AWAITING FEEDBACK PIAF not answering calls

[asmith]

Hello Experts,

I have spend days on it but PIAF is not sending out any voice for any incoming calls. I have received successful calls using Google Voice, but when I call this number 0597353070 which is pointing to my PIAF IP (its not registered as a trunk), the call is answered but I do not hear anything. Thank you if you take time to look into it.

Here is version info

───────────────────SYSTEM INFORMATION *VERIFIED*─────────────────────┐
│ Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE │
│ SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE │
│ Fail2ban = ONLINE | Internet = ONLINE | Ip6Tables = ONLINE │
│ Disk Free = ADEQUATE| Mem Free = ADEQUATE| NTPD = ONLINE │
│ SendMail = ONLINE | Samba = OFFLINE | Webmin = ONLINE │
│ Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A │
│ │
│ PIAF Installed Version = 2.0.6.4 under *KVM* │
│ FreePBX Version = 2.10.1.10 │
│ Running Asterisk Version = 1.8.23.1 │
│ Asterisk Source Version = 1.8.23.1 │
│ Dahdi Source Version = 2.7.0.1 │
│ Libpri Source Version = 1.4.14 │
│ IP Address = 10.0.2.85 on eth0 │
│ Operating System = CentOS release 6.4 (Final) │
│ Kernel Version = 2.6.32-358.6.2.el6.i686 - 32 Bit │
│ Incredible Version = 180




Logs are attached.

 

[jeffmac]

According to your log, in each attempt, the inbound call is received and a dial to extension 1000 is attempted, but there is some issue with that extension, as asterisk immediately responds with "Everyone is busy or congested at this time".

So you need to verify the status of your extension 1000.

Jeff

 

[asmith]

Extension 1000 is registered. But whenever I answer call it gets disconnected and caller receives busy tone.

If extension is not registered, I can see Asterisk playing Voice mail messages but caller doesnt hears them.

I figured it was a codecs issue because caller side was sending this in SIP INVITE

a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
m=audio 4770 RTP/AVP 18 18 18 96
c=IN IP4 213.254.211.70
a=rtpmap:18 G729/8000
a=rtpmap:18 G729A/8000
a=rtpmap:18 G729B/8000
a=sendrecv
a=rtpmap:96 telephone-event/8000

So I installed open source g729 codec but still all in vain, no voice. Thanks for taking out time and looking into it.

 

[leemason]

It looks like their could well be a codec incompatibility somewhere as you get this:

Capabilities: us - 0x8010c (ulaw|alaw|g729|h263), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)

Also I don't think your trunk config is setup properly because you also get:
Received incoming SIP connection from unknown peer to 0597353070

Can you post more details like:

1. The service provider you use
2. The trunk config
3. Extension config
4. SIP NAT settings

 

[asmith]

There is no trunk setup like I mentioned. That number is pointing to IP address of asterisk from a hardware voip switch.

Rest of the settings are all correct since Google Voice is configured and its working fine.

 

[rjaiswal]

I'm a bit confused. Without a trunk, how does piaf know that the VoIP gateway is allowed to send sip traffic to it?

 

[rjaiswal]

No matching peer for '059696804' from 'xxx.xxx.xxx.xx:2397'

My understanding is that without a proper authenticated peer, asterisk won't allow the call to connect.

I could be wrong, but I'd try to address this first.

 

[rossiv]

You are correct. You must create a trunk for Asterisk to accept the call.