IVR not transferring call to extension

FreePBX Applications / Modules
1

hi after turning on queue logs and adding a queue for some weird reason IVR does not work. logs shows as follow.

[2021-03-16 23:39:16] NOTICE[1909] chan_sip.c: Disconnecting call ‘SIP/xxxxxx-00000011’ for lack of RTP activity in 31 seconds
20211 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] pbx.c: Spawn extension (ivr-1, s, 11) exited non-zero on ‘SIP/xxxxxx-00000011’
20212 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] pbx.c: Executing [h@ivr-1:1] Hangup(“SIP/xxxxxx-00000011”, “”) in new stack
20213 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] pbx.c: Spawn extension (ivr-1, h, 1) exited non-zero on ‘SIP/xxxxxx-00000011’
20214 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] app_stack.c: SIP/xxxxxx-00000011 Internal Gosub(crm-hangup,s,1) start
20215 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] pbx.c: Executing [s@crm-hangup:1] NoOp(“SIP/xxxxxx-00000011”, “Sending Hangup to CRM”) in new stack
20216 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] pbx.c: Executing [s@crm-hangup:2] NoOp(“SIP/xxxxxx-00000011”, “HANGUP CAUSE: 44”) in new stack
20217 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] pbx.c: Executing [s@crm-hangup:3] ExecIf(“SIP/xxxxxx-00000011”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
20218 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] pbx.c: Executing [s@crm-hangup:4] NoOp(“SIP/xxxxxx-00000011”, “MASTER CHANNEL: 1615937925.17 = 1615937925.17”) in new stack
20219 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] pbx.c: Executing [s@crm-hangup:5] GotoIf(“SIP/xxxxxx-00000011”, “0?return”) in new stack
20220 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] pbx.c: Executing [s@crm-hangup:6] Set(“SIP/xxxxxx-00000011”, “__CRM_HANGUP=1”) in new stack
20221 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] pbx.c: Executing [s@crm-hangup:7] AGI(“SIP/xxxxxx-00000011”, “sangomacrm.agi”) in new stack
20222 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
20223 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] res_agi.c: <SIP/xxxxxx-00000011>AGI Script sangomacrm.agi completed, returning 0
20224 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] pbx.c: Executing [s@crm-hangup:8] Return(“SIP/xxxxxx-00000011”, “”) in new stack
20225 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] app_stack.c: Spawn extension (ivr-1, h, 1) exited non-zero on ‘SIP/xxxxxx-00000011’
20226 [2021-03-16 23:39:16] VERBOSE[25325][C-0000000c] app_stack.c: SIP/xxxxxx-00000011 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
20227 [2021-03-16 23:39:16] VERBOSE[25327][C-0000000c] app_mixmonitor.c: MixMonitor close filestream (mixed)
20228 [2021-03-16 23:39:16] VERBOSE[25327][C-0000000c] app_mixmonitor.c: End MixMonitor Recording SIP/xxxxxx-00000011

2

There are no log lines relevant to an IVR in what you provided.

Pls share log lines with a link to pastebin:
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs-PartII

3

what type of log should i be looking for?

4

Click the provided link (above) and follow the instructions.

5

here is the log on pastebin

https://pastebin.freepbx.org/view/7a22a2f4

6

No DTMF received, possible trunk misconfig or some change with the provider. Whatever is going on, it has nothing to do with any recent queue changes.

7

thank you

8

what possible dtmf error i should be looking into? i have a test call from the other day that shows non-codec capabilities

21431 [2021-03-17 00:10:08] VERBOSE[1909] chan_sip.c: — (20 headers 20 lines) —
21432 [2021-03-17 00:10:08] VERBOSE[1909] chan_sip.c: Sending to XXXXXXXX:x (NAT)
21433 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Sending to XXXXXX:x (NAT)
21434 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Using INVITE request as basis request -x@x
21435 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found peer ‘x’ for ‘+anonymous’ from x.x.x.x
21436 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] netsock2.c: Using SIP RTP TOS bits 184
21437 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] netsock2.c: Using SIP RTP CoS mark 5
21438 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found RTP audio format 8
21439 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found RTP audio format 0
21440 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found RTP audio format 18
21441 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found RTP audio format 110
21442 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found RTP audio format 117
21443 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found RTP audio format 119
21444 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found RTP audio format 3
21445 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found RTP audio format 101
21446 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found audio description format PCMA for ID 8
21447 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found audio description format PCMU for ID 0
21448 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found audio description format G729 for ID 18
21449 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found audio description format speex for ID 110
21450 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found audio description format speex for ID 117
21451 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found audio description format speex for ID 119
21452 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found audio description format GSM for ID 3
21453 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Found audio description format telephone-event for ID 101
21454 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|gsm|alaw|g729|speex|speex16|speex32)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
21455 [2021-03-17 00:10:08] VERBOSE[1909][C-0000000e] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

9

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