No audio when calling with our SIP

[jace]

Hi Guys,

vicibox redux 4.0.3; Vicidial VERSION: 2.8-407a BUILD: 130709-1350 from; Asterisk 1.4.44 from scratch. I'm also using eyebeam softphone 1.1 3007n stamp 17816.

Just new here.

after setting up our server, we tried calling with our SIP, we can hear the softphone ring as well us the number we are calling. The problem was, we can't hear anything from the other line and vice versa when they answered the phone. Ports 5060, 10000-20000 were already opened on the network. We can't make it work both on softphone and agent login to have a successful call. We are also able to access the vicidial outside the network through its external IP.

Any idea what would be the issue?? Please help..

Thanks,

[rrb555]

where is the phone extension located?LAN or on the internet?
try calling other phone extension first to see if you can make successful local phone extension call.

It will help if u post also your carrier settings and dial plan settings.

[jace]

Thanks rrb555 for the quick response..

the dial plan and carrier setting was provided by our SIP provider...

we are calling a US landline, we can hear the softphone ring and they were able also to answer the call. but we can't hear each other after...

Sorry, really new with vicidial. what do you mean by: "where is the phone extension located?LAN or on the internet? "

we setup the phone on Vicidial => Admin tab => Phone. and this is what we use on softphone configuration...it might be over the internet because we only access the vicidial through its external IP. softphone was configured on our workstation. (server and workstations are from different country)

Thanks

[williamconley]

dial plan entries cannot be provided by the carrier. vicidial has specific dialplan entry requirements.

we only charge $25 to set up a standard carrier (because we do it all the time), but it is entirely possible to argue for hours with the settings to make it work.

if you post your carrier settings (all of them, leave out IP/user/pass) we may be able to help you out. you can also search this forum for MANY previously helped newbies with carrier settings.

[jace]

Hi,

Here is our carrier setup and dial plan entry. Just for clarification, this carrier and dial plan entry was provided by our SIP provider.
We don't have Registration and Global string as this is only optional according to them... would it be possible that this will be the issue for not having audio during calls???

[carrier]
type=friend
host=xxx.xxx.xxx.xxx
canreinvite=no
disallow=all
allow=g729
allow=ulaw
dtmfmode=rfc2833
qualify=yes
nat=no
context=trunkinbound

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,n,Dial(SIP/${EXTEN:1}@carrier,60,tTor)
exten => _91NXXNXXXXXX,n,Hangup

Thanks,

[gardo]

One-way or no-audio issues most of the time are related to restrictive firewall/NAT settings. The easiest way to fix this is to put your server on a public IP (and make sure it's properly secured). If putting the server on the internet is not an option, then you might need to forward more ports (RTP) to the server (and enable uPNP). We normally forward UDP ports 10,000 to 65,000 for RTP or whatever is defined in /etc/asterisk/rtp.conf.

[williamconley]

not having audio after a call connects is always firewall related. it can be the "externip" setting in sip.conf (which is a firwall setting!). it can also be that the audio signal is coming from a different IP than the control signal ... often turning off the firewall will resolve that issue unless you have a private network for your vicidial server. in that case you sometimes have to forward ports 5060 and 10000-25000 (all udp) to be sure to get all your audio, but the externip usually "does the trick" for a private IP.

it can also be the "nat=" setting. If either your server or the carrier is behind a firewall, nat=yes. if neither is behind a firewall, nat=no.

[jace]

Hi Guys,

Just wanted to share that this issue has been resolved. The issue was on our voip provider side, though they were not able to isolate the issue and they keep on telling us that this is a firewall/ISP issue. Sure thing is that all needed ports were open. We have come up with this result after trying another voip provider. Everything works perfectly in this new provider. We haven't done anything, just entering the carrier setup then everything works.

Thanks guys for all your help. Really appreciated.

J