I install Asterisk 13 on CentOS7 and everything is fine. I also configured Asterisk dynamic realtime to register my SIP users and peers. I was able to place a call from one extension to another successfully.
My challenge now is placing a call to my SIP trunk provider.
I initially stored the sip trunk provider detail in the Mysql database just like sip users and peers, but I noticed that it could not register with the SIP provider.
I then added this line in the general section of my sip.conf:
register => 1234567:abcdef@callcentric.com
After adding the above line in the sip.conf, I was able to register, but yet I could not place a call using that trunk, rather I always got the error:
_ == Using SIP RTP CoS mark 5
– Executing [0112348023950246@NaatCast-1:1] Dial(“SIP/1000-00000004”, “SIP/0112348023950246@callcentric.com”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/0112348023950246@callcentric.com
[Feb 19 14:05:16] NOTICE[7593][C-00000003]: chan_sip.c:24002 handle_response_invite: Failed to authenticate on INVITE to ‘sip:1000@10.32.0.232;tag=as0d259ea6’
– SIP/callcentric.com-00000005 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/1000-00000004’ status is ‘CONGESTION’_
Below are my configurations:
SIP.CONF
[general]
allowguest=no
udpbindaddr=0.0.0.0:5060
disallowed_methods = UPDATE
srvlookup=yes
dtmfmode = rfc2833
register => 1234567:abcdef@callcentric.com
rtcachefriends=yes
EXTENSIONS.CONF
[NaatCast-1]
exten => _X.,1,Dial(SIP/${EXTEN})
exten => _011234.,1,Dial(SIP/${EXTEN}@callcentric.com)
MYSQL DB:
MariaDB [asterisk]> select * from SIP;
±—±------------±-----------±---------±------------±----------------±------±-----------±---------------------±--------±-----±-------±--------±----------±---------±------------±-------±------------±-----------±----------------±------------±------±-------------------±-------------±---------+
| id | NAME | context | dtmfmode | fromuser | host | port | ipaddr | nat | mailbox | deny | permit | qualify | secret | callerid | directmedia | type | defaultuser | regseconds | fromdomain | insecure | allow | disallowed_methods | videosupport | disallow |
±—±------------±-----------±---------±------------±----------------±------±-----------±---------------------±--------±-----±-------±--------±----------±---------±------------±-------±------------±-----------±----------------±------------±------±-------------------±-------------±---------+
| 1 | 1000 | NaatCast-1 | rfc2833 | NULL | dynamic | 5060 | 10.32.0.15 | force_rport, comedia | NULL | NULL | NULL | no | password0 | 1000 | yes | friend | 1000 | 1519070699 | NULL | port,invite | NULL | NULL | NULL | NULL |
| 2 | 1002 | NaatCast-1 | rfc2833 | NULL | dynamic | 49410 | 10.32.1.48 | force_rport, comedia | NULL | NULL | NULL | no | password1 | 1002 | yes | friend | 1002 | 1518771963 | NULL | NULL | NULL | NULL | NULL | NULL |
| 3 | callcentric | NaatCast-1 | NULL | 177729xxxxx | callcentric.com | 5060 | | NULL | NULL | NULL | NULL | NULL | password2 | NULL | no | peer | 177729xxxxx | 0 | callcentric.com | port,invite | ulaw | UPDATE | no | all |
±—±------------±-----------±---------±------------±----------------±------±-----------±---------------------±--------±-----±-------±--------±----------±---------±------------±-------±------------±-----------±----------------±------------±------±-------------------±-------------±---------+
The two extensions 1000 and 1002 are able to call each other.
My concerns now are these:
- Why am I getting the error shown in my log above whenever I try placing calls via the trunk?
- Do I still need to populate the DB with SIP trunk details after having the sip register configuration in the sip.conf?
- What is the best way to configure SIP trunk while implementing Asterisk ARA?
Please, I will appreciate anyone with a headway on how to make this work. @jcolp @johnkiniston @UncleWard @Pentium-5 @ambiorixg12 @gjoseph