Incoming Callers not redirected to Digital Receptionist


#1

Hi-

I’m a newbie to Asterisk, and in the process of setting up a working system. I’m kind of stuck with a problem regarding the Digital Receptionist, and I was hoping someone on this list might be able to shed some light on whats going on.

So basically, I have the SIP phones/extensions and outbound trunks configured (I’m using Telasip trunks for outbound), and I am able to make and receive calls from my SIP phones to external cell phones. I have a ‘front desk’ extension configured to be 1998, and I have another two phones with extension 1901 and 1902 for my end users.

I have a main DiD number through Telasip (lets call it 408-123-4567), and when that number is called, I want the following to happen:

  1. Phone rings three times at extension 1998 (front desk)
  2. If no one answers the phone, then the digital receptionist takes over and presents the caller with a menu (“dial *411 for the company directory”)
  3. If the caller does not enter any extension, then the call goes to voicemail for extension 1998

This seems like a pretty straightforward setup, and I have seen many examples of this on the 'net, but unfortunately none of them work for me. Regardless of what I do, an external caller who dials my DiD (408-123-4567) goes through step 1, then straight to step 3 - completely skipping step 2. In other words, the digital receptionist is never called upon to present the menu.

I have gone through the basic sanity checks:
A. When I dial 7777 from an internal extension (1901), I get the digital receptionist and am presented with the option to dial the company directory.
B. When I go to Amp -> Setup -> Incoming Calls and select ‘Extension 1901’ (or any extension, incl. 1998) instead of ‘Digital Receptionist’, callers from outside to the DiD are sent directly to the extension, and once again, steps 1 and 3 are executed in succession.

It is almost as if I’m missing some line in extensions-custom.conf that tells Asterisk to invoke the digital receptionist, (possibly between lines 5 and 6? I’m just guessing here…) Here is what the extensions_custom.conf looks like:

[tsvxsj-in]
exten => 4081234567,1,Answer
exten => 4081234567,2,Wait(1)
exten => 4081234567,3,Background(pls-hold-while-try)
exten => 4081234567,4,NoOp(Incoming call on TelaSIP #4081234567)
exten => 4081234567,5,Dial(SIP/1998,20,m)
exten => 4081234567,6,Voicemail(1998@default)
exten => 4081234567,7,Hangup

And here’s what the debug log looks like when a call comes in from the outside, and Asterisk is set to send calls to the Digital Receptionist:

asteriskCLI>
– Executing Answer(“SIP/telasip-username-e3f2”, “”) in new stack
– Executing Wait(“SIP/telasip-username-e3f2”, “1”) in new stack
– Executing BackGround(“SIP/telasip-username-e3f2”, “pls-hold-while-try”) in new stack
– Playing ‘pls-hold-while-try’ (language ‘en’)
– Executing NoOp(“SIP/telasip-username-e3f2”, “Incoming call for ArrayComm on TelaSIP #4081234567”) in new stack
– Executing Dial(“SIP/telasip-username-e3f2”, “SIP/1998|20|m”) in new stack
– Called 1998
– Started music on hold, class ‘default’, on channel ‘SIP/telasip-username-e3f2’
– SIP/1998-f4fa is ringing
– Nobody picked up in 20000 ms
– Stopped music on hold on SIP/telasip-username-e3f2
– Executing VoiceMail(“SIP/telasip-username-e3f2”, “1998@default”) in new stack
– Playing ‘vm-intro’ (language ‘en’)
– Playing ‘beep’ (language ‘en’)
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/default/1998/INBOX/msg0001 format: wav49, 0x99fbe40
– x=1, open writing: /var/spool/asterisk/voicemail/default/1998/INBOX/msg0001 format: wav, 0x9a002e0
– User hung up
– Recording was 2 seconds long but needs to be at least 3 - abandoning
== Spawn extension (tsvxsj-in, 4081234567, 6) exited non-zero on 'SIP/telasip-username-e3f2’
asterisk
CLI>

This seems like a pretty simple problem, and I’ve tried googling variants of ‘Asterisk Digital Receptionist Now Working’ and ‘Asterisk Digital Receptionist Problem’ with no results. I’m turning to you guys in the hope that someone will be able to tell me what I’m doing wrong. If there’s anything else (configs, debug logs) that I need to post, just let me know and I’ll do that as well.

Thanks in advance-

–Maxx


#2

First think you need to keep in mind in the US a phone will ring 10 times in 1 minute (about every 6 seconds). Next is most answering machines will answer after 4 rings. But it is up to you as to how many rings you will have before it gets passed on.

Next you need to really read about Dial() and the timeout values and exit status. Fallow with Directory(), Voicemail().

You might also decide you don’t want people calling into the PBX know what the extension of the voicemail box they are being forward to. If that is the case you might want to add Playback() before the VoiceMail() and look at the options you pass to VoiceMail().


#3

If you have extensions_custom.conf, then you’re most likely using AMP or freePBX – you shouldn’t be configuring this manually. Its pretty easy to setup via the GUI:

Create a Ring Group with a single member (1998)
The Destination if No Answer is set to your Digital Receptionist
Add a “Leave Voicemail” option to the DR.

PROFIT! :smile: