Need a help to troubleshooting on incoming calls from outside

I’m fairly new to Asterisk pbx. I have an Asterisk in our network which has been working fine. However, we noticed that the incoming call from outside to our main number stopped working. Since there are not that many calls on the main number, we don’t know when it stopped working. Anyways, I’ve been trying to troubleshoot on this issue and no luck so far. When a call is received from outside on the number, it should play the main greeting and give you an option to choose. However, when the call is made, it connects, nothing plays back, and disconnect after 5s. The following is what I captured from Asterisk.

[Nov 7 15:28:54] == Using SIP RTP CoS mark 5
[Nov 7 15:28:54] > 0x7f94f0015370 – Strict RTP learning after remote address set to:
[Nov 7 15:28:54] – Executing [17038441467@from_sipus:1] Goto(“SIP/sip_us-0000000c”, “cpc-menu,s,1”) in new stack
[Nov 7 15:28:54] – Goto (cpc-menu,s,1)
[Nov 7 15:28:54] – Executing [s@cpc-menu:1] Wait(“SIP/sip_us-0000000c”, “1.5”) in new stack
[Nov 7 15:28:56] – Executing [s@cpc-menu:2] BackGround(“SIP/sip_us-0000000c”, “custom/cpc_main”) in new stack
[Nov 7 15:28:56] – <SIP/sip_us-0000000c> Playing ‘custom/cpc_main.slin’ (language ‘en’)

I compared to the one that is working at another office. It looks like “Strict RTP switching” is not happening. Instead, calls are disconnected. I’ll really appreciate if you can provide any help.

Is this a pure Asterisk server which someone (maybe not you, no problem if so)
has configured with a crafted dialplan, or is this a FreePBX, Issabel or
similar system, which is based on Asterisk, and has a GUI interface?

If it is FreePBX, Issabel etc, then you would be far better off asking on their
forums / support sites / mailing lists, because they know what they did to
make Asterisk work in their way; we don’t.

If it’s pure Asterisk, then:

  1. please explain a bit about the setup you have:

a) which version of Asterisk?

b) which Operating System is it installed on, and which version?

c) is there any NAT involved in the route betwen the Asterisk server and the

  1. Please provide more detail than you already have about what occurs in the
    Asterisk console (I suggest at least “asterisk -Rvvv” to get sufficient
    verbosity) or verbose log file (if one is being created) when a call comes in.

  2. A SIP packet trace (eg: from tcpdump, tshark etc) would be helpful.

  3. Please don’t send screenshots; copy and paste plain text

  4. You said “incoming call from outside to our main number stopped working”
    and also “there are not that many calls on the main number” - does this mean
    that there are other inbound numbers which are still working without problem?

The more detail you can provide, the more likely it is we can try to help.


Thanks for your reply Antony,
Let me provide more information.
version of Asterisk : 16.4.0
OS : CentOS release 6.10
There is nothing involved in the route between the Asterisk and to the Internet. The main number is routed through Once the test call received on Asterisk, no audio can be heard and it disconnect the call. I think it has to do with audio stream.

I finally figured it out. Sip.conf file was missing two things.

I’m not sure how it got removed from sip.conf file. Once I put them, it started working.

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