregs
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send_contact_status_on_update_registration = yes
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taskprocessor_overload_trigger = global
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unidentified_request_count = 5
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unidentified_request_period = 5
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unidentified_request_prune_interval = 30
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use_callerid_contact = no
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user_agent = Asterisk
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[transport-udp]
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type = transport
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protocol = udp
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bind = 0.0.0.0:5060
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external_media_address = 190.124.44.160
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external_signaling_address = 190.124.44.160
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local_net = 192.168.0.0/255.255.0.0
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local_net = 10.0.0.0/255.0.0.0
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local_net = 172.16.0.0/12
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local_net = 169.254.0.0/255.255.0.0
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tos = cs3
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[transport-wss]
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type = transport
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protocol = wss
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bind = 0.0.0.0
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#include "pjsip-vicidial.conf"
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@@ -0,0 +1,165 @@
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; PJSIP Wizard Configuration Samples and Quick Reference
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;
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; This file has several very basic configuration examples, to serve as a quick
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; reference to jog your memory when you need to write up a new configuration.
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; It is not intended to teach PJSIP configuration or serve as an exhaustive
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; reference of options and potential scenarios.
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;
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; This file has two main sections.
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; First, manually written examples to serve as a handy reference.
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; Second, a list of all possible PJSIP config options by section. This is
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; pulled from the XML config help. It only shows the synopsis for every item.
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; If you want to see more detail please check the documentation sources
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; mentioned at the top of this file.
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; ============================================================================
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; NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
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;
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; This file does not maintain the complete option documentation.
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; ============================================================================
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; Documentation
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;
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; The official documentation is at http://wiki.asterisk.org
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; You can read the XML configuration help via Asterisk command line with
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; "config show help res_pjsip_config_wizard", then you can drill down through
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; the various sections and their options.
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;
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;============EXAMPLE WIZARD CONFIGURATION FOR A PHONE=======================
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; This config would create an endpoint, aor with dynamic contact, inbound
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; auth, a phoneprov object and a dialplan hint for extension 1000.
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;[myphone]
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;type = wizard
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;accepts_auth = yes
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;accepts_registrations = yes
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;has_phoneprov = yes
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;transport = ipv4
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;has_hint = yes
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;hint_exten = 1000
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;inbound_auth/username = testname
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;inbound_auth/password = test password
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;endpoint/allow = ulaw
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;endpoint/context = default
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;phoneprov/MAC = 001122aa4455
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;phoneprov/PROFILE = profile1
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;============EXAMPLE WIZARD CONFIGURATION FOR AN ITSP TRUNK=================
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; This ITSP has 2 servers available and requires registration.
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; This config would create an endpoint, an aor with 2 static contacts, an
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; outbound auth, an identify with 2 matches, and 2 registrations.
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;[mytrunk]
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;type = wizard
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;sends_auth = yes
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;sends_registrations = yes
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;transport = ipv4
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;remote_hosts = sip1.myitsp.com:5060,sip2.myitsp.com:5060
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;outbound_auth/username = testname
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;outbound_auth/password = test password
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;endpoint/allow = ulaw
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;endpoint/context = default
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;========================WIZARD SECTION OPTIONS===============================
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;[wizard]
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; SYNOPSIS: Provides configuration wizard for common scenarios.
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;sends_auth= ; Will create an outbound auth object for the endpoint and
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; registration.
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; If yes, outbound/username must be specified.
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; (default = "no")
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;accepts_auth= ; Will create an inbound auth object for the endpoint.
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; If yes, inbound/username must be specified.
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; (default = "no")
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;sends_registrations= ; Will create an outbound registration object and an
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; identify match for each host in remote_hosts (which
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; must be specified).
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; sends_auth must also be specified.
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; (default: "no")
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;sends_line_with_registrations= ; Setting this to true will cause the wizard to
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; skip the creation of an identify object to match
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; incoming requests to the endpoint and instead add the
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; line and endpoint parameters to the outbound
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; registration object.
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; (default: "no")
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;accepts_registrations= ; Will create an aor with dynamic contacts which will
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; accept registrations.
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; accepts_auth must also be specified.
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; (default: "no")
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;remote_hosts= ; A comma separated list of remote hosts in the form of
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; <ipaddress | hostname>[:port] [,<ipaddress | hostname>[:port] ] ...
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; If specified, a static contact for each host will be created
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; in the aor. If accepts_registrations is no, an identify
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; object is also created with a match line for each remote host.
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; If an aor/contact or match/identify is explicitly supplied,
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; remote_hosts will not be used to automatically create contacts
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; or matches respectively.
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; (default: "")
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;outbound_proxy= ; Setting this is a shortcut for setting
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; endpoint/outbound_proxy
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; aor/outbound_proxy
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; registration/outbound_proxy
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;transport= ; The transport to use for the endpoint and registrations
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; (default: the pjsip default)
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;server_uri_pattern= ; The pattern used to construct the registration
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; server_uri. The replaceable parameter ${REMOTE_HOST} isa
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; available for use.
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; (default: "sip:${REMOTE_HOST}")
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;client_uri_pattern= ; The pattern used to construct the registration client_uri.
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; The replaceable parameters ${REMOTE_HOST} and ${USERNAME}
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; are available for use.
