Hello FreePBX gurus :).
As a matter of fact we started discussion in IVR Question, which is pretty old topic. Because nobody answered for a long time I am oppening a new topic here.
The problem is:
- I have two intracompany sites with FreePBX boxes on each site. The servers are interconnected via a SIP trunk. All this happen inside the company cloud without direct access to the internet. On the FreePBX on site1 the IVR is composed. All calls between two sites are fine.
- Because we do not have a public telephone number (to which IVR to be assigned) I created a Misk application in order to reach the IVR by dialing an extension.
We have two FreePBX (computer based FreePBX 13.0.197.28) located in two different locations. Of course we have a trunk between them. All calls are perfect … exept the IVR.
If I try to dial the IVR extension from the phone connected to the FreePBX on which the IVR is created - everything is working just fine.
But If I try to dial the IVR extension from the phone connected to the secong FreePBX am getting a message “The destination cannot be reached”.
What I am doing wrong?
Thanks in advance for all opinions, sugestions … criticism … everithing
The #dicko advised me:
Your tie line trunk is from-internal?
the from-internal context does not include ivr’s,
the dialplan includes
Include => 'from-internal-additional-custom'
So you could try adding ivr-n
(where n is the one you want to be available ) to that context
[from-internal-additional-custom]
include => ivr-n
include => ivr-m
O.K. so as #dicko advised me:
- I checked the /etc/asterisk/ extensions_additional.conf
the records say -
[ivr-1] ; Welcome IVR
include => ivr-1-custom
include => from-did-direct-ivr - I add to the /etc/asterisk/extensions_custom.conf
[from-internal-additional-custom]
include => ivr-1 - So far I do not have responce from the remote office (will have after …may be an hour … people are “too busy to support me in this particurlar moment”.
But I got some strange effect - if I try to call IVR from the location it is buid (my current location - office 1) first I get a message “We have not receive the valid response. Please try again later.” and immediately after that I can hear the IVR welcome and I can use all IVR options.
Any Idea why?
#PitzKey asked me:
Please undo the custom changes, and please post call traces and screenshots of the trunk config from both servers.
So:
Trunk configuration for site 1 (main site)
General
Trunk Name: SIP_Trunk_to _Site2
Hide callerID: No
Otbound callerID: empty
CID Options: Allow Any SID
Maximum channel: empty
Asterisk trunk Dial Options: T
System
Continue if Busy: No
Disable Trunk: No
Monitor Trunk Failures: No
Dialed Number Manipulation Rules
Everytimg is empty including Outbound Dial Prefix
Sip settings
Outgoing
Trunk Name: Trunk_to_Site2
PEER Details: host=192.168.20.1
Username=user1
Secret=password
Type=peer
Fromuser=user1
Incoming
USER Context: user2
USER Details: secret=password
Type=user
Context=from-trunk
Outbound Routes
Route Settings
Route Name: SIP_Route_to Site2
Route CID: empty
Override Extension: No
Route Password: empty
Route Type: Nothing chosen
Music On Hold: default
Time Group: Permanent Route
Route Position: No Change
Trunk Sequence for Matched Routes: SIP_Trunk_to_Site2
Optional destination on Congestion: Normal Congestion
Dial Patterns
[prepend] prefix| [32XX]/caller id]
Import/Export Patterns nothing
Notifications
Email to: empty
Email from: [email protected]
Email Subject: PBX: A call has been placed via outbound route: {{ROUTENAME}}
Email Body: bla bla bla
Additional settings
Call recording: Do not care
PIN set: empty
Ibound routes
Nothing set
===========================================================
Trunk configuration for site 2 (secong site)
General
Trunk Name: SIP_Trunk_to _Site1
Hide callerID: No
Otbound callerID: empty
CID Options: Allow Any SID
Maximum channel: empty
Asterisk trunk Dial Options: T
System
Continue if Busy: No
Disable Trunk: No
Monitor Trunk Failures: No
Dialed Number Manipulation Rules
