Hi everyone, I’m willing to pay anyone that can help me with this as Im in dire need! Ive paid for tech support through freepbx but they haven’t answered my ticket from over 5 hours ago so im cryin out as im losing my hair by the minute.
I have two deployments that are up to date 5.211.65-21 all modules up to date. One is behind NAT one is on a public They were working fine yesterday but as of this morning, all incoming calls the caller gets no audio. Its silent for any voice, IVR, Music on hold, nothing. Outgoing calls work perfect. I have two other deployments that have no issue… Ive tried adding a new trunk from another VoIP provider but has same issue. I don’t know where to turn. Hoping someone can point me in the right direction!
Check your IT dept, make sure they have not made changes to any routers or switches. On a site I work, the IT guys constantly mess with ports and mess up our equipment all the time. I think they get bored and ask, who can I mess with today.
If you’re at home, again check for ports being blocked by firewall rules, maybe someone updated firewall rules. Normally one way audio is from closed ports or nat issues.
Thanks for the reply Deanot26508, I kind of am the IT department for these locations. I was hoping it might of been a firewall issue and so to eliminate possible firewall/nat issues I took one deployment and temporarily put it on a public IP in front of the firewall, still same exact issue. I’m kind of wondering if its a codec issue, but I have other deployments with the exact same settings for trunk/sip/codecs and they are having no issues. I’m going gray trying to figure this one out. I really wish paid support would get back to me
Spoke with paid support, they agreed it was really weird, had a couple guys tinkering with it until one decided to try a SipStation trunk which worked perfect! So now I’m trying to figure out what went wrong with my VoipInnovations and Flowroute trunks! One is IP based and the other is SIP Registration like SipStation is, neither work for incoming calls like SipStation does. So I’m still at a loss
This sounds like an RTP port issue under the Asterisk SIP settings, or a weird firewall rule.
I would also agree it sounds very much like a firewall/rtp port issue.
When I set our sip trunks up I had the big advantage of having access to the firewall and could trace what ports were required and allow them as required. Even then it took quite a bit of tweaking to get right.
RESOLVED / UPDATE
Ok turns out it had nothing to do with the firewall or router. I ended up calling my Origination provider (VoIPinnovations) and we teamed up with Level 3 (who I have all my DIDs with) and turns out that Comcast was the problem all along! While they weren’t using SIP ALG or anything like that, their actual route to Level 3’s Chicago Datacenters was broken. I called them and within minutes they had fixed the route and I was able to TraceRoute to Level 3’s IP address without issue. After that,calls worked perfect! To figure all this time I was blaming our equipment, what a waste of time!
I had a similar issue. It even happened during a call.
Found a ban even in /var/log/fail2ban.log that correlated with the exact time the one way audio issues cropped up. It’s a normal process with that sip switch (I think they use a metaswitch) to present 401’s as part of the sip negotiation. Had to white list them under ignoreip in /etc/fail2ban/jail.conf.