Pause and dial digits after SIP call connected

I need to call a third-party PBX and provide an extension number to the auto attendant. Searches have indicated that a Custom Destination will handle it. My (non-working) solution is based on an old post by Lorne Gaetz. I have created a feature code that finds its way to a custom destination using the following in /etc/asterisk/extensions_custom.conf:

[custom-dial-test]
exten => s,1,Dial(SIP/enventis/6515551212,60,D(wwwwwwwwww123))

Dialing the feature code results in this:

[2017-06-16 08:05:45] VERBOSE[20154] pbx.c: – Goto (custom-dial-test,s,1)
[2017-06-16 08:05:45] VERBOSE[20154] pbx.c: – Executing [s@custom-dial-test:1] Dial(“SIP/1074-00000060”, “SIP/enventis/6515551212,300,D(wwwwwwwwww123)”) in new stack
[2017-06-16 08:05:45] VERBOSE[20154] netsock2.c: == Using SIP RTP TOS bits 184
[2017-06-16 08:05:45] VERBOSE[20154] netsock2.c: == Using SIP RTP CoS mark 5
[2017-06-16 08:05:45] VERBOSE[20154] app_dial.c: – Called SIP/enventis/6515551212
[2017-06-16 08:05:45] VERBOSE[20154] app_dial.c: – SIP/enventis-00000061 is making progress passing it to SIP/1074-00000060
[2017-06-16 08:06:45] VERBOSE[20154] app_dial.c: – No one is available to answer at this time (1:0/0/0)
[2017-06-16 08:06:45] VERBOSE[20154] pbx.c: – Auto fallthrough, channel ‘SIP/1074-00000060’ status is ‘NOANSWER’

The phone number, of course, is a valid one–not 5551212. After a minute of busy tone, the call is terminated.

When I simply dial 6515551212 at the extension, the call completes OK.

[2017-06-16 08:12:08] VERBOSE[20179] pbx.c: – Executing [6512764838@from-internal:1] Macro(“SIP/1074-0000006d”, “user-callerid,LIMIT
,”) in new stack
[2017-06-16 08:12:08] VERBOSE[20179] pbx.c: – Executing [s@macro-user-callerid:1] Set(“SIP/1074-0000006d”, “AMPUSER=1074”) in new st
ack
[2017-06-16 08:12:08] VERBOSE[20179] pbx.c: – Executing [s@macro-user-callerid:2] GotoIf(“SIP/1074-0000006d”, “0?report”) in new sta
ck
[2017-06-16 08:12:08] VERBOSE[20179] pbx.c: – Executing [s@macro-user-callerid:3] ExecIf(“SIP/1074-0000006d”, “1?Set(REALCALLERIDNUM
=1074)”) in new stack

[2017-06-16 08:12:08] VERBOSE[20179] pbx.c: – Executing [s@macro-dialout-trunk:22] Dial(“SIP/1074-0000006d”, “SIP/enventis/6515551212,300,”) in new stack
[2017-06-16 08:12:08] VERBOSE[20179] netsock2.c: == Using SIP RTP TOS bits 184
[2017-06-16 08:12:08] VERBOSE[20179] netsock2.c: == Using SIP RTP CoS mark 5
[2017-06-16 08:12:08] VERBOSE[20179] app_dial.c: – Called SIP/enventis/6515551212
[2017-06-16 08:12:11] VERBOSE[20179] app_dial.c: – SIP/enventis-0000006e is making progress passing it to SIP/1074-0000006d
[2017-06-16 08:12:14] VERBOSE[20179] app_dial.c: – SIP/enventis-0000006e is making progress passing it to SIP/1074-0000006d
[2017-06-16 08:12:19] VERBOSE[20179] app_dial.c: – SIP/enventis-0000006e answered SIP/1074-0000006d

There’s a lot missing at the ellipses—I’m assuming that people familiar with this stuff know what must have been in there. I’m happy to provide the full log! (Right. This stuff does not happen when using the Custom Destination.)

Both the good and the bad seem to be doing the same thing, with different results:
[2017-06-16 08:05:45] VERBOSE[20154] app_dial.c: – Called SIP/enventis/6515551212

I’ve also tried simplifying this by omitting the “D(wwwwwwwwww123)” with the same result.

I’d really appreciate some suggestions!

–Don

Try modifying dialplan to:

[custom-dial-test]
exten => s,1,Dial(local/6515551212@from-internal,60,D(wwwwwwwwww123))

I was pretty sure that’s one of the things I had tried, but there must be a little difference. It appears to be working! (The DTMF sounds a little strange on my cell phone I’m using for testing. I’ll try the real path and see what happens.)

Thanks,

–Don