[asterisk-users] [External] [External] [External] [External] [External] Geo location 18.14.0-rc1 question
George Joseph
gjoseph at sangoma.com
Mon Aug 15 04:46:38 CDT 2022
On Sat, Aug 13, 2022 at 3:55 PM Dan Cropp <dan at amtelco.com> wrote:
> Thank you George.
>
>
>
> rc2 did fix the issue.
>
Whew.
>
>
> I am now able to program the variables in the location_info and pass
> values via the AMI Originate variables.
>
>
>
> Is there a way to make the location_info optional?
>
I _believe_ I can allow you to specify "location" parameters directly on a
profile which would be mutually exclusive with the location_reference
parameter. Let me look at it.
>
>
> On the same PJSIP Endpoint, we may need to originate calls with different
> requirements for the fields passed.
>
> For example
>
> Dialing a number for customer A, need to pass country, A1, A3, HNO, RD,
> STS, PC, FLR and ROOM.
>
> Dialing a number for customer B, need to pass country, A1, A2, and A3.
>
>
>
> Is it possible to program the Profile/Location to support values for all
> the settings, but have Asterisk ignore any settings where the value is
> blank?
>
A good idea. I can't think of any element that would need to be sent empty
but I'm wondering if I'll need to add a config option like
"suppress_empty_elements". Let me investigate.
>
>
> Right now, if I have HNO=${MYGEO_FLR} but the variable MYGEO_FLR is not
> set, it still passes the FLR in the sip as <ca:FLR/>
>
>
>
> I believe it’s perfectly fine to send the <ca:FLR/>
>
> My fear is some system may interpret a blank FLR differently than the FLR
> not being present at all.
>
>
>
> Dan
>
>
>
> *From:* asterisk-users <asterisk-users-bounces at lists.digium.com> *On
> Behalf Of *George Joseph
> *Sent:* Thursday, August 11, 2022 1:22 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users at lists.digium.com>
> *Subject:* Re: [External] [asterisk-users] [External] [External]
> [External] [External] Geo location 18.14.0-rc1 question
>
>
>
>
>
>
>
> On Thu, Aug 11, 2022 at 8:43 AM Dan Cropp <dan at amtelco.com> wrote:
>
> Thank you George.
>
>
>
> I am still running on asterisk 18.14.0-rc1 and have not retrieved the
> patches yet.
>
> Did this version have a bug with the variables?
>
>
>
> It's quite possible. RC2 was just released so you should try that.
>
>
>
> I’m trying the location_info and variables in the AMI Originate you
> recommended at the end of the previous e-mail.
>
>
>
> In case it’s not coming through correctly via e-mail, the variable names
> are preceeded with a single underscore in the AMI and in the location_info
> values.
>
>
>
>
>
> [IS_loc_5]
>
> type = location
>
> format = civicAddress
>
> location_info = country=${_MY_GEO_COUNTRY}
>
> location_info = A1=${_MY_GEO_NATIONAL_SUBDIVISION}
>
> location_info = A2=${_MY_GEO_NATSUB}
>
> location_info = A3=${_MY_GEO_CITY}
>
> location_info = HNO=${_MY_GEO_HNO}
>
> location_info = RD=${_MY_GEO_RD}
>
> location_info = STS=${_MY_GEO_STS}
>
> location_info = PC=${_MY_GEO_PC}
>
>
>
> [IS_prof_9]
>
> type = profile
>
> location_reference = IS_loc_5
>
> pidf_element = device
>
> profile_action = discard_incoming
>
> usage_rules = retransmission_allowed=yes
>
>
>
>
>
> [192.168.33.31]
>
> type = endpoint
>
> context = IS
>
> transport = transport1
>
> auth = auth14
>
> aors = 192.168.33.31
>
> accountcode = 20
>
> dtmf_mode = inband
>
> device_state_busy_at = 1600
>
> moh_passthrough = no
>
> identify_by = username,ip,header
>
> disallow = all
>
> allow = ulaw
>
> acl = acl1
>
> geoloc_incoming_call_profile = IS_prof_7
>
> geoloc_outgoing_call_profile = IS_prof_9
