[asterisk-users] Problems Solved, two left

[Steve Matzura]

And I think they're both small.


Solved: tcpdump showed no packets coming in, so I went to my DID 
provider's Website to discover to my intense embarrassment that the DID 
number had been set up forwarded to their voicemail. I got egg on my 
face for this one. I changed that setting to SIP/IAX and packets now 
arrive and go where they should. Two problems remain.


1. Still can't register my phone


The username and password are correct. I don't know what else to try.


2. Asterisk can't find the extension in my inbound context.


[May 23 18:34:12] NOTICE[46582]: res_pjsip_session.c:3968 new_invite:  
voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected because 
extension not found in context 'voipms-inbound'.


I changed the name of the context in pjsip's  to 'voipms-inbound' and 
removed reference to '[mycontext]' from pjsip.conf and extensions.conf 
as they were superfluous. The endpoint section of pjsip.conf now reads:


[voipms]
type = endpoint
transport = transport-udp
context = voipms-inbound

...


The bottom part of extensions.conf (with the phone number obfuscated) is 
now:


[voipms-inbound]
exten => 3115552368,1,Goto(hello,200,1)

[phones]
exten => 101,1,Dial(PJSIP/yealink)

[hello]
exten => 200,1,Answer()
         same => n,Playback(hello-world)
         same => n,Hangup()


The idea was for any inbound call to the public network number to 
immediately go to extension 200, play the message and hang up, and you 
could still call extension 200 to here it from inside.