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; (default: "sip:${USERNAME}@${REMOTE_HOST}")
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;contact_pattern= ; The pattern used to construct the aor contact.
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; The replaceable parameter ${REMOTE_HOST} is available
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; for use.
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; (default: "sip:${REMOTE_HOST}")
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;has_phoneprov= ; Will create a phoneprov object.
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; If yes, phoneprov/MAC must be specified.
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; (default: "no")
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;has_hint= ; Create hint and optionally a default application.
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; (default: "no")
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;hint_context ; Any hints created for this wizard will be placed in this
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; context.
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; (default: endpoint/context)
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;hint_exten ; If specified, a PJSIP/<wizard_id> hint will be created
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; for this extension in 'hint_context'.
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; context.
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; (default: none)
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;hint_application ; If specified, an extension will be placed in 'hint_context'
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; at priority 1 that calls this application. Could be any
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; valid dialplan expression like
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; "Gosub(stdexten,${EXTEN},1(${HINT}))"
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; (default: "Dial(${HINT})")
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;endpoint/<param> ; Any parameters to be passed directly to and validated
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;aor/<param> ; by their respective objects.
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;inbound_auth/<param>
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;outbound_auth/<param>
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;identify/<param>
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;registration/<param>
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;phoneprov/<param>
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;type= ; Must be of type wizard (default: "")
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#include "pjsip_wizard-vicidial.conf"
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@@ -0,0 +1,104 @@
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[general]
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context=trunkinbound ; Default context for incoming calls
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allowguest=no ; Allow or reject guest calls (default is yes)
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allowoverlap=no ; Disable overlap dialing support. (Default is yes)
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;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
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;realm=mydomain.tld ; Realm for digest authentication
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bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
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bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
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srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;domain=mydomain.tld ; Set default domain for this host
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;pedantic=yes ; Enable checking of tags in headers,
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;tos_sip=cs3 ; Sets TOS for SIP packets.
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;tos_audio=ef ; Sets TOS for RTP audio packets.
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;tos_video=af41 ; Sets TOS for RTP video packets.
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;maxexpiry=3600 ; Maximum allowed time of incoming registrations
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;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
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;defaultexpiry=120 ; Default length of incoming/outgoing registration
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;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
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;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
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;checkmwi=10 ; Default time between mailbox checks for peers
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;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
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;vmexten=voicemail ; dialplan extension to reach mailbox sets the
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disallow=all ; First disallow all codecs
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allow=ulaw ; Allow codecs in order of preference
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allow=gsm
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mohinterpret=default
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mohsuggest=default
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language=en ; Default language setting for all users/peers
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relaxdtmf=yes ; Relax dtmf handling
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trustrpid = no ; If Remote-Party-ID should be trusted
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sendrpid = yes ; If Remote-Party-ID should be sent
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progressinband=no ; If we should generate in-band ringing always
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;useragent=Asterisk PBX ; Allows you to change the user agent string
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;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
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;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
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dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
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;compactheaders = yes ; send compact sip headers.
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videosupport=no ; Turn on support for SIP video. You need to turn this on
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;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
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callevents=yes ; generate manager events when sip ua
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;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
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;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
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;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
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;regcontext=sipregistrations
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rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
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;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
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;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
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;sipdebug = yes ; Turn on SIP debugging by default, from
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;recordhistory=yes ; Record SIP history by default
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;dumphistory=yes ; Dump SIP history at end of SIP dialogue
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;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
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;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
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notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
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notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
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limitonpeers = yes ; Apply call limits on peers only. This will improve
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;t38pt_udptl = yes ; Default false
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;register => 1234:password@mysipprovider.com
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;registertimeout=20 ; retry registration calls every 20 seconds (default)
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;registerattempts=10 ; Number of registration attempts before we give up
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;externip = 192.168.1.1 ; Address that we're going to put in outbound SIP
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;externhost=test.test.com ; Alternatively you can specify a domain
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;externrefresh=10 ; How often to refresh externhost if
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localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
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localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
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localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
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localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
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nat=force_rport,comedia ; Global NAT settings (Affects all peers and users)
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canreinvite=no ; Asterisk by default tries to redirect the
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;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
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;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
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;rtsavesysname=yes ; Save systemname in realtime database at registration
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;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
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;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
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;ignoreregexpire=yes ; Enabling this setting has two functions:
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;domain=mydomain.tld,mydomain-incoming
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;domain=1.2.3.4 ; Add IP address as local domain
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;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
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;autodomain=yes ; Turn this on to have Asterisk add local host
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;fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
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jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
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jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
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jbmaxsize = 100 ; Max length of the jitterbuffer in milliseconds.
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jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
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jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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qualify=yes ; By default, qualify all peers at 2000ms
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limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers
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#include sip-vicidial.conf
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; register SIP account on remote machine if using SIP trunks
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; register => testSIPtrunk:test@10.10.10.16:5060
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;
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; setup account for SIP trunking:
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; [SIPtrunk]
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; disallow=all
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; allow=ulaw
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; allow=alaw
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; type=friend
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; username=testSIPtrunk
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; secret=test
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; host=10.10.10.16
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; dtmfmode=inband
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; qualify=1000
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Reference in New Issue
Block a user