PROBLEM: I want to setup call forwarding. When i set @huskervoip things are working but if i set @digitalphone-unlimited it is not. Below is the error i get on Asterisk CLI.
This is included in /etc/asterisk/extensions.conf
[digitalphone-unlimited]
#include /etc/asterisk/sip-extensions/external-providers/digitalphone-unlimited.conf
#include /etc/asterisk/sip-extensions/internal-extensions/*
[digitalphone-az]
#include /etc/asterisk/sip-extensions/external-providers/digitalphone-az.conf
#include /etc/asterisk/sip-extensions/internal-extensions/*
Included files under extensions --> 4024645971.conf
exten => 4024645971,1,Dial(SIP/18158340675@digitalphone-unlimited,30)
exten => 4024645971,2,VoiceMail(4024645971@digitalphone-voicemail)
exten => 4024645971,3,PlayBack(vm-goodbye)
exten => 4024645971,4,Wait(2)
exten => 4024645971,5,HangUp()
exten => 100,1,VoiceMailMain(@digitalphone-voicemail)
Included files under sip.conf --> 4024645971.conf
[4024645971]
type=friend
username=4024645971
secret=d2v.net
qualify=yes
host=dynamic
accountcode=joel
nat=yes
canreinvite=no
allow=all
mailbox=4024645971@digitalphone-voicemail
context=digitalphone-az
Asterisk CLI Error
-- Executing [4024645971@digitalphone-unlimited:1] Dial("SIP/4024648707-09d64118", "SIP/18158340675@digitalphone-unlimited|30") in new stack
[Dec 10 16:10:26] WARNING[4018]: chan_sip.c:2867 create_addr: No such host: digitalphone-unlimited
[Dec 10 16:10:26] WARNING[4018]: app_dial.c:1131 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [4024645971@digitalphone-unlimited:2] VoiceMail("SIP/4024648707-09d64118", "4024645971@digitalphone-voicemail") in new stack
-- <SIP/4024648707-09d64118> Playing 'vm-intro' (language 'en')
== Spawn extension (digitalphone-unlimited, 4024645971, 2) exited non-zero on 'SIP/4024648707-09d64118'
joel*CLI>