TIPS No IVR Audio (or any other asterisk sounds)

[nahrwoldinternet]

Hello Everyone!
I'm having a problem with my asterisk system after we changed router/firewalls. (Yes. I know... the answer is "It's a nat issue, fix your nat/firewall." Please Listen)

We've just switched from a Netgear router with a built-in SIP ALG to a custom built Ubuntu router/firewall to deal with some new scans that we're required to pass. We are running an IncrediblePBX/XIVO/Asterisk 13.11.2 server currently terminating two DIDs from DIDforsale. Everything ran fine through the Netgear router with port forwarding enabled, but we couldn't secure it very well, and it kept needing to be rebooted.

When we switched to the custom router/firewall, we could call out with 2-way audio perfectly fine. We couldn't call in and get our IVR to answer. After some troubleshooting, I noticed that the calls actually were connecting in the logs. I set the default incoming call extension to a single phone, and the call goes through with 2-way audio (although the calling party does not hear any ringing on their end, just silence until the called party picks up). If the called party does not pick up, the logs show voicemail activating, but the calling party doesn't hear anything.

Obviously, the problem is in the NAT/firewall. I get that. I've tried everything I can think of when it comes to port forwarding. What could cause audio to flow in both directions, but pre-recorded audio (yes, I've tried the asterisk default "Allison" sounds and IncrediblePBX Demo IVR as well) to not flow? Is there anything I can set in the interface/config files that would help?

Please help!

--Troy

 

[nahrwoldinternet]

Also seeing the following in the logs right before the call hangs up. I think it might have something to do with the audio problems, but can't find anything helpful on how to fix it through Google.

[Jan 23 18:20:11] WARNING[51260]: chan_sip.c:4064 retrans_pkt: Retransmission timeout reached on transmission b90d9b77-7b36-1236-dfb1-4f9b82eaa13e for seqno 118038725 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Jan 23 18:20:11] WARNING[51260]: chan_sip.c:4088 retrans_pkt: Hanging up call b90d9b77-7b36-1236-dfb1-4f9b82eaa13e - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

I read the Asterisk wiki, but can't make heads or tails of how to handle this.

 

[kyle95wm]

Did you set up your NAT settings within your PBX?

 

[islandtech]

If you didn't have a static external ip, you probably got a new ip from the isp.
check asterisk settings -> asterisk sip settings and see if the external ip is the same as configured

 

[nahrwoldinternet]

Did you set up your NAT settings within your PBX?

I did set them up. But this is Xivo, and there not exactly all in one place. Maybe I missed one of them? Could someone walk me through where all of the NAT settings are in Xivo? Or even bettet.. .could someone point me to the config file, because I can't find it?

If you didn't have a static external ip, you probably got a new ip from the isp.
check asterisk settings -> asterisk sip settings and see if the external ip is the same as configured

Yes, sir... we've had a static IP for several years now, and everything's been working fine. it's just strange because outgoing audio works... but IVR doesn't work... they should be the same thing, right? no port difference between the two...

Thank you all for you help... I'm hoping to get this thing running soon (i've been sick lately). My church has been without a phone system for too long.

 

[billsimon]

Please take a sip trace of a failed call and post it on pastebin. Any IP mismatches should then stand out.

"sip set debug on" from the Asterisk CLI.

 

[nahrwoldinternet]

Please take a sip trace of a failed call and post it on pastebin. Any IP mismatches should then stand out.

"sip set debug on" from the Asterisk CLI.

Please help... I can almost make heads or tails of this, but not really... https://pastebin.com/Vg01EdRV

Bad call (Trying to contact IVR, no response) first. Good call (call to one extension) second. I've dropped ##### at the start/end of each to make searching easier.

thank you all for your help!

 

[billsimon]

I see the internal IP in your Contact: header and in the SDP, and I think if you are using Asterisk's NAT functions these will be substituted with your public IP. Your call is getting dropped because Asterisk isn't getting an ACK back to its 200 OK to the incoming INVITE. There's no ACK because the Contact: is not routable and you are not going through a proxy (though it looks like there are several proxies between you and DID4Sale, I assume on their side of the network). So you still have some work to do on your NAT settings.

I don't know where it all is in Xivo. The general Asterisk config file would be sip.conf. You need to specify your localnets and externaddr

 

[nahrwoldinternet]

Well, I fixed this problem (But have now have another that I'll start a new thread for...) by scrapping Xivo and its 16-layer menu structure and starting fresh with IncrediblePBX 113-13. Now my incoming calls route just fine. And I'm back to the FrePBX settings that I've been using for the pas 7 or 8 years. I think it was a huge mistake on my part to try the "latest and greatest" when Xivo came out...

Thanks for all the help on this problem everyone!