<div>I really can't believe I'm having this issue and I'm almost sure it's an issue either with my SIP phone or something to do with me. But since I can classify it as a possible bug, I might as well ask you.
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<div>I've got a very simple dialplan going to test some Dial() options.</div>
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<div>300,1,Dial(SIP/josh,20,hH)</div>
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<div>I can place the call on hold, but I can't seem to get Asterisk to notice the 'hH' flag and allow the call to be ended by pressing * (star). Here's a Verbose 3 log from Asterisk:</div>
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<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"> -- Executing [300@internal:1] Dial("SIP/nick-0914fa88", "SIP/josh|20|hH") in new stack
<br> -- Called josh<br> -- SIP/josh-0915e518 is ringing<br> -- SIP/josh-0915e518 answered SIP/nick-0914fa88<br> -- Packet2Packet bridging SIP/nick-0914fa88 and SIP/josh-0915e518<br> -- Started music on hold, class 'default', on SIP/josh-0915e518
<br> -- Stopped music on hold on SIP/josh-0915e518<br> == Spawn extension (internal, 300, 1) exited non-zero on 'SIP/nick-0914fa88'</blockquote></div>
<div><br clear="all">According to the Asterisk Dial() docs, the flags hH should enforce Asterisk staying in line of the stream and Asterisk should be able to notice either side pressing *. But for the moment, pressing * does nothing.
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<div>In the off chance someone wants the Peer information on both clients:</div>
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<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"> Context : internal<br> Subscr.Cont. : <Not set><br> Language : <br> AMA flags : Unknown
<br> Transfer mode: open<br> CallingPres : Presentation Allowed, Not Screened<br> Callgroup : <br> Pickupgroup : <br> Mailbox : <br> VM Extension : asterisk<br> LastMsgsSent : 32767/65535<br> Call limit : 0
<br> Dynamic : Yes<br> Callerid : "" <><br> MaxCallBR : 384 kbps<br> Expire : 3139<br> Insecure : no<br> Nat : Always<br> ACL : No<br> T38 pt UDPTL : No<br> CanReinvite : No
<br> PromiscRedir : No<br> User=Phone : No<br> Video Support: No<br> Trust RPID : No<br> Send RPID : No<br> Subscriptions: Yes<br> Overlap dial : No<br> DTMFmode : rfc2833<br> LastMsg : 0<br> ToHost :
<br> Addr->IP : <a href="http://66.236.37.85">66.236.37.85</a> Port 1262<br> Defaddr->IP : <a href="http://0.0.0.0">0.0.0.0</a> Port 5060<br> Def. Username: nick<br> SIP Options : (none)<br> Codecs : 0x8000e (gsm|ulaw|alaw|h263)
<br> Codec Order : (none)<br> Auto-Framing: No <br> Status : OK (192 ms)<br> Useragent : Grandstream BT120 <a href="http://1.0.8.17">1.0.8.17</a><br> Reg. Contact : <a href="mailto:sip:nick@192.168.2.102">
sip:nick@192.168.2.102</a></blockquote></div>
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<div><br>-- <br>/Nick </div>