Everytimg is empty including Outbound Dial Prefix
Sip settings
Outgoing
Trunk Name: Trunk_to_Site1
PEER Details: host=192.168.10.1
Username=user2
Secret=password
Type=peer
Fromuser=user2
Incoming
USER Context: user1
USER Details: secret=password
Type=user
Context=from-trunk
Outbound Routes
Route Settings
Route Name: SIP_Route_to Site1
Route CID: empty
Override Extension: No
Route Password: empty
Route Type: Nothing chosen
Music On Hold: default
Time Group: Permanent Route
Route Position: No Change
Trunk Sequence for Matched Routes: SIP_Trunk_to_Site1
Optional destination on Congestion: Normal Congestion
Dial Patterns
[prepend] prefix| [34XX]/caller id]
Import/Export Patterns nothing
Notifications
Email to: empty
Email from: [email protected]
Email Subject: PBX: A call has been placed via outbound route: {{ROUTENAME}}
Email Body: bla bla bla
Additional settings
Call recording: Do not care
PIN set: empty
Ibound routes
Nothing set
The logs:
And there is the log from Site2 trying to call the IVR on Site1:
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] netsock2.c: Using SIP VIDEO TOS bits 136
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] netsock2.c: Using SIP VIDEO CoS mark 6
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] netsock2.c: Using SIP RTP TOS bits 184
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] netsock2.c: Using SIP RTP CoS mark 5
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] app_stack.c: SIP/Trunk_to_Site1-0000000e Internal Gosub(func-apply-sipheaders,s,1(3)) start
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf(“SIP/Trunk_to_Site1-0000000e”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:2] NoOp(“SIP/Trunk_to_Site1-0000000e”, “Applying SIP Headers to channel SIP/Trunk_to_Site1-0000000e”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:3] Set(“SIP/Trunk_to_Site1-0000000e”, “TECH=SIP”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:4] Set(“SIP/Trunk_to_Site1-0000000e”, “SIPHEADERKEYS=Alert-Info”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:5] While(“SIP/Trunk_to_Site1-0000000e”, “1”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:6] Set(“SIP/Trunk_to_Site1-0000000e”, “sipheader=unset”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:7] ExecIf(“SIP/Trunk_to_Site1-0000000e”, “1?SIPRemoveHeader(Alert-Info:)”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:8] ExecIf(“SIP/Trunk_to_Site1-0000000e”, “0?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:9] ExecIf(“SIP/Trunk_to_Site1-0000000e”, “0?Set(sipheader=http://127.0.0.1;info=unset)”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:10] ExecIf(“SIP/Trunk_to_Site1-0000000e”, “0?Set(sipheader=http://127.0.0.1unset)”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:11] ExecIf(“SIP/Trunk_to_Site1-0000000e”, “0?SIPAddHeader(Alert-Info:unset)”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:12] ExecIf(“SIP/Trunk_to_Site1-0000000e”, “0?Set(PJSIP_HEADER(add,Alert-Info)=unset)”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:13] EndWhile(“SIP/Trunk_to_Site1-0000000e”, “”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:5] While(“SIP/Trunk_to_Site1-0000000e”, “0”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@func-apply-sipheaders:14] Return(“SIP/Trunk_to_Site1-0000000e”, “”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] app_stack.c: Spawn extension (from-trunk-sip-Trunk_to_Site1, 3400, 1) exited non-zero on ‘SIP/Trunk_to_Site1-0000000e’
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] app_stack.c: SIP/Trunk_to_Site1-0000000e Internal Gosub(func-apply-sipheaders,s,1(3)) complete GOSUB_RETVAL=
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] app_dial.c: Called SIP/Trunk_to_Site1/3400
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] app_dial.c: SIP/Trunk_to_Site1-0000000e answered SIP/3222-0000000d
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] app_stack.c: SIP/Trunk_to_Site1-0000000e Internal Gosub(sub-send-obroute-email,s,1(3400,3400,3,1614850334,col. S.Paskalev,3222)) start
