NEED MORE INFO EC2 Green, No Audio but soft phones connect to server

[ItsMatt]

Hello esteemed Gurus,
I have problems with my setup making and receiving calls. There is no audio on outgoing calls, incoming calls go straight to voicemail.
I have Google voice with three numbers and extensions 701, 702, 703 configured.
I have inbound routes for each number to go to each extension 701,2,3 and the inbound calls ring 6 or so times, then go to voicemail, but the softphone does not ring.
Extension to extension, 703 to 701 on two softphones, for example, the ringback tone is heard on caller 703 but the 701 does not ring.
701 to 703 is different
instantly indicates call established but nothing is heard. The 703 phone did not ring.
I have setup an EC2 green server and through learning pangs have configured travelin man, setup my dyndns' and my softphone connects on ext. 701 and 703 I have set all ports in the security group that are pertinent by the tutorial to be open to the IP of our office and applied that security group to the Elastic IP.
I can SSH into the server using Putty, and have HTTP access to the admin pages as well.

I am a new FreePBX/PIAF user and am beta testing this for rollout to a group of users at two different sites.
I may be better off with a RentPBX or the like, but am always up for a learning experience.
I really want to get this going and any suggestions you all may have are welcome.
Thanks

 

[Stewart]

Could be a codec issue. Could you post the Asterisk log for the call?

 

[Hyksos]

Hello esteemed Gurus,
I have problems with my setup making and receiving calls. There is no audio on outgoing calls, incoming calls go straight to voicemail.
Which voicemail? Each number goes to the matched destination extension voicemail?

I have Google voice with three numbers and extensions 701, 702, 703 configured.
I have inbound routes for each number to go to each extension 701,2,3 and the inbound calls ring 6 or so times, then go to voicemail, but the softphone does not ring.
Which voicemail do you end up in?

Extension to extension, 703 to 701 on two softphones, for example, the ringback tone is heard on caller 703 but the 701 does not ring.
it doesn't ring but does it go to voicemail? does that work? leave a msg..?

701 to 703 is different
instantly indicates call established but nothing is heard. The 703 phone did not ring.
I have setup an EC2 green server and through learning pangs have configured travelin man, setup my dyndns' and my softphone connects on ext. 701 and 703 I have set all ports in the security group that are pertinent by the tutorial to be open to the IP of our office and applied that security group to the Elastic IP.
Travelin man, dyndns, asterisk sip settings, ec2 security, iptables.... too many layers for now if it doesn't work.
You need to step it down than rebuild successively. Stop iptables, then retest.
Post your asterisk sip settings and your ec2 security if unsure.
Where are the extensions? Behind same router? On same pc? at different site and behind different router?
Switch client... always easy to try... you didn't name yours.
Extensions settings? Nat?

I can SSH into the server using Putty, and have HTTP access to the admin pages as well.
Linux CLI: asterisk -rvvvvvv
is nice to start
asterisk CLI: sip set debug on
asterisk CLI: rtp set debug on
can be useful when the problem is hard to understand, more work is needed to analyse those.

I am a new FreePBX/PIAF user and am beta testing this for rollout to a group of users at two different sites.
I may be better off with a RentPBX or the like, but am always up for a learning experience.
Check your cost. Watch your elastic IP chewing your wallet when you stop using it but still keep it. (read the pricing good, look for elastic ip pricing.)

I really want to get this going and any suggestions you all may have are welcome.
Thanks

with a client that has logs, and asterisk -rvvvvvvv and the debug options... the answer is lying in the data.
You can post tons of screenshots and logs. Logs have to represent the right info... and all that has to be sanitized for your protection.
Learn to read /var/log/asterisk/full after you've watched a call in asterisk CLI.
You can empty it before starting a test to read and extract what is relevant to the last test only:
cat /dev/null > /var/log/asterisk/full

Don't leave your box with lax security and iptables stopped if you're not doing tests, your pants are down during that time.
Consider closing the http access when not using it via SSH, use a strong SSH and maint passwords. But consider closing http... this apache is not suited for public exposure.
You don't want to expose the SIP, it's why you're trying to use travelin man to get just the right holes... Dont expose http .
If at the end, everything is only accessible to white listed IP... then you're good. But usually people don't do that with everything since you can lock yourself out.
SSH can be left public if secured properly.

EC2 boxes are pounded by attacks, I left one unattended and started a timer, took hours someone was in... You can't leave it unsecured even a night...
Check your plan... EC2 compared to OVH or rentpbx is rarely a good choice money wise... It depends... of course. You can do maths!