Asterisk Phonebook Pause

General Help
1

I’m trying to setup some speed dials in the *Phonebook, and can’t find a way to enter a pause. For instance I want to call a 800 number, and then dial an extension. An example would be 8009999999w1111, 8009999999#w1111 and multiple variations I’ve found online. Everything I’ve tried gives me an error ‘Please enter a valid Number’.

2

You can’t do this via the FreePBX web interface. You have to do a ‘database put’ from the Asterisk CLI.

3

Sorry, I did not know the syntax by heart. Should have told you to do a database show so you could see the keys.

Anyway please send a debug trace when the call fails.

4

Ok so I added the number with the database put(not too much info out there on how to) but now I get ‘cannot be completed as dialed’. Do I have to change the dial plan or something?

5

Sorry I’m such a noob, but by debug trace do you mean this?

== Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [789****#w107@from-internal:1] ResetCDR("SIP/107-000004cf", "") in new stack -- Executing [789****#w107@from-internal:2] NoCDR("SIP/107-000004cf", "") in new stack -- Executing [789****#w107@from-internal:3] Progress("SIP/107-000004cf", "") in new stack -- Executing [789****#w107@from-internal:4] Wait("SIP/107-000004cf", "1") in new stack -- Executing [789****#w107@from-internal:5] Progress("SIP/107-000004cf", "") in new stack -- Executing [789****#w107@from-internal:6] Playback("SIP/107-000004cf", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack -- <SIP/107-000004cf> Playing 'silence/1.gsm' (language 'en') -- <SIP/107-000004cf> Playing 'cannot-complete-as-dialed.gsm' (language 'en') -- <SIP/107-000004cf> Playing 'check-number-dial-again.gsm' (language 'en') -- Executing [789****#w107@from-internal:7] Wait("SIP/107-000004cf", "1") in new stack -- Executing [789****#w107@from-internal:8] Congestion("SIP/107-000004cf", "20") in new stack [2011-08-19 14:43:10] WARNING[32757]: channel.c:4622 ast_prod: Prodding channel 'SIP/107-000004cf' failed == Spawn extension (from-internal, 789****#w107, 8) exited non-zero on 'SIP/107-000004cf' -- Executing [h@from-internal:1] Hangup("SIP/107-000004cf", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/107-000004cf'

I’ve also tried it without the #. I’m pretty sure it has to do with my outbound route, but I have no idea how to enter the pause. Oh and thanks for your help!!

6

W’s only work for DAHDI trunks.

SIP sends everything in an invite message, no concept of in band audio.

7

Oh, so there’s no way to enter a speed dial with a pause then? Well that stinks!

8

There is no way to do it with SIP trunks. It’s not a FreePBX issue.

You could write some custom code that would dial the channel then use the send dtmf application to transmit in channel DTMF.

9