<div>After a Good Call from a PSTN phone if I do a "sip prune realtime peer 9013XX9XX8" (9013XX9XX8 being the phone number of the Agent/Member) then I can call the number again and not get the issue. So this has something to do with the stuff that is put in my peer table after a call.</div>
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<div>Any ideas? </div>
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<div class="gmail_quote">On Tue, Jun 14, 2011 at 5:18 PM, Duane Larson <span dir="ltr"><<a href="mailto:duane.larson@gmail.com">duane.larson@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div>One more piece to add. I had mentioned before that I could get a call from a PSTN user to work the first time. So here is all the output of a Good call from a PSTN user after I have performed a "RELOAD" on asterisks CLI</div>
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<div><a href="http://pastebin.com/9RSvQsmN" target="_blank">http://pastebin.com/9RSvQsmN</a></div>
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<div>And when the caller or agent hangs this call up all calls from the PSTN afterward get put in the queue automatically and the agent never gets called.<br><br></div>
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<div class="h5">
<div class="gmail_quote">On Tue, Jun 14, 2011 at 4:37 PM, Duane Larson <span dir="ltr"><<a href="mailto:duane.larson@gmail.com" target="_blank">duane.larson@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div>Ok. Something isn't right. With a user that is local to my SIP user database calls the queue phone number everything works without issue. It is when a remote user (like someone from the PSTN) calls the queue phone number that the caller gets put into the queue and the agent/member doesn't receive the call. I have captured debugs from OpenSIPS and Asterisk and I can't really see any difference. I also executed the commands you told me where I could. Here are the debugs</div>
<div> </div>
<div>Good call from local SIP user to Queue</div>
<div>LocalUser -> OpenSIPSProxy -> Asterisk (then asterisk calls the agent/member) -> OpenSIPSProxy -> Agent</div>
<div><a href="http://pastebin.com/Fa9y3CXQ" target="_blank">http://pastebin.com/Fa9y3CXQ</a></div>
<div> </div>
<div> </div>
<div> </div>
<div>Bad call from PSTN Caller to Queue</div>
<div>PSTN Gatway -> OpenSIPSB2BUA -> OpenSIPSProxy -> Asterisk (then asterisk doesn't call Agent/Member for some reason)</div>
<div><a href="http://pastebin.com/VBA9nGAs" target="_blank">http://pastebin.com/VBA9nGAs</a></div>
<div> </div>
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<div>Thanks for looking at this. Currently this happens every time. Any call from a local user gets put in queue and agent is called right away, but any call from PSTN user gets put in queue and agent isn't called but the agent shows as </div>
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<div>Asterisk18*CLI> queue show<br><a href="http://irock.com/" target="_blank">irock.com</a> has 1 calls (max unlimited) in 'ringall' strategy (6s holdtime, 131s talktime), W:0, C:12, A:15, SL:0.0% within 0s<br>
Members: <br> SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls (last was 1991 secs ago)<br> Callers: <br> 1. SIP/9013XX9XX8-0000002d (wait: 0:02, prio: 0)</div>
<div> </div>
<div>When it is a good call and I do "queue show" I see this</div>
<div>Asterisk18*CLI> queue show<br><a href="http://irock.com/" target="_blank">irock.com</a> has 0 calls (max unlimited) in 'ringall' strategy (5s holdtime, 131s talktime), W:0, C:12, A:16, SL:0.0% within 0s<br>
Members: <br> SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls (last was 2079 secs ago)<br> No Callers</div>
<div> </div>
<div><strong>How come with the Bad Call the Agent/Member shows up in a "queue show" as being a Member and a Caller???</strong></div>
<div><br><br> </div>
<div class="gmail_quote">
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<div></div>
<div>On Mon, Jun 13, 2011 at 11:58 PM, Satish Barot <span dir="ltr"><<a href="mailto:satish4asterisk@gmail.com" target="_blank">satish4asterisk@gmail.com</a>></span> wrote:<br></div></div>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div>
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<div><br>I am not sure but seems like Agent channel not being released from Asterisk.<br><br>Next time when this happens, try 'core show channels' to check whether Agent channel is released or not.<br><font color="#888888"><br>
[SATISH]</font>
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<div></div>
<div><br><br>
<div class="gmail_quote">On Mon, Jun 13, 2011 at 9:12 PM, Duane Larson <span dir="ltr"><<a href="mailto:duane.larson@gmail.com" target="_blank">duane.larson@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Yesterday I rebooted the server and it seems to be working again. Not sure what the reboot might have changed. Hopefully it doesn't happen again but I can't be sure. To answer your question I have the sip.conf in my mysql database and in MySQL I have callcounter set to yes. I don't have a column of 'qualify' in my database for the sip users. For my config I am using OpenSIPS as the register and proxy. Asterisk is only used for voicemail and ACD/Hunt groups.
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<div class="gmail_quote">On Mon, Jun 13, 2011 at 12:53 AM, Satish Barot <span dir="ltr"><<a href="mailto:satish4asterisk@gmail.com" target="_blank">satish4asterisk@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote"><br>Provide the entry for Agent SIP/9013XX9XX8 along with parameters 'callcounter' and 'qualify' from sip.conf. <br>
<br>Also provide CLI outputs of 'core show channels',sip show peers' and 'queue show' when...<br><br>(1)First caller enters the Queue<br>(2)First caller gets connected with Agent<br>(3)First caller gets disconnected from Agent<br>
(4)Second caller enters the Queue<br><br>You may have sequences changed for step no 3 and 4 in your scenario.<br><br><br>[SATISH]</blockquote></div></div></div></blockquote></div></div></div><br></div></div>
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</div></blockquote></div><br><br clear="all"><br>-- <br>--<br>*--*--*--*--*--*<br>Duane<br>*--*--*--*--*--*<br><font color="#888888">--<br></font></blockquote></div><br><br clear="all"><br></div></div>-- <br>--<br>*--*--*--*--*--*<br>
Duane<br>*--*--*--*--*--*<br><font color="#888888">--<br></font></blockquote></div><br><br clear="all"><br>-- <br>--<br>*--*--*--*--*--*<br>Duane<br>*--*--*--*--*--*<br>--<br>