>
>
>
>
>
>
>
> Using a telnet session, I connect up via AMI and login. Then I attempt to
> Originate. The call goes through, but none of the location_info settings
> are being updated
>
>
>
> Action: Originate
>
> Channel: PJSIP/1234 at 192.168.33.31
>
> Exten: createcall
>
> Context: IS
>
> Priority: 1
>
> Timeout: 60000
>
> CallerID: John Smith <8005551234>
>
> Variable:
> _MY_GEO_COUNTRY=US,_MY_GEO_NATSUB=Florida,_MY_GEO_CITY=Orlando,_MY_GEO_HNO=100,_MY_GEO_RD=Main,_MY_GEO_STS=Street,CALLERID(num-pres)=allowed_passed_screen
>
> Async: true
>
>
>
>
>
> [08/10 15:04:12.470] DEBUG[1774] manager.c: Running action 'Originate'
>
> [08/10 15:04:12.470] DEBUG[1907] chan_pjsip.c: 1234 at 192.168.33.31
> Topology: <0:audio-0:audio:sendrecv (slin)>
>
> [08/10 15:04:12.470] DEBUG[1614] chan_pjsip.c: 1234 at 192.168.33.31
>
> [08/10 15:04:12.470] DEBUG[1614] res_pjsip_session.c: 192.168.33.31 1234
> Topology: <0:audio-0:audio:sendrecv (slin)>
>
> [08/10 15:04:12.470] DEBUG[1614] dsp.c: Setup tone 1100 Hz, 500 ms,
> block_size=160, hits_required=21
>
> [08/10 15:04:12.470] DEBUG[1614] dsp.c: Setup tone 2100 Hz, 2600 ms,
> block_size=160, hits_required=116
>
> [08/10 15:04:12.470] DEBUG[1614] chan_pjsip.c: 192.168.33.31
>
> [08/10 15:04:12.470] DEBUG[1614] chan_pjsip.c: Direct media no glare
> mitigation
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session/pjsip_session_caps.c:
> '192.168.33.31' Caps for outgoing audio call with pref 'remote_merge' -
> remote: (slin) local: (ulaw) joint: (ulaw)
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
>
> [08/10 15:04:12.471] DEBUG[1614] chan_pjsip.c:
>
> [08/10 15:04:12.471] DEBUG[1907] chan_pjsip.c: 192.168.33.31
>
> [08/10 15:04:12.471] DEBUG[1907] channel_internal_api.c: <initializing>:
> Formats: (none)
>
> [08/10 15:04:12.471] DEBUG[1907] channel_internal_api.c: Channel is being
> initialized or destroyed
>
> [08/10 15:04:12.471] DEBUG[1907] stasis.c: Creating topic. name:
> channel:1660161852.8, detail:
>
> [08/10 15:04:12.471] DEBUG[1907] stasis.c: Topic 'channel:1660161852.8':
> 0x7f19c0023210 created
>
> [08/10 15:04:12.471] DEBUG[1907] channel.c: Channel 0x7f19c00048d0
> 'PJSIP/192.168.33.31-00000005' allocated
>
> [08/10 15:04:12.471] DEBUG[1907] chan_pjsip.c: Topology:
> <0:audio-0:audio:sendrecv (ulaw)> Formats: (ulaw)
>
> [08/10 15:04:12.471] DEBUG[1907] chan_pjsip.c: Compatible? yes
>
> [08/10 15:04:12.471] DEBUG[1907] channel_internal_api.c:
> PJSIP/192.168.33.31-00000005: MultistreamFormats: (ulaw)
>
> [08/10 15:04:12.471] DEBUG[1907] channel_internal_api.c: Set native
> formats but not topology
>
> [08/10 15:04:12.471] DEBUG[1907] channel_internal_api.c:
> PJSIP/192.168.33.31-00000005: <0:audio-0:audio:sendrecv (ulaw)>
>
> [08/10 15:04:12.471] DEBUG[1907] channel_internal_api.c: Used provided
> topology
>
> [08/10 15:04:12.471] DEBUG[1907] chan_pjsip.c:
>
> [08/10 15:04:12.471] DEBUG[1907] chan_pjsip.c: Channel:
> PJSIP/192.168.33.31-00000005
>
> [08/10 15:04:12.471] DEBUG[1626] manager.c: Examining AMI event:
>
> Event: Newchannel^M
>
> Privilege: call,all^M
>
> Channel: PJSIP/192.168.33.31-00000005^M
>
> ChannelState: 0^M
>
> ChannelStateDesc: Down^M
>
> CallerIDNum: <unknown>^M
>
> CallerIDName: <unknown>^M
>
> ConnectedLineNum: <unknown>^M
>
> ConnectedLineName: <unknown>^M
>
> Language: en^M
>
> AccountCode: 20^M
>
> Context: IS^M
>
> Exten: s^M
>
> Priority: 1^M
>
> Uniqueid: 1660161852.8^M
>
> Linkedid: 1660161852.8^M
>
> ^M
>
>
>
> [08/10 15:04:12.471] DEBUG[1569] threadpool.c: Increasing threadpool
> stasis/pool's size by 1
>
