IVR Extension Issue

I have a working (for the most part) FreePBX system. DID to Extension works fine, VM, etc… however Im having an issue when using a IVR. Recording plays, but when I select/press a number it does not take me to the extension I set up. I know the extension works as when I do a direct DID to extension no issues. The problem, MUST be in the IVR at least that’s my assumption as I can’t seem to find it.

I’m using a Grandstream 1405 phone, and its working, VM notification, etc… and as I said the DID to extension works great also.

Can someone help me, seems that the IVR setup should be somewhat simple… but I have been wrong before! :slight_smile:

Thanks much!

brian

Added: relaxdtmf=yes to the sip.conf file and rebooted the server. All key presses are now picked up. Just in case someone is have the same issue…
-b

Added: relaxdtmf=yes to the sip.conf file and rebooted the server. All key presses are now picked up. Just in case someone is have the same issue…
-b

Can anyone help? Im kinda stuck and need to get this into a testing phase for the customer.
Thanks!

You provided no information on your system, versions, how it was installed (by hand or distro, if distro which one), trunk type etc.

I suspect you have misconfigured your DTMF. It’s not the IVR.

Sorry, you are correct… I seem to forget that’s important! :slight_smile:

System Info (please let me know if you need something else)

Install from Distro (ISO) using stable 64bit 3.211.63-10 with Asterisk 11
The trunk is from Sipstation, all ports are forwarded as recommended
Its running on a Atom proc, 4GB RAM

As for the DTMF, I have not changed anything, so if there is a setting that NEEDS to be changed that may very well be it.

THANK YOU for the reply, MUCH appreciated!

-b

Are you dialing the IVR internally or externally?

Phone connected is a GRandstream 2 line GPX1405, sorry forgot that…

If logs help please let me know!

thanks again!

Externally, it just seems to NOT “notice” that I have pressed a key… :slight_smile:

As for the DTMF, on the Extension is was set to the RFC2833, Ic an change it to Auto, but other posts seem to steer AWAY from that.

Sorry, I admit, n00b! :slight_smile:

