Status is 'CHANUNAVAIL'

Asterisk Asterisk Support
1

I successfully connected IP phones and analog phones in internal network.
Now I also want to communicate with outside world using PSTN line.
But I got the following error on Asterisk CLI:

== Using SIP RTP CoS mark 5

0x7f485c004990 – Strict RTP learning after remote address set to: 192.168.1.10:16406
– Executing [9867019489@internal:1] Dial(“SIP/line1-00000018”, “SIP/9867019489@pstn,60”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/9867019489@pstn
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/line1-00000018’ status is ‘CHANUNAVAIL’

I’m using Linksys SPA 3102 ATA to communicate with the PSTN line.
Anyone here, help me to fix this??

2

Please read the forum before posting, as there are several threads (probably unrelated), from people receiving CHANUNAVAIL on SIP destinations.

3

Yes I read the forum, but I found my problem is quite different than posted ever. So I created a new topic. Can you please help me in this problem?

4

Can you please explain how your problem is completely different, and also provide the detailed logging asked for in the other cases.

CHANUNAVAIL is a secondary report, typically meaning the peer is undefined, unregistered, or failing on qualify.

5

I’m using Asterisk with Linksys 3102 ATA for PSTN connectivity. And I’m a beginner and completely new to asterisk, so I’m confused!

6

If you want my all configuration I’ll post here, if it help to solve problem.

7

We almost certainly need logs in sufficient detail to reveal the primary error.

8

Thank you @david551, I solved that problem by just changing Register value no to yes on SPA 3102 ATA.
Now I’m building the IVR menu that response to the customer by saying to dial extension number if you know otherwise dial 0 for customer support.
Suppose caller dialed 0 and talked to the customer support representative, if support representative can’t solve caller’s problem and caller wants to talk to other high level staff and support representative forward that call to the high level staff.
For this what should I add to the extensions.conf ?

9

Your specification is too imprecise, and, in any case, these forums are not here to provide free design services.

10

I’m not asking about free design services. I’m just asking about one line code that I can integrate to my extensions.conf. If you know please help me.
Just asking about a line that take dtmf input while calling.

11

I know this is something simple that I am just missing. I can not transfer a call with DTMF tones. I can transfer a call with the “transfer” button on my sip phones. DTMF works in the IVR, voicemail, and so on. But if I am in a call and press ##, nothing happens. (Same with *2 and *1 as well) I need this as not all the phones have a transfer button.

12

There are options[1] to the Dial application which are used to turn on the DTMF based transfer functionality.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial

13

Thanks @jcolp for your kind help.