No Sound, need help

[savagenoob]

I am running a fresh install of the new GoAutoDial CE 3.0, kernel 2.6.18-348.12.1.el5. This is the second server I have setup, I have read the manual front and back. I can hear "You are currently the only agent in this conference", when I login. I can make outgoing calls, but no sound coming in or out. I have these settings in my carrier screen:

Code: Select all

[switch2voip]
 fromdomain=sip.switch2voip.us
 fromuser=8XXXXXXXXXX
 host=sip.switch2voip.us
 insecure=port,invite
 secret=XXXX
 type=peer
 username=8XXXXXXXXX
 user=8XXXXXXXXXX
 canreinvite=no
 videosupport=no
 disallow=all
allow=gsm
allow=ulaw
allow=alaw
fromuser= {CALLER ID}
port=5060


I have enabled the 3 codecs on both X Lite and Zoiper, using both IAX and SIP.... still no sound. Was wondering if anyone could give me some solid troubleshooting tips on this one....

[williamconley]

if you have a private ip on the server, you need to put in the externip in sip.conf

if you have done that (and reloaded asterisk) and still have no sound or if you do not have a private ip, it will be a firewall related issue. either the sending location cannot get a packet out of their network in your direction or your firewall is stopping the packets from entering your location. one way sound is almost always the firewall.

[savagenoob]

Yeah this office has a robust firewall, may be the issue. So lets say that server IP is 192.168.10.158 and I configure a softphone on 192.168.10.289, could the firewall be blocking the sound ports? I did change the sip.conf as well.

[williamconley]

if the agent is local to the server on a private IP make sure nat=no and that the ip range is defined as local in sip.conf

also be sure the phone is aware that this is a local server connection

your firewall should not even be invoked if these are on the same physical network and 192.168.10.xx subnet.

[savagenoob]

OK, so nat=yes on sip.conf, I changed that. The local ip on the sip.conf shows 192.168.0.0/255.255.0.0 but I dont know how to specify the IP range, which would be 192.168.10.1 - 192.168.10.255. Can you give me an example setting for that?

[williamconley]

192.168.0.0/255.255.0.0 encompasses 192.168.10.1 - 192.168.10.255 (it covers 192.168.anything)

[savagenoob]

ok, then it should be working. still no sound. checked the firewall logs and no blocked packets or ports. I am out of ideas.

[savagenoob]

OK, so I brought in another working server into the network to test, and it is getting sound fine. I even copied and pasted the sip.conf from the working one to the non-working one and still no sound. I am completely baffled.

[savagenoob]

Maybe my CLI log will give some insight to the sound issue? Can someone tell me how to debug this situation?

Code: Select all

[Jul 17 12:37:37]     -- Accepting AUTHENTICATED call from 192.168.10.242:
       > requested format = gsm,
       > requested prefs = (),
       > actual format = ulaw,
       > host prefs = (ulaw|gsm),
       > priority = mine
[Jul 17 12:37:37]     -- Executing [919165454545@default:1] AGI("IAX2/7001-12256", "agi://127.0.0.1:4577/call_log") in new stack
[Jul 17 12:37:37]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul 17 12:37:37]     -- Executing [9191654545450@default:2] Dial("IAX2/7001-12256", "SIP/Switch2Voip/191654545450||tTor") in new stack
[Jul 17 12:37:37]     -- Called Switch2Voip/191654545450
[Jul 17 12:37:42]     -- SIP/Switch2Voip-00000000 is making progress passing it to IAX2/7001-12256
[Jul 17 12:37:42]     -- SIP/Switch2Voip-00000000 answered IAX2/7001-12256
[Jul 17 12:37:53]     -- Executing [h@default:1] DeadAGI("IAX2/7001-12256", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----11") in new stack
[Jul 17 12:37:53]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----11 completed, returning 0
[Jul 17 12:37:53]   == Spawn extension (default, 9191654545450, 2) exited non-zero on 'IAX2/7001-12256'
[Jul 17 12:37:53]     -- Hungup 'IAX2/7001-12256'
[Jul 17 12:38:01]   == Parsing '/etc/asterisk/manager.conf': [Jul 17 12:38:01] Found
[Jul 17 12:38:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 12:38:02]   == Parsing '/etc/asterisk/manager.conf': [Jul 17 12:38:02] Found
[Jul 17 12:38:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 12:38:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 12:38:03]   == Manager 'sendcron' logged off from 127.0.0.1


[nigel_john]

[Jul 17 12:37:53] -- Executing [h@default:1] DeadAGI("IAX2/7001-12256", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----11") in new stack
[Jul 17 12:37:53] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -16-----11 completed, returning 0
[Jul 17 12:37:53] == Spawn extension (default, 9191654545450, 2) exited non-zero on 'IAX2/7001-12256'
[Jul 17 12:37:53] -- Hungup 'IAX2/7001-12256'

I think the call is hanging up in one second.
And the reason looks to be your dialer context.

[williamconley]

I'd like a bit of clarity:

IAX calls on a NON-LOCAL IP address, but the previous ip range you showed was 192.168 which is a local IP.

However, if this call terminates at the moment of sound initiation, it is likely a handshake related issue such as a missing codec (for instance: if your system claims g729 but does not have it installed ... the call will terminate at the moment sound should start). this would show in IAX debug.