MY question: I just installed the latest IncrediblePBX for the RaspberryPi 13-13.7 with Asterisk 13.22.0, with all of the updates. I have not made any manual changes to GVsip1. Google Voice is working, and a couple of extensions as well. I can’t get the Vitelity inbound route to work. Vitelity Outbound works, the DID is correct, I called Vitelity on that. I have left CID to “Any”. I even created a route from Any DID and Any CID, and still no answer. At one point it was working, but then GVsip1 was not, now GVsip1 is working and now Vitelity inbound is not. The extensions are working.
I don’t have a fixed external IP, and my internal IP is 192.168.1.206.
Is this possibly the problem? The only external inputs is Google Voice, and Vitelity Inbound.
I have had bad luck with dynamic dns providers
My DID is correct.
But Vitelity sends me an email that “chanunavail” error. In the log I see:
So Vitelity is getting thru. From the report information, you can see that
Both Vitelity inbound and outbound are registered, but unmonitored.
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
163/163 192.168.1.201 D No No A 5061 OK (19 ms)
165/165 192.168.1.164 D Yes Yes A 5060 OK (115 ms)
701/701 192.168.1.164 D Yes Yes A 5060 OK (148 ms)
702/702 192.168.1.101 D No No A 34787 OK (19 ms)
vitel-inbound/xxxxxxxxxxxx 64.2.142.116 Yes Yes 5060 Unmonitored
vitel-outbound/xxxxxxxxxxx 64.2.142.216 Yes Yes 5060 Unmonitored
6 sip peers [Monitored: 4 online, 0 offline Unmonitored: 2 online, 0 offline]
Is it the vitality inbound registration that asterisk thinks fails?
asterisk -rddddddddd
I am new so I only get two links, and there is no definition of a link. So I will post this,
What am I missing?
Where else should I look?
Thanks for your help.
I am not experienced any more in Asterisk or Linux. It has been 10 years since the last time I created a Asterisk system, so I am very much out of practice. I thank you in advance for any help, or direction if I am in the wrong forum for this type of question.
David, Thank you for your reply. I am not an Asterisk expert. It has been a decade since I installed it last. Let me know what terms might I explain more. The Asterisk output was from debug mode. I don’t know how to get the contents of the Sip Packets. Here is what I did:
Connected to Asterisk 13.22.0 currently running on pi-new (pid = 6142)
Core debug was 9 and is now 10.
If you can tell me how to get the Sip contents I will do my best to post it here.
If you mean the Sip Settings, here is the Vitelity Inbound Configuration:
username=xxxxxxxxx
type=friend
t38pt_udptl=no
t38pt_tcp=no
t38pt_rtp=no
secret=zzzzzzzzz
insecure=port,invite
nat=yes host=inbound18.vitelity.net
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
OK, I am going to work on tcpdump in the right format. But first the Vitelity Inbound quick busy was partially resolved. Vitelity was not sending the inbound information to Asterisk. Somewhere along the way the SIP routing on the Vitelity side was lost. For those who might have the same problem, here is the solution from Vitelity Support. Make sure you have an endpoint in the Subaccount:
To update the routing method for your numbers, please follow the
I can’t figure out how to get the tcpdump in a printable format so that I can post it here. The first 1xx chars are in binary, so not translated to printable. Here is what I did for tcpdump:
apt-get install tcpdump
tcpdump --version
tcpdump version 4.9.0
libpcap version 1.6.2
OpenSSL 1.0.1t 3 May 2016
tcpdump -A -w /var/log/asterisk/tcplog -i eth0 -T pgm
-w write to a file
-i eth0 (Ethernet Interface)
-T pgm (type of data)
I tried -A, -b, -d options separately, and still I have binary in the first 1xx characters. I read every option in the man tcpdump and still could not figure this out. Any help would be appreciated. Thanks
Now, I have to search for a new topic. The first time I connect with Vitelity, is rings many times and I can answer and connect. The next time after a successful connection, it rings once, and then I get an extension is busy response. Any suggestions while I work on getting the logs?
If you want to present protocol traces for packets that actually reached Asterisk, you want to use the in built protocol logging options, e.g. sip set debug on. People here are most familiar with debugging from them.
If you are capturing SIP that Asterisk is not seeing, you can read the tcpdump file into wireshark.
tcpdump doesn’t have the deep knowledge of protocols to be able to print out SIP cleanly, although tcpdump -r /var/log/asterisk/tcplog would analyze it to the extent that it can do so.