> [08/10 15:04:12.471] DEBUG[1908] chan_pjsip.c:
> PJSIP/192.168.33.31-00000005 Topology: <0:audio-0:audio:sendrecv (ulaw)>
>
> [08/10 15:04:12.471] DEBUG[1908] chan_pjsip.c: 'call' task pushed
>
> [08/10 15:04:12.471] DEBUG[1774] manager.c: Examining AMI event:
>
> Event: Newchannel^M
>
> Privilege: call,all^M
>
> Channel: PJSIP/192.168.33.31-00000005^M
>
> ChannelState: 0^M
>
> ChannelStateDesc: Down^M
>
> CallerIDNum: <unknown>^M
>
> CallerIDName: <unknown>^M
>
> ConnectedLineNum: <unknown>^M
>
> ConnectedLineName: <unknown>^M
>
> Language: en^M
>
> AccountCode: 20^M
>
> Context: IS^M
>
> Exten: s^M
>
> Priority: 1^M
>
> Uniqueid: 1660161852.8^M
>
> Linkedid: 1660161852.8^M
>
> ^M
>
>
>
> [08/10 15:04:12.471] DEBUG[1774] manager.c: Examining AMI event:
>
> Event: Newexten^M
>
> Privilege: dialplan,all^M
>
> Channel: PJSIP/192.168.33.31-00000005^M
>
> ChannelState: 0^M
>
> ChannelStateDesc: Down^M
>
> CallerIDNum: <unknown>^M
>
> CallerIDName: <unknown>^M
>
> ConnectedLineNum: <unknown>^M
>
> ConnectedLineName: <unknown>^M
>
> Language: en^M
>
> AccountCode: 20^M
>
> Context: IS^M
>
> Exten: s^M
>
> Priority: 1^M
>
> Uniqueid: 1660161852.8^M
>
> Linkedid: 1660161852.8^M
>
> Extension: s^M
>
> Application: AppDial2^M
>
> AppData: (Outgoing Line)^M
>
> ^M
>
>
>
> [08/10 15:04:12.471] DEBUG[1774] manager.c: Examining AMI event:
>
> Event: VarSet^M
>
> Privilege: dialplan,all^M
>
> Channel: PJSIP/192.168.33.31-00000005^M
>
> ChannelState: 0^M
>
> ChannelStateDesc: Down^M
>
> CallerIDNum: <unknown>^M
>
> CallerIDName: <unknown>^M
>
> ConnectedLineNum: <unknown>^M
>
> ConnectedLineName: <unknown>^M
>
> Language: en^M
>
> AccountCode: 20^M
>
> Context: IS^M
>
> Exten: s^M
>
> Priority: 1^M
>
> Uniqueid: 1660161852.8^M
>
> Linkedid: 1660161852.8^M
>
> Variable: _MY_GEO_STS^M
>
> Value: Street^M
>
> ^M
>
>
>
> [08/10 15:04:12.471] DEBUG[1774] manager.c: Examining AMI event:
>
> Event: VarSet^M
>
> Privilege: dialplan,all^M
>
> Channel: PJSIP/192.168.33.31-00000005^M
>
> ChannelState: 0^M
>
> ChannelStateDesc: Down^M
>
> CallerIDNum: <unknown>^M
>
> CallerIDName: <unknown>^M
>
> ConnectedLineNum: <unknown>^M
>
> CallerIDName: <unknown>^M
>
> ConnectedLineNum: <unknown>^M
>
> ConnectedLineName: <unknown>^M
>
> Language: en^M
>
> AccountCode: 20^M
>
> Context: IS^M
>
> Exten: s^M
>
> Priority: 1^M
>
> Uniqueid: 1660161852.8^M
>
> Linkedid: 1660161852.8^M
>
> Variable: _MY_GEO_RD^M
>
> Value: Main^M
>
> ^M
>
>
>
> [08/10 15:04:12.471] DEBUG[1626] manager.c: Examining AMI event:
>
> Event: Newexten^M
>
> Privilege: dialplan,all^M
>
> Channel: PJSIP/192.168.33.31-00000005^M
>
> ChannelState: 0^M
>
> ChannelStateDesc: Down^M
>
> CallerIDNum: <unknown>^M
>
> CallerIDName: <unknown>^M
>
> ConnectedLineNum: <unknown>^M
>
> ConnectedLineName: <unknown>^M
>
> Language: en^M
>
> AccountCode: 20^M
>
> Context: IS^M
>
> Exten: s^M
>
> Priority: 1^M
>
> Uniqueid: 1660161852.8^M
>
> Linkedid: 1660161852.8^M
>
> Extension: s^M
>
> Application: AppDial2^M
>
> AppData: (Outgoing Line)^M
>
> ^M
>
>
>
> [08/10 15:04:12.471] DEBUG[1626] manager.c: Examining AMI event:
>
> Event: VarSet^M
>
> Privilege: dialplan,all^M
>
> Channel: PJSIP/192.168.33.31-00000005^M
>
> ChannelState: 0^M
>
> ChannelStateDesc: Down^M
>
> CallerIDNum: <unknown>^M
>
> CallerIDName: <unknown>^M
>
> ConnectedLineNum: <unknown>^M
>
> ConnectedLineName: <unknown>^M
>
> Language: en^M
>
> AccountCode: 20^M
>
> Context: IS^M
>
> Exten: s^M
>
> Priority: 1^M
>
> Uniqueid: 1660161852.8^M
>
> Linkedid: 1660161852.8^M
>
> Variable: _MY_GEO_STS^M
>
> Value: Street^M
>
> ^M
>
>
>
> ….
