Dialplan Dial() help

[Slovniek]

Hello, I want to call from my SoftPhone (Zoiper) to my cell phone with the help of dialplan.
Here’s my extensions.conf

[outbound-voipms]
exten => _.!,1,NoOp(Entering user defined context from-internal-custom in extensions_custom.conf)
exten => _.!,2,Dial(Local/13034985509@voipms,,D(wwwwwwww${EXTEN}))
exten => _.!,3,Hangup()

my goal it to dial this number → 1 303 498 5509 or if it’s possible, the one I dialed in Zoiper

my sip.conf

[general]
register => username665:password001@amsterdam1.voip.ms:5060

[voipms]
canreinvite=no
context=outbound-voipms
host=amsterdam1.voip.ms
host=dynamic
secret=password001
type=friend
username=username665
disallow=all
allow=ulaw
fromuser=username665
trustrpid=yes
sendrpid=yes

Changed the username and password for obvious reasons.

When I dial the number in Zoiper, I get these messages (in CLI)

Packet timed out after 32000ms with no response
  == Using SIP RTP CoS mark 5
    -- Executing [13034985509@outbound-voipms:1] NoOp("SIP/voipms-0000000b", "Entering user defined context from-internal-custom in extensions_custom.conf") in new stack
    -- Executing [13034985509@outbound-voipms:2] Dial("SIP/voipms-0000000b", "Local/13034985509@voipms,,D(wwwwwwww13034985509)") in new stack
[May 27 14:44:43] NOTICE[2620][C-000003d6]: core_local.c:737 local_call: No such extension/context 13034985509@voipms while calling Local channel
    -- Couldn't call Local/13034985509@voipms
  == Everyone is busy/congested at this time (0:0/0/0)
    -- Executing [13034985509@outbound-voipms:3] Hangup("SIP/voipms-0000000b", "") in new stack
  == Spawn extension (outbound-voipms, 13034985509, 3) exited non-zero on 'SIP/voipms-0000000b'
    -- Executing [h@outbound-voipms:1] NoOp("SIP/voipms-0000000b", "Entering user defined context from-internal-custom in extensions_custom.conf") in new stack
    -- Executing [h@outbound-voipms:2] Dial("SIP/voipms-0000000b", "Local/13034985509@voipms,,D(wwwwwwwwh)") in new stack
    -- Caller hung up before dial.
  == Spawn extension (outbound-voipms, h, 2) exited non-zero on 'SIP/voipms-0000000b'
  == Using SIP RTP CoS mark 5

is there anything I’m missing? Could anyone point me in the right direction?
Thank you in advance

[tootai]

Hello. You have to use SIP, not Local as voipms is your provider

[cige]

Hello;
You should Chek the context you are using when making outbound calls.
Use your context to make outgoing calls.

[Slovniek]

That seemed to do something. However I have a new issue now.
When I dial the number I’m trying to call, I instantly get a new incoming call from voipms, which when I answer, starts playing music on both of these calls

[david551]

You should not be using chan_sip on new systems, it will not exist in the version of Asterisk released later this year.

Using _. is not a good idea, as it matches h, and other special extensions.

@Slovniek

host=amsterdam1.voip.ms
host=dynamic

You can’t have both.

@Slovniek

username=username665

This is not useful, on its own, and in all cases with an explicit host address.

@Slovniek

1 303 498 5509

What does this represent? Why do you need a third party service with explicit DTMF? What is wrong with dialling the cell number directly.

@Slovniek

When I dial the number I’m trying to call, I instantly get a new incoming call from voipms, which when I answer, starts playing music on both of these calls

Is it your own number? You appear to be calling yourself, and to listen to both calls on the phone, one has to be on hold, so generating music on hold.

I get the impression is that you are trying things at random, without really understanding them.

[Slovniek]

@david551

You should not be using chan_sip on new systems, it will not exist in the version of Asterisk released later this year.

Where can I switch to up-to-date alternative?

@david551

Using _. is not a good idea, as it matches h, and other special extensions.

What should I use? Any random number would work?

@david551

What does this represent? Why do you need a third party service with explicit DTMF? What is wrong with dialling the cell number directly.

This is a phone number +1 (303)-498-5509

@david551

Is it your own number? You appear to be calling yourself, and to listen to both calls on the phone, one has to be on hold, so generating music on hold.

It’s my SECOND number, totally air gapped from this environment.

@david551

I get the impression is that you are trying things at random, without really understanding them.

While I followed Home - Asterisk Documentation and the Asterisk docs, I’m kinda lost, that’s true

[tootai]

Asterisk 20 _X.

[david551]

The replacement for chan_sip is chan_pjsip

[Slovniek]

I am aware of this fact, however due to the setup being a slightly different than for sip, I decided to learn this first. I am also aware that it should be identical in functions as well, I just couldn’t follow the setup the same way as for chan_sip for the expected results.

Went through the docs, that should’ve been for this topic and even though, none of it appears to work. Matter of fact, everything except for this Dial() function worked just fine, I have a suspicion that it’s caused by incorrect communication with the SIP trunk, which feels weird as it’s registered successfully.

Used reference:
https://wiki.voip.ms/article/Asterisk_SIP https://wiki.asterisk.org/wiki/display/AST/Contexts%2C+Extensions%2C+and+Priorities
Hello World - Asterisk Project - Asterisk Project Wiki