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@sub-send-obroute-email:1] GotoIf(“SIP/Trunk_to_Site1-0000000e”, “0?sendEmail”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@sub-send-obroute-email:2] NoOp(“SIP/Trunk_to_Site1-0000000e”, “email notifications disabled…exiting.”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] pbx.c: Executing [s@sub-send-obroute-email:3] Return(“SIP/Trunk_to_Site1-0000000e”, “”) in new stack
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] app_stack.c: Spawn extension (from-trunk-sip-Trunk_to_Site1, , 1) exited non-zero on ‘SIP/Trunk_to_Site1-0000000e’
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] app_stack.c: SIP/Trunk_to_Site1-0000000e Internal Gosub(sub-send-obroute-email,s,1(3400,3400,3,1614850334,col. S.Paskalev,3222)) complete GOSUB_RETVAL=
[2021-03-04 11:32:15] VERBOSE[15290][C-00000007] bridge_channel.c: Channel SIP/Trunk_to_Site1-0000000e joined ‘simple_bridge’ basic-bridge <0692f1c7-49eb-498d-98f6-68c3eac9ba7f>
[2021-03-04 11:32:15] VERBOSE[15264][C-00000007] bridge_channel.c: Channel SIP/3222-0000000d joined ‘simple_bridge’ basic-bridge <0692f1c7-49eb-498d-98f6-68c3eac9ba7f>
[2021-03-04 11:32:26] VERBOSE[15290][C-00000007] bridge_channel.c: Channel SIP/Trunk_to_Site1-0000000e left ‘simple_bridge’ basic-bridge <0692f1c7-49eb-498d-98f6-68c3eac9ba7f>
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] bridge_channel.c: Channel SIP/3222-0000000d left ‘simple_bridge’ basic-bridge <0692f1c7-49eb-498d-98f6-68c3eac9ba7f>
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] app_macro.c: Spawn extension (macro-dialout-trunk, s, 33) exited non-zero on ‘SIP/3222-0000000d’ in macro ‘dialout-trunk’
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Spawn extension (from-internal, 3400, 11) exited non-zero on ‘SIP/3222-0000000d’
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Executing [h@from-internal:1] Macro(“SIP/3222-0000000d”, “hangupcall”) in new stack
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/3222-0000000d”, “1?theend”) in new stack
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/3222-0000000d”, “0?Set(CDR(recordingfile)=)”) in new stack
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Executing [s@macro-hangupcall:4] Hangup(“SIP/3222-0000000d”, “”) in new stack
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/3222-0000000d’ in macro ‘hangupcall’
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/3222-0000000d’
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] app_stack.c: SIP/3222-0000000d Internal Gosub(crm-hangup,s,1) start
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Executing [s@crm-hangup:1] NoOp(“SIP/3222-0000000d”, “Sending Hangup to CRM”) in new stack
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Executing [s@crm-hangup:2] NoOp(“SIP/3222-0000000d”, “HANGUP CAUSE: 16”) in new stack
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Executing [s@crm-hangup:3] ExecIf(“SIP/3222-0000000d”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Executing [s@crm-hangup:4] NoOp(“SIP/3222-0000000d”, “MASTER CHANNEL: 1614850334.13 = 1614850334.13”) in new stack
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Executing [s@crm-hangup:5] GotoIf(“SIP/3222-0000000d”, “0?return”) in new stack
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Executing [s@crm-hangup:6] Set(“SIP/3222-0000000d”, “__CRM_HANGUP=1”) in new stack
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Executing [s@crm-hangup:7] AGI(“SIP/3222-0000000d”, “sangomacrm.agi”) in new stack
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] res_agi.c: <SIP/3222-0000000d>AGI Script sangomacrm.agi completed, returning 0
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] pbx.c: Executing [s@crm-hangup:8] Return(“SIP/3222-0000000d”, “”) in new stack
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/3222-0000000d’
[2021-03-04 11:32:26] VERBOSE[15264][C-00000007] app_stack.c: SIP/3222-0000000d Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
I’ve forgotten to place the log file from Site1.
