[asterisk-users] Congestion outbound only with ATA boxes

[Royce Souther]

I have an Asterisk server it runs great with SIP phones, soft SIP phones
(twinkle) and a soft SIP phone app on my Android phone but I am having
problems getting two ATA boxes working. I have a Linksys PAP2T, it is
unlocked and I have used them before with no problems. I was able to
receive calls with from any local SIP phone or from my Link2VoIP connection
via the Internet but it could not call out. It could not call out to the
Link2VoIP or any of the SIP phones. I spent a lot of time going over the
configureation for this Asterisk server and the settings in the Linksys
PAP2T box but could not get it to work. I removed the Linksys PAP2T and
replaced it with an HT503 because I read a lot of good recommendations for
this device. It seems to have almost the same problem. I say almost because
when the Linksys would get congestion I would hear the Asterisk recording
tell me "All circuits are busy now, good-bye" but the HT503 only gets a
busy tone.

All the SIP phones can call out no problem but these two ATA boxes that I
am trying to use the FXS ports to connect old analog POTS phones to are not
working.

I have turned on the debug in Asterisk and can see the point where I get
congestion but I don't know how to make Asterisk give me more details as to
why I am getting congestion. Can anyone help me to get more details about
this problem?

I traced the debug from a working SIP phone as it makes an outgoing call
and from the HT503 as it tries to make a call. Everything is identical
right up to the point where the HT503 gets a congestion instruction from
the Asterisk server.
Here is the debug output just at the point where it happens.

    -- AGI Script dialparties.agi completed, returning 0
    -- Executing [s at macro-dial:7] Dial("SIP/302-08221a38", "SIP/301||tr")
in new stack
    -- Called 301
Home*CLI>
<--- Transmitting (NAT) to 192.168.0.100:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.100:5060
;branch=z9hG4bK443855200;received=192.168.0.100;rport=5060
From: <sip:302 at 192.168.0.1>;tag=1257222779
To: <sip:301 at 192.168.0.1>;tag=as201c8013
Call-ID: 979693319-5060-5 at 192.168.0.100
CSeq: 41 INVITE
User-Agent: FPBX-2.4.0(1.4.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:301 at 192.168.0.1>
Content-Length: 0


<------------>
    -- SIP/301-0822de30 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s at macro-dial:8] Set("SIP/302-08221a38",
"DIALSTATUS=CONGESTION") in new stack
    -- Executing [s at macro-exten-vm:10] Set("SIP/302-08221a38",
"SV_DIALSTATUS=CONGESTION") in new stack
    -- Executing [s at macro-exten-vm:11] GosubIf("SIP/302-08221a38",
"0?docfu|1") in new stack
    -- Executing [s at macro-exten-vm:12] GosubIf("SIP/302-08221a38",
"0?docfb|1") in new stack
    -- Executing [s at macro-exten-vm:13] Set("SIP/302-08221a38",
"DIALSTATUS=CONGESTION") in new stack
    -- Executing [s at macro-exten-vm:14] NoOp("SIP/302-08221a38", "Voicemail
is novm") in new stack
    -- Executing [s at macro-exten-vm:15] GotoIf("SIP/302-08221a38",
"1?s-CONGESTION|1") in new stack
    -- Goto (macro-exten-vm,s-CONGESTION,1)
    -- Executing [s-CONGESTION at macro-exten-vm:1]
PlayTones("SIP/302-08221a38", "congestion") in new stack
Audio is at 192.168.0.1 port 10162
Adding codec 0x100 (g729) to SDP

-- 
Easy, fast GUI development.
http://PerlQt.wikidot.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120131/60921c35/attachment.htm>