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.<br>
<br>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.<br><br>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?<br>
<br>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.<br>
Here is the debug output just at the point where it happens.<br><br> -- AGI Script dialparties.agi completed, returning 0<br> -- Executing [s@macro-dial:7] Dial("SIP/302-08221a38", "SIP/301||tr") in new stack<br>
-- Called 301<br>Home*CLI> <br><--- Transmitting (NAT) to <a href="http://192.168.0.100:5060">192.168.0.100:5060</a> ---><br>SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK443855200;received=192.168.0.100;rport=5060<br>
From: <<a href="mailto:sip%3A302@192.168.0.1">sip:302@192.168.0.1</a>>;tag=1257222779<br>To: <<a href="mailto:sip%3A301@192.168.0.1">sip:301@192.168.0.1</a>>;tag=as201c8013<br>Call-ID: <a href="mailto:979693319-5060-5@192.168.0.100">979693319-5060-5@192.168.0.100</a><br>
CSeq: 41 INVITE<br>User-Agent: FPBX-2.4.0(1.4.19.1)<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <<a href="mailto:sip%3A301@192.168.0.1">sip:301@192.168.0.1</a>><br>
Content-Length: 0<br><br><br><------------><br> -- SIP/301-0822de30 is circuit-busy<br> == Everyone is busy/congested at this time (1:0/1/0)<br> -- Executing [s@macro-dial:8] Set("SIP/302-08221a38", "DIALSTATUS=CONGESTION") in new stack<br>
-- Executing [s@macro-exten-vm:10] Set("SIP/302-08221a38", "SV_DIALSTATUS=CONGESTION") in new stack<br> -- Executing [s@macro-exten-vm:11] GosubIf("SIP/302-08221a38", "0?docfu|1") in new stack<br>
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/302-08221a38", "0?docfb|1") in new stack<br> -- Executing [s@macro-exten-vm:13] Set("SIP/302-08221a38", "DIALSTATUS=CONGESTION") in new stack<br>
-- Executing [s@macro-exten-vm:14] NoOp("SIP/302-08221a38", "Voicemail is novm") in new stack<br> -- Executing [s@macro-exten-vm:15] GotoIf("SIP/302-08221a38", "1?s-CONGESTION|1") in new stack<br>
-- Goto (macro-exten-vm,s-CONGESTION,1)<br> -- Executing [s-CONGESTION@macro-exten-vm:1] PlayTones("SIP/302-08221a38", "congestion") in new stack<br>Audio is at 192.168.0.1 port 10162<br>Adding codec 0x100 (g729) to SDP<br clear="all">
<br>-- <br>Easy, fast GUI development.<br><a href="http://PerlQt.wikidot.com">http://PerlQt.wikidot.com</a><br>