There it is:
[2021-03-04 11:32:15] VERBOSE[1985][C-00000030] netsock2.c: Using SIP VIDEO TOS bits 136
[2021-03-04 11:32:15] VERBOSE[1985][C-00000030] netsock2.c: Using SIP VIDEO CoS mark 6
[2021-03-04 11:32:15] VERBOSE[1985][C-00000030] netsock2.c: Using SIP RTP TOS bits 184
[2021-03-04 11:32:15] VERBOSE[1985][C-00000030] netsock2.c: Using SIP RTP CoS mark 5
[2021-03-04 11:32:15] VERBOSE[4732][C-00000030] pbx.c: Executing [3400@from-trunk:1] Set(“SIP/user2-00000041”, “__FROM_DID=3400”) in new stack
[2021-03-04 11:32:15] VERBOSE[4732][C-00000030] pbx.c: Executing [3400@from-trunk:2] NoOp(“SIP/user2-00000041”, “Received an unknown call with DID set to 3400”) in new stack
[2021-03-04 11:32:15] VERBOSE[4732][C-00000030] pbx.c: Executing [3400@from-trunk:3] Goto(“SIP/user2-00000041”, “s,a2”) in new stack
[2021-03-04 11:32:15] VERBOSE[4732][C-00000030] pbx_builtins.c: Goto (from-trunk,s,2)
[2021-03-04 11:32:15] VERBOSE[4732][C-00000030] pbx.c: Executing [s@from-trunk:2] Answer(“SIP/user2-00000041”, “”) in new stack
[2021-03-04 11:32:15] VERBOSE[4732][C-00000030] pbx.c: Executing [s@from-trunk:3] Log(“SIP/user2-00000041”, “WARNING,Friendly Scanner from 10.150.33.2”) in new stack
[2021-03-04 11:32:15] WARNING[4732][C-00000030] Ext. s: Friendly Scanner from 10.150.33.2
[2021-03-04 11:32:15] VERBOSE[4732][C-00000030] pbx.c: Executing [s@from-trunk:4] Wait(“SIP/user2-00000041”, “2”) in new stack
[2021-03-04 11:32:17] VERBOSE[4732][C-00000030] pbx.c: Executing [s@from-trunk:5] Playback(“SIP/user2-00000041”, “ss-noservice”) in new stack
[2021-03-04 11:32:17] VERBOSE[4732][C-00000030] file.c: <SIP/user2-00000041> Playing ‘ss-noservice.ulaw’ (language ‘en’)
[2021-03-04 11:32:22] VERBOSE[4732][C-00000030] pbx.c: Executing [s@from-trunk:6] SayAlpha(“SIP/user2-00000041”, “3400”) in new stack
[2021-03-04 11:32:22] VERBOSE[4732][C-00000030] file.c: <SIP/user2-00000041> Playing ‘digits/3.ulaw’ (language ‘en’)
[2021-03-04 11:32:23] VERBOSE[4732][C-00000030] file.c: <SIP/user2-00000041> Playing ‘digits/4.ulaw’ (language ‘en’)
[2021-03-04 11:32:24] VERBOSE[4732][C-00000030] file.c: <SIP/user2-00000041> Playing ‘digits/0.ulaw’ (language ‘en’)
[2021-03-04 11:32:25] VERBOSE[4732][C-00000030] file.c: <SIP/user2-00000041> Playing ‘digits/0.ulaw’ (language ‘en’)
[2021-03-04 11:32:26] VERBOSE[4732][C-00000030] pbx.c: Executing [s@from-trunk:7] Hangup(“SIP/user2-00000041”, “”) in new stack
[2021-03-04 11:32:26] VERBOSE[4732][C-00000030] pbx.c: Spawn extension (from-trunk, s, 7) exited non-zero on ‘SIP/user2-00000041’
[2021-03-04 11:32:26] VERBOSE[4732][C-00000030] pbx.c: Executing [h@from-trunk:1] Macro(“SIP/user2-00000041”, “hangupcall,”) in new stack
[2021-03-04 11:32:26] VERBOSE[4732][C-00000030] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/user2-00000041”, “1?theend”) in new stack
[2021-03-04 11:32:26] VERBOSE[4732][C-00000030] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2021-03-04 11:32:26] VERBOSE[4732][C-00000030] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/user2-00000041”, “0?Set(CDR(recordingfile)=)”) in new stack
[2021-03-04 11:32:26] VERBOSE[4732][C-00000030] pbx.c: Executing [s@macro-hangupcall:4] Hangup(“SIP/user2-00000041”, “”) in new stack
[2021-03-04 11:32:26] VERBOSE[4732][C-00000030] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/user2-00000041’ in macro ‘hangupcall’
[2021-03-04 11:32:26] VERBOSE[4732][C-00000030] pbx.c: Spawn extension (from-trunk, h, 1) exited non-zero on ‘SIP/user2-00000041’
[2021-03-04 11:37:02] VERBOSE[1911] asterisk.c: Remote UNIX connection
[2021-03-04 11:37:02] VERBOSE[5117] asterisk.c: Remote UNIX connection disconnected
[2021-03-04 11:37:02] VERBOSE[1911] asterisk.c: Remote UNIX connection
[2021-03-04 11:37:02] VERBOSE[5119] asterisk.c: Remote UNIX connection disconnected
And may be I have to expain that the extension 3222 from Site 2 is trying to call extension 3400 on Site1. The 3400 is a Misc application on Site1 with the next settings:
Enable: Yes
Description: reach IVR by extension
Feature Code: 3400
Destination: IVR
Welcome IVR
The voice responce (the one site2 hear) is something like - “The number is not in service”