Remote extension no audio making calls but receiving OK

[marlonbaez]

Hi,

I install an Orgasmatron III on DELL T100 and configured few remote extensions.
I can receive calls perfectlyon remote extensions, but when I try to make calls from the remote extension to Voicemail, another extension or anything, I got no audio both ways.

I configure sip_nat.conf but I needed to remove because then I can't receive calls on remote extensions.

My T100 has 2 ethernet, 1 for localnet (192.168.1.x) and 2 for internet (10.0.0.x) behind a firewall. I'm also got vitelity sip trunk working OK both ways.

Any help?

Thanks in advance.

 

[Boolah]

You need to edit your sip_general_custom.conf file, not your sip_nat.conf file.

See this post for more info: http://www.pbxinaflash.com/forum/showpost.php?p=26253&postcount=2

 

[wardmundy]

Read the knol.

 

[marlonbaez]

Thanks, but after I did this you suggest then I can not receive calls on remote extension and also no audio in outgoing calls from remote extension.

I removed again from sip_custom.conf and I can receive calls on remote extension with audio both ways. But no audio in outgoing calls from remote extension.

Please help.

Thanks in advance.

 

[Boolah]

Do you have a static IP for your Internet connection? Is this the IP address you're using for the "externip" setting? If you have a dynamic Internet IP address, do you have a dynamic DNS hostname which maps to your dynamic IP address? Is this hostname what you're using for the "externhost" setting? Which network are your phones connected to, the 10/8 network or 192.168.1/24 network?

BTW - why do you have two private networks connected to your PIAF box and how do they fit into your network as a whole? You might be double-natting someplace, which could cause significant problems.

We need far more detailed information about your setup than what you're providing in order to help you.

 

[marlonbaez]

I got dynamic IP from ISP and I'm using dyndns on 10.0.0.0 network for remote extensions and vitelity SIP account (working OK both ways).

I got a second NIC on 192.168.1.0 for internal phones (working OK)

Everything is working OK, but no audio (both ways) on outgoing calls from remote extensions.
When I run the ip.sh for updating sip_custom.conf then my remote extensions stopped to work. I included both networks on sip_custom.conf then I needed to clean sip_custom.conf and remote extensions start to work again but no audio in outgoing calls.
I set rtp debug and asterisk is sending packets to correct IP from my remote extension but no audio anyway.
I can receive calls with audio both ways on remote extension.

 

[Boolah]

Presuming you have some mechanism to update your dyndns hostname with your current IP address (like a router or third party software), your sip_custom.conf file should look similar to the below, in which case, you would not need to run the ip.sh script at all:

nat=yes
externhost=<dyndns hostname>
externrefresh=300
localnet=10.0.0.0/255.255.255.0
localnet=192.168.1.0/255.255.255.0

It sounds as though this issue is network related, rather than PAIF (or even Asterisk) specific.

Why do you have two separate private networks (one for remote extensions and one for local extensions)? I presume that since you have remote extensions on a private network, you must be running a VPN? How are the three networks you have logically routed (the two private networks and your WAN). If you're runng a routed VPN (as opposed to bridged), this would make it four networks. How is this VPN logically connected to your other networks?

 

[marlonbaez]

I got my internal network 192.168.1.0 on 1 NIC.
I got Thomson ADSL modem from my ISP in 10.0.0.1 and connected to second NIC on my PIAF on 10.0.0.0 network.

With sip_custom.conf configured like you told, my remote extension stopped to work both ways. Without this is receiving calls but no audio making calls.

I'll try changing my ADSL modem with new model, maybe is modem related.

 

[Boolah]

If you're using your PIAF box as a router (which appears to be the case), you should put your DSL modem in bridge mode, let your PIAF box connect directly to your ISP (through your DSL modem) and remove the 10.0.0.0/24 network.

You will also need to run the ip.sh script to update your dyndns hostname and automatically generate your sip_custom.conf file.

Once again, how are your remote extensions connecting to your 10.0.0.0/24 network, or are they in fact connecting to your WAN IP which is port forwdared to your PIAF box?

 

[marlonbaez]

I'm not using my PIAF as router.
I was running the script to update hostname.
I'm connecting my extensions to WAN IP which is port forwarded to my PIAF box.
I got this same configuration with just 1 NIC on other office and works perfect. But not this one with 2 NIC. I don't know why.

 

[foneman]

Ports

Just a thought, We were running into a similar issue with an install and some remote extensions. The remotes were on Comcast and now they block port 5060. We could receive calls but could not make them and also could not actually see it registered. After many long hours, we tried to move to port 5071 ( or some like) and VIOLA! it worked.
Just a thought....