>
>
>
> [08/10 15:04:12.471] DEBUG[1614] chan_pjsip.c:
> PJSIP/192.168.33.31-00000005 Topology: <0:audio-0:audio:sendrecv (ulaw)>
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005: Processing streams
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005: Processing stream 0:audio-0:audio:sendrecv
> (ulaw)
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005 Adding position 0
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c: Creating new media
> session
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c: Setting media
> session as default for audio
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c: Done
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005 Stream: 0:audio-0:audio:sendrecv (ulaw)
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_sdp_rtp.c:
> PJSIP/192.168.33.31-00000005 Type: audio 0:audio-0:audio:sendrecv (ulaw)
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_sdp_rtp.c: Transport transport1
> bound to 0.0.0.0: Using it for RTP media.
>
> [08/10 15:04:12.471] DEBUG[1614] rtp_engine.c: Using engine 'asterisk' for
> RTP instance '0x7f19c80c3bf0'
>
> [08/10 15:04:12.471] DEBUG[1614] res_rtp_asterisk.c: (0x7f19c80c3bf0) RTP
> allocated port 17214
>
> [08/10 15:04:12.471] DEBUG[1614] res_rtp_asterisk.c: (0x7f19c80c3bf0) ICE
> creating session 0.0.0.0:17214 (17214)
>
> [08/10 15:04:12.471] DEBUG[1614] res_rtp_asterisk.c: (0x7f19c80c3bf0) ICE
> create
>
> [08/10 15:04:12.471] DEBUG[1614] res_rtp_asterisk.c: (0x7f19c80c3bf0) ICE
> add system candidates
>
> [08/10 15:04:12.471] DEBUG[1614] res_rtp_asterisk.c: (0x7f19c80c3bf0) ICE
> add candidate: 192.168.33.33:17214, 2130706431
>
> [08/10 15:04:12.471] DEBUG[1614] rtp_engine.c: RTP instance
> '0x7f19c80c3bf0' is setup and ready to go
>
> [08/10 15:04:12.471] DEBUG[1614] res_rtp_asterisk.c: (0x7f19c80c3bf0) ICE
> stopped
>
> [08/10 15:04:12.471] DEBUG[1614] res_rtp_asterisk.c: (0x7f19c80c3bf0) RTCP
> setup on RTP instance
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_sdp_rtp.c: RC: 1
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c: Handled
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005: Stream 0:audio-0:audio:sendrecv (ulaw) added
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005: Done with 0:audio-0:audio:sendrecv (ulaw)
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005: Adding bundle groups (if available)
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005: Copying connection details
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005: Processing media 0
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005: Media 0 reset
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005
>
> [08/10 15:04:12.471] DEBUG[1614] res_pjsip_session.c:
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005: Method is INVITE
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_geolocation.c:
> PJSIP/192.168.33.31-00000005
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_geolocation.c:
> PJSIP/192.168.33.31-00000005: There was no geoloc datastore
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_geolocation.c:
> PJSIP/192.168.33.31-00000005: There are no geoloc profiles on this channel
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_geolocation.c:
> PJSIP/192.168.33.31-00000005: There are now 1 geoloc profiles to be sent
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_geolocation.c: ep: 'IS_prof_9'
> EffectiveLoc:
> country=${_MY_GEO_COUNTRY},A1=${_MY_GEO_NATIONAL_SUBDIVISION},A2=${_MY_GEO_NATSUB},A3=${_MY_GEO_CITY},HNO=${_MY_GEO_HNO},RD=${_MY_GEO_RD},STS=${_MY\
>
> _GEO_STS},PC=${_MY_GEO_PC}
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_geolocation.c:
> PJSIP/192.168.33.31-00000005
>
> [08/10 15:04:12.472] DEBUG[1614] res_geolocation/geoloc_eprofile.c:
> PJSIP/192.168.33.31-00000005
>
> [08/10 15:04:12.472] DEBUG[1614] res_geolocation/geoloc_eprofile.c:
> PJSIP/192.168.33.31-00000005
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_COUNTRY
> result is '' from channel
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_COUNTRY
> result is '' from headp
