IVR Question

dicko · February 27, 2021, 11:03pm

In

/etc/asterisk/extensions_custom.conf

hmaznev · March 1, 2021, 8:51am

Thank you #dicko. I will try and will tell you the result.
BTW if the attempt is not successful - I will open a new topic. Somebody could have the same problem with using IVR in corporate intranet with more than one FreePBX boxes.
Thanks again.

PitzKey · March 1, 2021, 9:46am

Don’t. Please don’t create duplicate topics.
As mentioned, post screenshots and logs.

You can post a call trace via pastebin:
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs-PartII

hmaznev · March 1, 2021, 12:26pm

So … finaly what to do/ To create a new topic or to continue here?

lgaetz · March 1, 2021, 12:27pm

Continue here, the posts were already split into a new topic.

hmaznev · March 1, 2021, 12:44pm

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?
which log or what to search for in the general log?

hmaznev · March 1, 2021, 12:50pm

BTW after i made changes I reload the FreePBX - fwconsole reload

PitzKey · March 1, 2021, 5:43pm

Please undo the custom changes, and please post call traces and screenshots of the trunk config from both servers

hmaznev · March 4, 2021, 9:23am

Sorry for the delay, we have had a national holiday.
First at all the trunk configuration from both sites:
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

hmaznev · March 4, 2021, 9:41am

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=

hmaznev · March 4, 2021, 9:42am

The voice responce (the one site2 hear) is something like - “he line is connected. The number is not in service”

hmaznev · March 4, 2021, 10:04am

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

hmaznev · March 4, 2021, 11:45am

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

hmaznev · March 5, 2021, 3:56pm

Sorry Igaetz, but here no one answer to my question. I will open a new topic with a hope … somebody to read and anwer to it

Stewart1 · March 5, 2021, 4:01pm

On Site 1, create an Inbound Route with DID Number set to 3400 and CallerID Number left blank. Set the Destination to the desired IVR. Test.

lgaetz · March 5, 2021, 4:06pm

I have already split all these posts off into a new topic. Don’t create a new one for this one issue.

edit: /sigh

PitzKey · March 7, 2021, 8:56pm

@hmaznev You were told NOT to create a new topic, but you went ahead and ignored us.
I asked you to share the logs via pastebin link and post screenshots of your settings. It is very hard to read your hundreds of lines of config settings and logs on the forums.

We want to help you, please listen to us. Please.

hmaznev · March 8, 2021, 7:42am

Thank you very much #Stewart1!!! This solved my problem.
I suspect that it is relatet to the inbound routes … but was not confident (lack of expirience with FreePBX).

hmaznev · March 8, 2021, 7:50am

Sorry #PitzKey, please excuse my ignorance.
I will follow your advice since now and forever.
I have never used tha pastebin so far … but I will in every new case … thanks for remind me and tough me something new.
The reason(s) I ignore your command not to create a new topic was 1. Nobody answer me long time and of course I was thinking that nobody read an old topics, and 2. Frankly - I every time was searching for this exact topic long time. If I create a new one - it appears in my dashboard and access to it is pretty fast.
Please ones again - excuse my ignorance. I promise - I will follow the rules.

system · June 3, 2021, 11:06pm

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