Hi, that could be caused when your upstream offers "100rel" and your Asterisk does not get a response fast enough from your upstream. Is your outbound peer monitored by the qualify feature (qualify=yes)? Then asterisk should calculate the round-trip-time until a response arrives and should not resend the packets too fast. If that does not help, you could play around with the "timert1" settings in your peer's SIP configuration.
Also, this sounds to me like a bug on the carriers side. It seems they are maybe offering "100rel" to you, but do not send any SIP/1xx responses regarding your INVITE so your Asterisk resends these INVITEs because they are assumed lost. Since these INVITEs all have the very same Call-ID and CSeq number, your carrier's equipment should be able to determine these packets are regarding the same call. Again, I think your carrier should fix this problem on his site. If he wants to enforce rate-limiting to INVITEs he should do it right by honouring the Call-IDs and sequence numbers. If you like you can anyway send me your trace off-list, maybe there's something other weird going on. Greetings Max Am 22.02.2017 um 18:57 schrieb Jeff LaCoursiere: > > Hello, > > I have two upstream providers we use for US termination. The dialplan sends > calls out the "primary" and if that fails for specific reasons, it sends the > same call out the "secondary". This has worked well for us when we are lazy > about keeping balances up, for example. > > Starting a few days ago ALL calls sent to the 'primary' were returned as > busy, though the secondary terminated them fine. We have a balance, and > funny enough international calls are going through fine, just not US calls. > I opened a ticket. > > The response form the carrier is that our asterisk is sending four > simultaneous invites within one second, and for that reason the call is > rejected. > > I did a packet trace and was able to confirm this is true - only US calls > sent to this carrier cause our end to send four identical simultaneous > invites. When it fails, a single invite for the same call is sent to the > secondary, which is terminated without issue. > > Happy to send the SIP trace if any would care to see it, but is there a > reason anyone can think of that our asterisk (11.11.0) would suddenly start > doing this? It may be that it has been doing it all along, and our carrier > just started rejected calls that come in this way, I'm not sure. > > Cheers, > > j > > -- Viele Grüße aus dem Tal Max Grobecker -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