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable
> _MY_GEO_NATIONAL_SUBDIVISION result is '' from channel
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable
> _MY_GEO_NATIONAL_SUBDIVISION result is '' from headp
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_NATSUB
> result is '' from channel
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_NATSUB
> result is '' from headp
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_CITY
> result is '' from channel
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_CITY
> result is '' from headp
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_HNO
> result is '' from channel
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_HNO
> result is '' from headp
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_RD
> result is '' from channel
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_RD
> result is '' from headp
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_STS
> result is '' from channel
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_STS
> result is '' from headp
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_PC
> result is '' from channel
>
> [08/10 15:04:12.472] DEBUG[1614] pbx_variables.c: Variable _MY_GEO_PC
> result is '' from headp
>
> [08/10 15:04:12.472] DEBUG[1614] res_geolocation/geoloc_civicaddr.c:
> PJSIP/192.168.33.31-00000005
>
> [08/10 15:04:12.472] DEBUG[1614] res_geolocation/geoloc_civicaddr.c:
> PJSIP/192.168.33.31-00000005: Done
>
> [08/10 15:04:12.472] DEBUG[1614] res_geolocation/geoloc_eprofile.c:
> PJSIP/192.168.33.31-00000005: Done
>
> [08/10 15:04:12.472] DEBUG[1614] res_geolocation/geoloc_eprofile.c:
> PJSIP/192.168.33.31-00000005: Done
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_geolocation.c: body:
> 0x7f19c8048390 0
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_geolocation.c: cid: '
> whjwo at 192.168.33.33' uri: 'cid:whjwo at 192.168.33.33' pidf_index: 0
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_geolocation.c:
> PJSIP/192.168.33.31-00000005: PIDF-LO added with cid 'whjwo at 192.168.33.33'
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_geolocation.c: ix: 0 of 1
> LocRef: cid:whjwo at 192.168.33.33
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_geolocation.c:
> PJSIP/192.168.33.31-00000005: Geolocation: <cid:whjwo at 192.168.33.33>
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip/pjsip_resolver.c: Performing
> SIP DNS resolution of target '192.168.33.31'
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip/pjsip_resolver.c: Transport
> type for target '192.168.33.31' is 'UDP transport'
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip/pjsip_resolver.c: Target
> '192.168.33.31' is an IP address, skipping resolution
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005 Event: TSX_STATE Inv State: CALLING
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005: Source of transaction state change is TX_MSG
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_session.c:
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005 TSX State: Calling Inv State: CALLING
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_session.c: Nothing delayed
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005 TSX State: Calling Inv State: CALLING
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_session.c: Topology: Pending:
> <0:audio-0:audio:sendrecv (ulaw)> Active: (null topology)
>
> [08/10 15:04:12.472] DEBUG[1614] res_pjsip_session.c:
>
> [08/10 15:04:12.472] DEBUG[1614] chan_pjsip.c: RC: 0
>
> [08/10 15:04:12.473] DEBUG[1613] res_pjsip/pjsip_distributor.c: Searching
> for serializer associated with dialog dlg0x7f19c80a2798 for Response msg
> 100/INVITE/cseq=21131 (rdata0x7f19b4001768)
>
> [08/10 15:04:12.473] DEBUG[1613] res_pjsip/pjsip_distributor.c: Found
> serializer pjsip/outsess/192.168.33.31-000000a6 associated with dialog
> dlg0x7f19c80a2798
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005 Method: INVITE Status: 100
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005 TSX State: Proceeding Inv State: CALLING
