I have installed freepbx on centos 5.
all the system is work fine but when i call a GSM phone for example the call is established but i dont hear the other side and he can’t hear me also.
please i need help.
also i want ask if it’s nesserray to install asterisk-sounds (its not mentionned on the documentation).
What type of phones, what type of trunks and what versions of Asterisk, FreePBX etc?
Re. the sounds, if you run ‘make menuselect’ after the ./configure stage in the Asterisk source directory, there are (I think) three sections relating to the sounds near the bottom of the menu.
I normally go through and select all the english ones, just to be sure I have consistency over different installations.
It will then download them as required during the ‘make install’ stage.
i have xlite phones, trunk iax2, Asterisk 18.104.22.168, freepbx 2.3
the problem is that i hear audio if the i do internat calls (between extension) example SIP2000 calls SIP 2001 its correct and work fine, but the problem when i call outside i dont hear the communication, i hear the ring but when the other side answer i dont hear any thing and he also dont hear anything
You don’t say what kind of trunks you are using. Providing complete details up front helps instead of us having to guess on things…
So here I go guessing: you are using a SIP trunk, with a router/firewall in between the server and the provider.
If I’m right then 95% of the time the answer is that you have not setup the firewall and/or nat’ing properly.
See: http://nerdvittles.com/index.php?p=216 I didn’t write it but in the top 10 paragraphs it will cover this and probably several more issues you might have happening.
ill try to resume all :
i have 3 boxs in the same network :
- Trixbox Box :trunk iax2, sip extensions, the trunk is registrated ==> internal calls work fine but external calls the call is established, i hear the ring, but i dont hear the communication.
- Freebpx : Centos 5, asterisk 22.214.171.124, sames config (trunk and SIP) ==> same problem
- Old Freebpx Box : same config (trunk and SIP ) but it works Ok
==> when i add any Box, the problem come
Things to try:
Check the sip_nat.conf file in /etc/asterisk and make sure the externip & localnet parameters are correct.
(You may have to create it).
Unless each machine has it’s own unique public IP address, you are going to have problems or at least complexities…
You normally need to forward a range of SIP & RTP ports from the internet to ‘The PBX’ machine, but with more than one PBX behind the same NAT system, you have to pick which ports go to which machine…
I’m guessing it should be possible to split both the SIP range and RTP range over the machines, ie.
5060-5064 to original box,
5065-5069 to box 2
5070-5074 to box 3
10001-13999 to original
14000-16999 to box 2
17000-19999 to box 3
And set the matching ranges in rtp.conf on the respective machines.
IAX2 uses one port, that needs to go to the machine with the IAX2 trunk.
You will then have to change the trunk setups in the new machines to ensure they use SIP ports in the correct range for the machine you are working on.
I’ve never tried any of this, but it should be something of a start on the solution.