Linksys Pap2-NA - Trixbox the voice doesn't pass through

Asterisk Asterisk Support
1

Hello, I am facing an issue transferring the voice from my Trixbox to the phone behind Linksys PAP2-NA. Phone rings but when picked up nothing can be heard. The caller keep listening to phone ringing like there is no answer.
My configuration is as follows:
Trixbox runs on cloud server (direct IP, no NAT)
Linksys Pap2-NA is behind the NAT (Router Edimax BR-6428nS).
So in general it looks like (Trixbox)------------(Edimax BR-6428nS)—NAT-----(Linksys PAP2)
Both lines in Linksys PAP2 are connected.
First line (port 5060) is connected directly to provider sipme.me, second one (port 5061) is connected to my Trixbox.
When calling the first line through sipme.me everything is just fine. Here’s the log from Linksys PAP2:

[quote]<151>Aug 1 22:15:28 687F74D9E449 [0:0]AUD ALLOC CALL (port=16432)
<151>Aug 1 22:15:28 687F74D9E449 [0:0]RTP Rx Up
<159>Aug 1 22:15:30 687F74D9E449 [0]Off Hook
<151>Aug 1 22:15:30 687F74D9E449 [0]CID interrupted
<151>Aug 1 22:15:30 687F74D9E449 CC:Connected
<151>Aug 1 22:15:30 687F74D9E449 [0:0]ENC INIT 18
<151>Aug 1 22:15:30 687F74D9E449 [0:0]RTP Tx Up (pt=18->5250c83e:6334)
<151>Aug 1 22:15:30 687F74D9E449 [0:0]RTCP Tx Up
<151>Aug 1 22:15:30 687F74D9E449 [0:0]RTP Rx 1st PKT @16432(3)
<151>Aug 1 22:15:30 687F74D9E449 [0:0]DEC INIT 18
<151>Aug 1 22:15:54 687F74D9E449 [0:0]LAT-- 7(3)
<151>Aug 1 22:16:06 687F74D9E449 [0:0]LAT-- 6(3)<159>Aug 1 22:16:10 687F74D9E449 [0]On Hook
<151>Aug 1 22:16:10 687F74D9E449 [0:0]AUD Rel Call
<151>Aug 1 22:16:10 687F74D9E449 DLG Terminated
<151>Aug 1 22:16:10 687F74D9E449 Sess Terminated
<151>Aug 1 22:16:42 687F74D9E449 CC:Clean Up
<151>Aug 1 22:16:42 687F74D9E449 — OBJ POOL STAT —
<151>Aug 1 22:16:42 687F74D9E449 OP:RTPRXB = 96 ( 96 192)
<151>Aug 1 22:16:42 687F74D9E449 OP:RTPREB = 40 ( 40 48)
<151>Aug 1 22:16:42 687F74D9E449 OP:RTPTXB = 64 ( 64 108)
<151>Aug 1 22:16:42 687F74D9E449 OP:TIMEOU = 101 (120 40)
<151>Aug 1 22:16:42 687F74D9E449 OP:SIPCOR = 0 ( 1 28)
<151>Aug 1 22:16:42 687F74D9E449 OP:SIPCTS = 30 ( 32 568)
<151>Aug 1 22:16:42 687F74D9E449 OP:SIPSTS = 31 ( 32 3492)
<151>Aug 1 22:16:42 687F74D9E449 OP:SIPAUS = 6 ( 8 588)
<151>Aug 1 22:16:42 687F74D9E449 OP:SIPDLG = 10 ( 10 140)
<151>Aug 1 22:16:42 687F74D9E449 OP:SIPSES = 12 ( 12 8196)
<151>Aug 1 22:16:42 687F74D9E449 OP:SIPREG = 0 ( 4 448)
<151>Aug 1 22:16:42 687F74D9E449 OP:SIPLIN = 0 ( 2 140)
<151>Aug 1 22:16:42 687F74D9E449 OP:SUBDLG = 2 ( 2 6436)
<151>Aug 1 22:16:42 687F74D9E449 OP:STUNTS = 16 ( 16 68)
[/quote]
When calling through my Trixbox (from another extension or trunk, it doesn’t matter) recepient hears nothing when he picks up the line, caller hears tones like noone picked up the phone. Here’s the log

Edimax doesn’t have any explicit port forwarding, also UPnP module is switched off (after I switched it on, Linksys was unable to register).
I think this is not just a simple issue of Edimax router because call passes through when calling through sipme.me to the very same Edimax / Linksys PAP2-NA, but it fails when calling through my Trixbox.
I have no STUN server configured in my Linksys, it works fine with sipme.me without STUN server.
Any ideas? I do not attach logs from Trixbox, I took a look and found nothing interesting: it just reports that phone is ringing and that’s it.
I can add these logs if that helps. Thanks in advance!

2

At a guess, the router recognizes the 5060 traffic and enables a transparent proxy, but doesn’t recognize the 5061 traffic, so is not translating the SDP.

3

david55,
It’s a great guess, thank you very much. I will try to switch ports between PAP2 lines, so calls from my Asterix will pass through, calls from sipme.me won’t.
There is another fact I haven’t mentioned. Calls can be issued from both lines of PAP2-NA and voice passes Ok. The problem I mentioned exist only when calling from outside to PAP2-NA line.
If your guess is correct, there is no way I can switch more than 1 SIP device behind Edimax router. Am I right?

4

Depends whether the device is NAT aware.

5

I have set my line to Asterisk to be the only one and to allocate port 5060.
The problem remained the same. Caller hears ringing tones even after the phone is picked up.
Here’s the log file:

Any ideas? Is there log anywhere else which can be relevant?

6

Asterisk can provide a lot of logging, including the actual SIP dialogue and how it has interpreted it.

7

I see.
This is a log from Asterisk. I am calling from extension 10000 to extension 52107. As I wrote before, I hear ringing signals on extension 10000, phone rings at extension 52107 but when picked up, nothing can be heard. Ringing signals go on even after phone at 52107 is picked up.

8

That trace shows that SIP/52107 never answered. You will probably have to turn on sip debug to get any more detail.

It also looks like an Asterisk GUI dialplan, rather than a hand-coded one. It is usually better to use a forum for the specific GUI.