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005: Response is 100 Trying
>
> [08/10 15:04:12.473] DEBUG[1614] chan_pjsip.c:
> PJSIP/192.168.33.31-00000005: Status: 100
>
> [08/10 15:04:12.473] DEBUG[1614] chan_pjsip.c:
> PJSIP/192.168.33.31-00000005: Not queueing anything
>
> [08/10 15:04:12.473] DEBUG[1614] chan_pjsip.c:
> PJSIP/192.168.33.31-00000005
>
> [08/10 15:04:12.473] DEBUG[1614] chan_pjsip.c:
> PJSIP/192.168.33.31-00000005: Status: 100
>
> [08/10 15:04:12.473] DEBUG[1614] chan_pjsip.c:
> PJSIP/192.168.33.31-00000005
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c: Nothing delayed
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:
> PJSIP/192.168.33.31-00000005 TSX State: Proceeding Inv State: CALLING
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c: Topology: Pending:
> <0:audio-0:audio:sendrecv (ulaw)> Active: (null topology)
>
> [08/10 15:04:12.473] DEBUG[1614] res_pjsip_session.c:
>
>
>
> The SIP INVITE is as follows (all the civic address settings are blank,
> despite the AMI VarSet events showing the variables were set on the channel)
>
>
>
> [08/10 15:04:12.472] VERBOSE[1614] res_pjsip_logger.c: <--- Transmitting
> SIP request (2265 bytes) to UDP:192.168.33.31:5060 --->
>
> INVITE sip:1234 at 192.168.33.31 SIP/2.0^M
>
> Via: SIP/2.0/UDP 192.168.33.33:5060
> ;rport;branch=z9hG4bKPj37ef75eb-0ddc-4802-baa7-0921ff30ff8a^M
>
> From: "John Smith" <sip:8005551234 at 192.168.33.33
> >;tag=1720fabc-afd7-46ad-aa08-ff84140a7add^M
>
> To: <sip:1234 at 192.168.33.31>^M
>
> Contact: <sip:asterisk at 192.168.33.33:5060>^M
>
> Call-ID: c0a41f60-08d7-43e3-9901-39860a8001f5^M
>
> CSeq: 21131 INVITE^M
>
> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
> CANCEL, UPDATE, PRACK, MESSAGE, REFER^M
>
> Supported: 100rel, timer, replaces, norefersub, histinfo^M
>
> Session-Expires: 1800^M
>
> Min-SE: 90^M
>
> Geolocation: <cid:whjwo at 192.168.33.33>^M
>
> Geolocation-Routing: no^M
>
> Max-Forwards: 70^M
>
> User-Agent: Asterisk PBX 18.14.0-rc1^M
>
> Content-Type:
> multipart/mixed;boundary=8542a9ce-7ac4-47f7-8c17-d42771ebd564^M
>
> Content-Length: 1470^M
>
> ^M
>
> ^M
>
> --8542a9ce-7ac4-47f7-8c17-d42771ebd564^M
>
> Content-Type: application/sdp^M
>
> Content-Length: 181^M
>
> ^M
>
> v=0^M
>
> o=- 823099435 823099435 IN IP4 192.168.33.33^M
>
> s=Asterisk^M
>
> c=IN IP4 192.168.33.33^M
>
> t=0 0^M
>
> m=audio 17214 RTP/AVP 0^M
>
> a=rtpmap:0 PCMU/8000^M
>
> a=ptime:20^M
>
> a=maxptime:150^M
>
> a=sendrecv^M
>
> ^M
>
> --8542a9ce-7ac4-47f7-8c17-d42771ebd564^M
>
> Content-ID: <whjwo at 192.168.33.33>^M
>
> Content-Type: application/pidf+xml^M
>
> Content-Length: 1009^M
>
> ^M
>
> <?xml version="1.0"?>
>
> <presence xmlns="urn:ietf:params:xml:ns:pidf"
> xmlns:ca="urn:ietf:params:xml:ns:pidf:geopriv10:civicAddr"
> xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:fn="
> http://www.w3.org/2005/xpath-functions" xmlns:gbp="urn:ietf:params:xml:\
>
> ns:pidf:geopriv10:basicPolicy" xmlns:gml="http://www.opengis.net/gml"
> xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:gs="
> http://www.opengis.net/pidflo/1.0" xmlns:date="
> http://exslt.org/dates-and-times" entity="IS_prof_9">
>
> <dm:device>
>
> <gp:geopriv>
>
> <gp:location-info>
>
> <ca:civicAddress xml:lang="en">
>
> <ca:country/>
>
> <ca:A1/>
>
> <ca:A2/>
>
> <ca:A3/>
>
> <ca:HNO/>
>
> <ca:RD/>
>
> <ca:STS/>
>
> <ca:PC/>
>
> </ca:civicAddress>
>
> </gp:location-info>
>
> <gp:usage-rules>
>
> <gp:retransmission_allowed>yes</gp:retransmission_allowed>
>
> </gp:usage-rules>
>
> </gp:geopriv>
>
> <dm:timestamp>2022-08-10T20:04:12Z</dm:timestamp>
>
> </dm:device>
>
> </presence>
>
> ^M
>
> --8542a9ce-7ac4-47f7-8c17-d42771ebd564--^M
>
>
>
>
>
> *From:* asterisk-users <asterisk-users-bounces at lists.digium.com> *On
> Behalf Of *George Joseph
> *Sent:* Wednesday, August 10, 2022 1:34 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users at lists.digium.com>
> *Subject:* Re: [External] [asterisk-users] [External] [External]
> [External] Geo location 18.14.0-rc1 question
>
>
>
>
>
>
>
> On Wed, Aug 10, 2022 at 11:25 AM Dan Cropp <dan at amtelco.com> wrote:
>
> Thank you George.
