[resolve] Call Failed : Reason Code 503 on my Ip Phone

Asterisk Asterisk Support
21

Your contact= line for belgium-voip-provider may be incorrect, but as you’ve removed it I can’t tell you precisely how. It should be sip:<hostname> and that is all.

22

Sorry, just a quick remark, are you sure your sip provider accepts calls in national format?

My experience is that most sip providers only accept calls in international format… just my 2 cents…

23

I type on my ip phone only the prefix with the number : 04 XXX XX XX (For fix phone), the 04 is the prefix for Liege in Belgium. Because my provider is already in Belgium. But yes, I can make national call with this provider.

It’s work when I do It with FreePBX because I have a GUI Web. But I want to understand what happen in the Asterisk from scratch… And In FreePBX I can’t custom the configuration files. That why I make a fresh install of Asterisk 13.19.2 on a Ubuntu Server 16.04 LTS.

24

Any idea ? I’m connected to my provider Belgium Voip

IPBX-Asterisk*CLI>
    -- Contact belgium-voip-provider/sip:<userid>@voip.belgium-voip.com:5060 is now Reachable.  RTT: 99.014 msec
  == Endpoint belgium-voip-provider is now Reachable
IPBX-Asterisk*CLI> pjsip show endpoint belgium-voip-provider

But always this message error in the console CLI:

 Executing [042680871@from-internal:1] Dial("PJSIP/102-00000002", "PJSIP/042680871@belgium-voip-provider") in new stack
    -- Called PJSIP/042680871@belgium-voip-provider
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/102-00000002' status is 'CHANUNAVAIL'
25

I change my dialplan to this one in my extensions.conf

extensions_conf
extensions_conf.JPG1509×951 73.3 KB

But I have this message now in the console CLI of Asterisk:

Setting global variable 'SIPDOMAIN' to '192.168.40.55'
    -- Executing [042680871@from-internal:1] NoOp("PJSIP/102-00000027", "####### [from-internal] ########") in new stack
    -- Executing [042680871@from-internal:2] Set("PJSIP/102-00000027", "CALLERID(num)=042771412") in new stack
    -- Executing [042680871@from-internal:3] Dial("PJSIP/102-00000027", "PJSIP/042680871@belgium-voip-provider") in new stack
    -- Called PJSIP/042680871@belgium-voip-provider
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [042680871@from-internal:4] Hangup("PJSIP/102-00000027", "") in new stack
  == Spawn extension (from-internal, 042680871, 4) exited non-zero on 'PJSIP/102-00000027'

On the screen of my ip phone the messsage below:
488 Not Acceptable
Try different vocoder

26

I resolved this issue. The problem was the codecs that I didn’t use in my configuration. So I just add the line below in the file pjsip.conf for the extensions and the trunk. :slight_smile:

allow = alaw,ulaw,gsm,g723,g722

Now, I can make some outgoing calls. :blush:

27

You should contact your trunking provider and get from them what codecs they support.

There’s no point in offering codecs they won’t accept.

Likely they only accept ALAW or ULAW depending on where you are.