Log output attached…

/START LOG
[2013-06-10 10:47:15] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:1] NoOp(“SIP/fpbx-1-351a933a-00000034”, “Catch-All DID Match - Found 9166256568 - You probably want a DID for this.”) in new stack
[2013-06-10 10:47:15] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:2] Goto(“SIP/fpbx-1-351a933a-00000034”, “ext-did,s,1”) in new stack
[2013-06-10 10:47:15] VERBOSE[14752][C-0000002b] pbx.c: – Goto (ext-did,s,1)
[2013-06-10 10:47:15] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:1] ExecIf(“SIP/fpbx-1-351a933a-00000034”, “1?Set(__FROM_DID=s)”) in new stack
[2013-06-10 10:47:15] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:2] Gosub(“SIP/fpbx-1-351a933a-00000034”, “app-blacklist-check,s,1()”) in new stack
[2013-06-10 10:47:15] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/fpbx-1-351a933a-00000034”, “0?blacklisted”) in new stack
[2013-06-10 10:47:15] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:2] Set(“SIP/fpbx-1-351a933a-00000034”, “CALLED_BLACKLIST=1”) in new stack
[2013-06-10 10:47:15] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:3] Return(“SIP/fpbx-1-351a933a-00000034”, “”) in new stack
[2013-06-10 10:47:15] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:3] Set(“SIP/fpbx-1-351a933a-00000034”, “CDR(did)=s”) in new stack
[2013-06-10 10:47:15] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:4] Gosub(“SIP/fpbx-1-351a933a-00000034”, “cidlookup,cidlookup_2,1()”) in new stack
[2013-06-10 10:47:15] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:1] Set(“SIP/fpbx-1-351a933a-00000034”, “CURLOPT(httptimeout)=7”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:2] Set(“SIP/fpbx-1-351a933a-00000034”, “CALLERID(name)=”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:3] Set(“SIP/fpbx-1-351a933a-00000034”, “current_hour=2013-06-10 10”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:4] Set(“SIP/fpbx-1-351a933a-00000034”, “last_query_hour=2013-06-10 10”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:5] Set(“SIP/fpbx-1-351a933a-00000034”, “total_hourly_queries=2”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:6] ExecIf(“SIP/fpbx-1-351a933a-00000034”, “0?Set(DB(cidlookup/opencnam_total_hourly_queries)=0)”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:7] ExecIf(“SIP/fpbx-1-351a933a-00000034”, “0?Set(DB(cidlookup/opencnam_total_hourly_queries)=0)”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:8] Set(“SIP/fpbx-1-351a933a-00000034”, “DB(cidlookup/opencnam_total_hourly_queries)=3”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:9] ExecIf(“SIP/fpbx-1-351a933a-00000034”, “0?System(/var/lib/asterisk/bin/opencnam-alert.php)”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:10] Set(“SIP/fpbx-1-351a933a-00000034”, “DB(cidlookup/opencnam_last_query_hour)=2013-06-10 10”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:11] Return(“SIP/fpbx-1-351a933a-00000034”, “”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:5] ExecIf(“SIP/fpbx-1-351a933a-00000034”, “1 ?Set(CALLERID(name)=19162517761)”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:6] Set(“SIP/fpbx-1-351a933a-00000034”, “__CALLINGPRES_SV=prohib_passed_screen”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:7] Set(“SIP/fpbx-1-351a933a-00000034”, “CALLERPRES()=allowed_not_screened”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:8] Goto(“SIP/fpbx-1-351a933a-00000034”, “ivr-1,s,1”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Goto (ivr-1,s,1)
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:1] Set(“SIP/fpbx-1-351a933a-00000034”, “TIMEOUT_LOOPCOUNT=0”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:2] Set(“SIP/fpbx-1-351a933a-00000034”, “INVALID_LOOPCOUNT=0”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:3] Set(“SIP/fpbx-1-351a933a-00000034”, “_IVR_CONTEXT_ivr-1=”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:4] Set(“SIP/fpbx-1-351a933a-00000034”, “_IVR_CONTEXT=ivr-1”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:5] Set(“SIP/fpbx-1-351a933a-00000034”, “__IVR_RETVM=RETURN”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:6] GotoIf(“SIP/fpbx-1-351a933a-00000034”, “0?skip”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:7] Answer(“SIP/fpbx-1-351a933a-00000034”, “”) in new stack
[2013-06-10 10:47:16] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:8] Wait(“SIP/fpbx-1-351a933a-00000034”, “1”) in new stack
[2013-06-10 10:47:17] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:9] Set(“SIP/fpbx-1-351a933a-00000034”, “IVR_MSG=custom/arenaivr”) in new stack
[2013-06-10 10:47:17] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:10] Set(“SIP/fpbx-1-351a933a-00000034”, “TIMEOUT(digit)=3”) in new stack
[2013-06-10 10:47:17] VERBOSE[14752][C-0000002b] func_timeout.c: – Digit timeout set to 3.000
[2013-06-10 10:47:17] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:11] ExecIf(“SIP/fpbx-1-351a933a-00000034”, “1?Background(custom/arenaivr)”) in new stack
[2013-06-10 10:47:17] VERBOSE[14752][C-0000002b] file.c: – <SIP/fpbx-1-351a933a-00000034> Playing ‘custom/arenaivr.slin’ (language ‘en’)
[2013-06-10 10:47:45] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:12] WaitExten(“SIP/fpbx-1-351a933a-00000034”, “10,”) in new stack
[2013-06-10 10:47:47] NOTICE[1786] chan_sip.c: Disconnecting call ‘SIP/fpbx-1-351a933a-00000034’ for lack of RTP activity in 31 seconds
[2013-06-10 10:47:47] VERBOSE[14752][C-0000002b] pbx.c: == Spawn extension (ivr-1, s, 12) exited non-zero on ‘SIP/fpbx-1-351a933a-00000034’
[2013-06-10 10:47:47] VERBOSE[14752][C-0000002b] pbx.c: – Executing [[email protected]:1] Hangup(“SIP/fpbx-1-351a933a-00000034”, “”) in new stack
[2013-06-10 10:47:47] VERBOSE[14752][C-0000002b] pbx.c: == Spawn extension (ivr-1, h, 1) exited non-zero on ‘SIP/fpbx-1-351a933a-00000034’

END LOG\

Try enabling Signal RINGING on the inbound route to that IVR.

Ok enabled but no change… same thing, just kind of ignores me… starting to think its my wife on the other end… :slight_smile:

I also set it to go to the extension on a timeout… but that is not working, it just hangs up… hmmm, interesting…

DTMF setting on the Grandstream gateway MUST match trunk in FreePBX.

Setting for the trunk:
disallow=all
allow=ulaw&g729
context=from-pstn
type=peer
insecure=very
qualify=yes
sendrpid=yes
trustrpid=yes
dtmfmode=rfc2833
GrandSteam has the Send DTMF “via RTP RFC2833” checked on both lines.
There IS a DTMF Payload Type setting currently at 101.
Thanks VERY much for all the replies!!!

Ok so been changing settings, moved to v10 and v1.8. IVR plays, and now it even responds and says “you have not selected…” but the IVR pass to the extensions still does not work… Phones are set to the SAME RFC, etc…

So at this point I need to look at alternatives, as the customer REALLY wants to look at going to a PBX system in house. Maybe the phone, who knows… but I have tried a Polycom phone also, along with X-Lite soft phone same thing…

Im sure its something Im doing or not doing, but it also seems like this SHOULD be easy enough. So are there alternatives, i.e. does say PBX In a Box have a different set, seems all Asterisk, so…? I would even pay support however given that I have been messing with it for two+ days, Im not sure an hour of support would sort it out…

THANKS to ALL the replies, it IS appreciated!

if yes, have you enabled extension dialing in the ivr.

Hmmm, is that a separate setting somewhere… ? I have the 1 --> 101, 2 --> 102 settings… unless Im passing over something.

On a side note… SIPCAT is NOT even CLOSE to FreePBX…