Asterisk [Aheeva] totaly random call drops

Hello everyone. Finally after months of search and tries I give up and really need your help.

So the problem. I came across an asterisk box (v1.4) running a commercial call center named Aheeva (its a cr@p btw, 0 customization, try to change anything and you are doomed). Calls drop random from agents (IAX2 softphones) and from the 4 hardphones (Thomson 3020). I tried everything from sip.conf changes (rtptimout,rtpkeepalive, externip, nat, codecs, qualify etc) to even new ethernet cabling. Also I have contacted all providers (PRI, BRI) to check their lines and its all fine.

Let me inform you about the network setup. All networking pass through a 3com 4000 48port and another 3com 24port (cant rember the name) switch. Asterisk calls through 2x Sangoma Vegas. 1 x Vega 50 with 4xBRIs and 1x Sangoma Vega 400 with 4x PRI.

So I’m giving you my best log that caprtures a dropped call. It’s messy because all agents work at the time. I can’t capture it alone because the issue is RANDOM.

As I write in the 1st line of the log. Drop time approx. 14:23 - Extension OUT2463077332
And the link to download it. It’s big so cant paste it here app.box.com/s/wgwetytxbi3vq9f4en21
Ask for anything needed. Box’s IP is 192.168.1.6 And thanks anyone who touches the log file. It’s really a war out there

sip.conf

Asterisk 1.4 is obsolete.

It looks as though Asterisk isn’t responding with ACK to redundant 200 OK responses, so this means that, if the network loses the first ACK, the call will eventually fail.

If that bug existed in some versions of 1.4, it has long since gone. You should definitely make sure you are using the end of life release of 1.4 and preferably use at least a current version 1.8.

Rly my hands are tied because of the “aheeva” license. I’m stuck to 1.4 asterisk version.

To be exact these guys use a modded version, i think, because they provided an aheeva-asterisk source package.(Btw after the installation,and while they registered the product, they ssh’ed and deleted all the installation files, LAME!) The exact version if it helps is Asterisk 1.4.43.19226-AheevaCCS-3.2.13

Is there any way that I stopped asterisk from sending ACKs?

Also thanks 1000 times for answering.

If the peer doesn’t get an ACK, they will repeat the final status (200 OK). Asterisk has to skip most of the processing this. I guess you have a version with a bug that results in it suppressing too much processing. The first ACK would need to get lost for this to happen, so you could try improving the network.

Really, in this sort of situation, is is up to the supplier that is using the modified version to provide support. However, it is a legal obligation that they release the modified source code to you on request (unless they supplied the binary more than five (might be three) years ago. Unfortunately, 1.4.43 is the next to last version of 1.4, and there is no fix listed for a problem like this. Of course, it might even be their changes that broke it.

I think you have issues.asterisk.org/jira/browse/ASTERISK-17259 although it does seem to have been fixed before 1.4 went security fixes only. A more careful look at the SVN logs should confirm or deny this.

Thanks man rly appreciate your infos. I’ll try to make some tweaks on LAN network, maybe try another switch. Cabling is new, recently changed it.

And another question that came up. Maybe it’s an idiot one.

Doesn’t Asterisk send ACKs, or the ACKs does not reach the other end? I’m just having some scruples about the good “behaviour” of the VEGA 400 PRI gateway. Vega Syslog though doesn’t report errors.

Assuming the logs are from Asterisk, it is not sending at least some of the ACKs. I didn’t check if it sent the first one. I assume it sent the first one, but that was lost by the network,. A bug then cut in and prevented it re-sending it.

You were totally right man. I finally learned that they indirectly accepted that the version is buggy and they try to sell the new version now. You said “However, it is a legal obligation that they release the modified source code to you on request” but even that is translated by them to “Give me money for anything!”

They don’t even provide the files for reinstallation even the license is payed for that old version. Thanks anyway again david55. At least I can stop now searching for “ghost” problems. oO and they are monitoring this conversation. They are angry because I said earlier that their poduct is CRAP. May I add that their support is also CRAP. And please Aheeva guys, I and only I said CRAP not your unhappy clients. Please don’t put words from this conversation to your mails. It’s irrelevant.

[quote=“kamui”]You were totally right man. I finally learned that they indirectly accepted that the version is buggy and they try to sell the new version now. You said “However, it is a legal obligation that they release the modified source code to you on request” but even that is translated by them to “Give me money for anything!”
.[/quote]

I think you are allowed to charge the actual cost (with no profit element) for providing the source code, but if they are charging any more than that the Free Software Foundation is not going to be happy about the abuse of their licence, although it would have to be Digium that took any legal action.

The one reservation is that you have to request the source within three years of receiving the binary.

The licence is, actually, designed so that it is much better for the source to be provided at the same time as the binary.

They cannot stop you from redistributing the binaries of Asterisk itself, including customisations, although any custom dialplan and external programs may be on more restrictive licences.

They don’t have to provide you with subsequent modifications that they made.