Call Delays, redundant SIP

hello, I’ve installed a new AsteriskNow system for the first time. Calls are being delayed significantly when there is a queue. Is this a SIP trunk issue? My default paid SIP account (through commpeak) says they provide 20 channels… which I thought was well enough. However, do I need a redundant SIP trunk as well?


This is an ITSP issue.

So commpeak. What is the suggested / standard way to fix this?

Should I changed the SIP provider? Add a redundant SIP provider?


Obtain SIP traces, with timestamps, as evidence that they have a problem, then talk to the ITSP. If that fails to resolve the problem, change provider.

Note there isn’t a standard way, as this is not a normal problem.

Hello, an update.

I did a SIP trace using tcpdump:

Then analyzed with wireshark… using the telophony pull down.

I am just not real good yet about knowing what things mean. For the most part, my system is doing better… fixed a server timezone issue mainly. My question with analyzing the trace I’m seeing a lot of “401 Unauthorized” lines. When I look at the “flow” of a line, it is usually like so:

5071  INVITE SDP -->  5060
5071 <-- 401 Unauthorized 5060
(last line repeated 10 times)

Is this a spam attack?.. or do I need to open up port 5071 on my firewall??.. or??


Insufficient information. If the peer sent the INVITE, they failed to receive the 401 response, or it contained a contact address they couldn’t use.