Thanks for the reply!<br><br>I've played around with R to solve this (probably should have mentioned that), however I wasn't able to make it work. The message is still played (this message is from the provider). It will move to the next line in the dialplan, but as soon as users hear the message they hang up.<br>
<br>Since the progress code comes before actual audio is played to the caller there has to be a way of catching this and dealing with it in the dialplan, but nothing I've tried so far works.<br><br><div class="gmail_quote">
On Wed, Oct 29, 2008 at 12:25 AM, Juan Rodríguez <span dir="ltr"><<a href="mailto:jerdguez@gmail.com">jerdguez@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Try using a R or r on the Dial command, the R option is better for you in my opinion.<div><br></div><div>i.e <span style="border-collapse: collapse;">Dial(Zap/G2/1${EXTEN},30,R) or </span><span style="border-collapse: collapse;">Dial(Zap/G2/1${EXTEN}|30|R)</span></div>
<div><span style="border-collapse: collapse;"><br></span></div><div><span style="border-collapse: collapse;">The R option is going to generate a ring tone when the callee indicates ringing and is going wait for an Answer. As Progress is just for early media, you wont get that message. </span></div>
<div><span style="border-collapse: collapse;"><br></span></div><div><span style="border-collapse: collapse;">For more info on the Dial command see:</span></div><div><span style="border-collapse: collapse;"><br>
</span></div><div><span style="border-collapse: collapse;"><a href="http://www.voip-info.org/wiki-Asterisk+cmd+Dial" target="_blank">http://www.voip-info.org/wiki-Asterisk+cmd+Dial</a><br></span></div><div><br></div>
<div><br></div><div><br><div class="gmail_quote"><div><div></div><div class="Wj3C7c">On Tue, Oct 28, 2008 at 6:56 PM, arkda <span dir="ltr"><<a href="mailto:thrills3k3r@gmail.com" target="_blank">thrills3k3r@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div class="Wj3C7c">
Some additional information.<br><br>I played with ${DIALEDSTATUS} in place of ${HANGUPCAUSE} and got an unusual result:<br><br>[Oct 28 16:50:54] WARNING[17503]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission NzJlOWI0NjI5NTMwMmEwZTExYzZiZTM5YWY4MDk0MzA. for seqno 2 (Critical Response)<br>
<br>This occurs about a second after the user hangs up on the error message being played from the provider. I have a feeling it's trying to execute the next step in the dialplan but unable since the caller hung up.<br>
<br>Thoughts, criticism, insults all welcome!<div><div></div><div><br><br><div class="gmail_quote">On Tue, Oct 28, 2008 at 12:53 PM, arkda <span dir="ltr"><<a href="mailto:thrills3k3r@gmail.com" target="_blank">thrills3k3r@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi,<br><br>I've ran into an issue with a PRI provider in a major metropolitan area that I haven't needed to deal with before and I was hoping someone might have some insight on how to handle this within the Asterisk dialplan.<br>
<br>At this location users can't always tell if a number is long distance or not (there are a lot of area codes and prefixes in the vicinity). Additionally, users are required by the provider to dial the full 10 digit number even if a call is local since a local call could be for a few different area codes and prefixes. The problem is the provider requires a 1 in front of the number for long distance calls, but errors out if the call has a 1 in front and the call is local.<br>
<br>As a result, users are complaining that they are constantly having to redial with or without the 1. I've tracked down this behavior when a call fails:<br><br> -- Executing [5551515121@internal:1] Set("SIP/user9-b696fb58", "GROUP(default)=dialpool") in new stack<br>
-- Executing [5551515121@internal:2] GotoIf("SIP/user9-b696fb58", "1?5") in new stack<br> -- Goto (internal,5551515121,5)<br> -- Executing [5551515121@internal:5] Set("SIP/user9-b696fb58", "GROUP(default)=dialpool") in new stack<br>
-- Executing [5551515121@internal:6] Answer("SIP/user9-b696fb58", "") in new stack<br> -- Executing [5551515121@internal:7] Set("SIP/user9-b696fb58", "CALLERID(num)=5552223333") in new stack<br>
-- Executing [5551515121@internal:8] Set("SIP/user9-b696fb58", "CALLERID(name)=HiThere") in new stack<br> -- Executing [5551515121@internal:9] NoOp("SIP/user9-b696fb58", "--out the pri--") in new stack<br>
-- Executing [5551515121@internal:10] Dial("SIP/user9-b696fb58", "Zap/G2/15551515121") in new stack<br> -- Requested transfer capability: 0x00 - SPEECH<br> -- Called G2/15551515121<br> -- Zap/22-1 is proceeding passing it to SIP/user9-b696fb58<br>
-- PROGRESS with cause code 31 received<br> -- Zap/22-1 is making progress passing it to SIP/user9-b696fb58<br> -- Hungup 'Zap/22-1'<br> == Spawn extension (internal, 5551515121, 10) exited non-zero on 'SIP/user9-b696fb58'<br>
<br>The above call was a call that is considered local by the provider. The caller is then redirected to a message (by the provider) saying 'You do not need to dial a one or zero...' and the message repeats indefinitely.<br>
<br>I'd like to figure out how to handle this in the dial plan so users do not even know anything happened. To test to see if I could stop the call progress and reroute it I've tried this so far:<br><br>exten => _NXXXXXXXXX,1,Set(GROUP(default)=dialpool)<br>
exten => _NXXXXXXXXX,2,GotoIf($[${GROUP_COUNT(priout@default)}<19]?5)<br>exten => _NXXXXXXXXX,3,GotoIf($[${GROUP_COUNT(congest@default)}>18]?BLOCK)<br>exten => _NXXXXXXXXX,4,NoOp<br>exten => _NXXXXXXXXX,5,Set(GROUP(default)=dialpool)<br>
exten => _NXXXXXXXXX,6,Answer()<br>exten => _NXXXXXXXXX,7,Set(CALLERID(num)=${CLR})<br>exten => _NXXXXXXXXX,8,Set(CALLERID(name)=HiThere)<br>exten => _NXXXXXXXXX,9,NoOp(--out the pri--)<br>; Primary Dialout<br>
exten => _NXXXXXXXXX,10,Dial(Zap/G2/1${EXTEN})<br>exten => _NXXXXXXXXX,11,GotoIf,($[${HANGUPCAUSE} = 31]?YAY)<br>exten => _NXXXXXXXXX,12,Hangup()<br>; Call limiter<br>exten => _NXXXXXXXXX,n(BLOCK),Answer()<br>
exten => _NXXXXXXXXX,n(BLOCK),Playback(all-circuits-busy-now)<br>exten => _NXXXXXXXXX,n(BLOCK),Playback(pls-try-call-later)<br>exten => _NXXXXXXXXX,n(BLOCK),Hangup()<br>; 1 tester<br>exten => _NXXXXXXXXX,n(YAY),Answer()<br>
exten => _NXXXXXXXXX,n(YAY),Playback(beep)<br>exten => _NXXXXXXXXX,n(YAY),Hangup()<br><br>It doesn't work. The user simply hangs up when the message is heard and the next line in the dialplan isn't followed. How can I detect that a call has received a progress code 31 then reroute it to another extension? From what I found on <a href="http://voip-info.org" target="_blank">voip-info.org</a> ( <a href="http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+hangupcause" target="_blank">http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+hangupcause</a> ) this should work. What am I missing?<br>
<br>The server is running Asterisk <a href="http://1.4.21.1" target="_blank">1.4.21.1</a>, zaptel 1.4.11, libpri 1.4.5, compiled from source.<br><br>Thanks in advance!<br>
</blockquote></div><br>
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