>
>
>
> Looking forward to working with the changes. I will retrieve them when
> the next release candidate comes out.
>
>
>
>
>
> A quick question on using variables to pass custom Geo Location settings
> on via an AMI Originate.
>
>
>
>
>
> If my AMI originate request looks something like this…
>
> Action: Originate
>
> Channel: PJSIP/1234 at 192.168.x.x
>
> Exten: createcall
>
> Context: mycontext
>
> Priority: 1
>
> Timeout: 60000
>
> CallerID: John Smith <8005551234>
>
> Variable:
> _MY_GEO_COUNTRY=US,_MY_GEO_NATSUB=Florida,_MY_GEO_CITY=Orlando,_MY_GEO_HNO=100,_MY_GEO_RD=Main,_MY_GEO_STS=Street
>
> Async: true
>
>
>
> Do I need to program the location_variables in the profile like this?
>
>
>
> [1]
>
> type = profile
>
> pidf_element = device
>
> profile_action = discard_incoming
>
> usage_rules = retransmission_allowed=yes
>
> location_variables = country=${_MY_GEO_COUNTRY}
>
> location_variables = A1=${_MY_GEO_NATIONAL_SUBDIVISION}
>
> location_variables = A2=${_MY_GEO_NATSUB}
>
> location_variables = A3=${_MY_GEO_CITY}
>
> location_variables = HNO=${_MY_GEO_HNO}
>
> location_variables = RD=${_MY_GEO_RD}
>
> location_variables = STS=${_MY_GEO_STS}
>
> location_variables = PC=${_MY_GEO_PC}
>
>
>
> Or would I need to program the location_info_refinements in the profile to
> use those variables?
>
>
>
> location_info_refinement is what you want. location_variables defines
> *new* variables to use in addition to those on the channel. You'd use
> these if you had variables that for some reason you didn't want on the
> channel itself.
>
>
>
> However... The profile you defined above doesn't have a location
> reference to refine so you'd need at least a dummy location with a format
> of civicAddress.
>
>
>
> [mylog]
>
> type = location
>
> format = civicAddress
>
>
>
> Then in your profile...
>
> [1]
>
> type = profile
>
> pidf_element = device
>
> profile_action = discard_incoming
>
> usage_rules = retransmission_allowed=yes
>
> location_reference = myloc
>
> location_variables = country=${_MY_GEO_COUNTRY}
>
> location_variables = A1=${_MY_GEO_NATIONAL_SUBDIVISION}
>
> ...
>
>
>
> You can also do this which might actually be faster...
>
> [mylog]
>
> type = location
>
> format = civicAddress
>
> location_info = country=${_MY_GEO_COUNTRY},
> A1=${_MY_GEO_NATIONAL_SUBDIVISION}
>
> location_info = A2=${_MY_GEO_NATSUB}, ...
>
>
>
> [1]
>
> type = profile
>
> pidf_element = device
>
> profile_action = discard_incoming
>
> usage_rules = retransmission_allowed=yes
>
> location_reference = myloc
>
>
>
> This way you don't need to use location_info_refinement at all.
>
> IIRC this saves having to parse location_info_refinement and bounce it
> against
>
> the original location_info which could be empty.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Dan
>
>
>
>
>
> *From:* asterisk-users <asterisk-users-bounces at lists.digium.com> *On
> Behalf Of *George Joseph
> *Sent:* Wednesday, August 10, 2022 8:58 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users at lists.digium.com>
> *Subject:* Re: [External] [asterisk-users] [External] [External] Geo
> location 18.14.0-rc1 question
>
>
>
> Sorry for the delay but this turned out to be a bit more complex than I
> anticipated.
>
> There are reviews up on Gerrit for the 16 and 18 branches that address the
> issues below as well as clean up the implementation, plug some memory
> leaks, etc.
>
> 16: https://gerrit.asterisk.org/c/asterisk/+/18896
>
> 18: https://gerrit.asterisk.org/c/asterisk/+/18897
>
>
>
> I anticipate these will make it into the next set of release candidates
> which are due to be cut tomorrow.
>
>
>
> Give them a try.
>
>
>
> On Wed, Aug 3, 2022 at 1:51 PM George Joseph <gjoseph at sangoma.com> wrote:
>
> Looks like it'll be tomorrow before I can get the patch up. I ran into
> some strange issues.
>
>
>
> On Tue, Aug 2, 2022 at 1:43 PM Dan Cropp <dan at amtelco.com> wrote:
>
> Thank you George
>
>
>
> *From:* asterisk-users <asterisk-users-bounces at lists.digium.com> *On
> Behalf Of *George Joseph
> *Sent:* Tuesday, August 2, 2022 2:40 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users at lists.digium.com>
> *Subject:* Re: [External] [asterisk-users] [External] Geo location
> 18.14.0-rc1 question
>
>
>
>
>
>
>
> On Tue, Aug 2, 2022 at 1:35 PM George Joseph <gjoseph at sangoma.com> wrote:
>
>
>
>
>
> On Tue, Aug 2, 2022 at 1:13 PM Dan Cropp <dan at amtelco.com> wrote:
>
> Is the allow_routing setting on the geolocation Wiki Profile also not
> fully implemented?
>
>
>
> Well, 99% of the code is there. The 1% is parsing the config option. Not
> sure how I missed that.
>
> I'll have a patch up first thing in the morning UTC-6.
>
> I'll call it "allow_use_for_routing" in profile.
>
>
>
> Actually just "allow_routing_use"
>
>
>
>
>
>
>
> In the code, I see geolocation_routing used instead of allow_routing.
>
>
>
> Tried both and Asterisk indicates it cannot find suitable setting so it
> doesn’t create the profile object.
>
>
>
> Dan
>
>
>
> *From:* Dan Cropp
> *Sent:* Tuesday, August 2, 2022 10:04 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users at lists.digium.com>
> *Subject:* RE: [External] [asterisk-users] Geo location 18.14.0-rc1
> question
>
>
>
> Thank you George.
>
>
>
> *From:* asterisk-users <asterisk-users-bounces at lists.digium.com> *On
> Behalf Of *George Joseph
> *Sent:* Tuesday, August 2, 2022 9:57 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users at lists.digium.com>
> *Subject:* Re: [External] [asterisk-users] Geo location 18.14.0-rc1
> question
>
>
>
>
>
>
>
> On Tue, Aug 2, 2022 at 8:46 AM Dan Cropp <dan at amtelco.com> wrote:
>
> I believe I have everything configured correctly, but Asterisk is
> complaining about my configuration
>
>
>
> It is complaining about confidence settings.
>
>
>
> From the Asterisk Geolocation Implementation Wiki, I believe I have this
> set correctly.
>
>
>
> Sub-parameters:
>
> - value: A percentage indicating the confidence or "unknown".
> - pdf: "unknown", "normal" or "rectangular"
> Example: confidence = value=80, pdf=unknown
> If no confidence parameter is specified, the default is 95%.
> See RFC7459
> <https://wiki.asterisk.org/wiki/display/AST/Geolocation+Reference+Information#GeolocationReferenceInformation-rfc7459> for
> the exact definition of this parameter.
>
>
>
>
>
> [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option
> suitable for category 'IS_loc_1' named 'confidence' at line 12 of
>
> [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create
> an object of type 'location' with id 'IS_loc_1' from configuration file
> 'geolocation.conf'
>
>
>
> [IS_loc_1]
>
> type = location
>
> format = civicAddress
>
> confidence = value=95, pdf=unknown
>
> location_info = country=US,A1=Wisconsin,A3=Madison
>
> location_info = HNO=4800,RD=Main,STS=Drive,PC=53704
>
>
>
> Remove the confidence param for now. I documented it before I
> implemented it. :)
>
>
>
>
>
>
>
> Also seeing problems with location_refinement setting.
>
> Again, I believe my setting matches what is on the Asterisk Geolocation
> Implementation wiki.
>
>
>
> [08/02 09:30:03.724] ERROR[682944] config_options.c: Could not find option
> suitable for category 'IS_prof_20' named 'location_refinement' at line 56 of
>
> [08/02 09:30:03.724] ERROR[682944] res_sorcery_config.c: Could not create
> an object of type 'profile' with id 'IS_prof_20' from configuration file
> 'geolocation.conf'
>
>
>
> [IS_prof_20]
>
> type = profile
>
> profile_action = prefer_incoming
>
> pidf_element = person
>
> usage_rules = retransmission_allowed=no
>
> location_reference = IS_loc_22
>
> location_refinement = ROOM=292
>
> location_refinement = FLR=1
>
>
>
> Pffft. I renamed this to "location_info_refinement" to better match the
> "location_info" parameter in the Location object. I forgot to rename it in
> the wiki documentation. If you just change the name it should work.
>
>
>
>
>
>
>
>
>
>
>
> Dan
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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