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--></style></head><body lang=EN-CA link=blue vlink=purple style='word-wrap:break-word'><div class=WordSection1><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span style='font-size:12.0pt;color:black'>From: </span></b><span style='font-size:12.0pt;color:black'>asterisk-users <asterisk-users-bounces@lists.digium.com> on behalf of "Joshua C. Colp" <jcolp@sangoma.com><br><b>Reply-To: </b>Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b>Date: </b>Wednesday, September 7, 2022 at 10:23 AM<br><b>To: </b>Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b>Subject: </b>Re: [asterisk-users] Channel names with semicolons<o:p></o:p></span></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><div><p class=MsoNormal>On Wed, Sep 7, 2022 at 11:17 AM Antony Stone <<a href="mailto:Antony.Stone@asterisk.open.source.it">Antony.Stone@asterisk.open.source.it</a>> wrote:<o:p></o:p></p></div><div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm'><p class=MsoNormal>On Wednesday 07 September 2022 at 11:44:54, Antony Stone wrote:<br><br>> Hi.<br><br>This is a follow-up to an email I posted earlier today to the list, although I <br>haven't seen it come back yet. If it's under moderation for some reason, I <br>hope some kindly admin will release it :)<o:p></o:p></p></blockquote><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>There's nothing in the moderator queue that I can see.<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm'><p class=MsoNormal><br>> I'm trying to deal with a problem regarding putting a call on hold and then<br>> later resuming it. I am using chan_sip throughout, and Asterisk 16.<br><br><snip detail from previous email><br><br>> The main thing which is puzzling me about this is that I see examples of<br>> both Local/number@context-00000ce9;1 and Local/number@context-00000ce9;2<br>> during the processing of the calls.<br>> <br>> What is the significance of the number following the semi-colon?<br>> <br>> I also see in verbose logging output:<br>> <br>> [2022-09-07 09:37:57.310706] pbx VERBOSE[29148]: dial.c:598 in<br>> handle_frame: Local/number@context-00000ce9;1 answered<br>> <br>> [2022-09-07 09:37:57.310792] pbx VERBOSE[29155][C-00001265]:<br>> bridge_channel.c:2252 in bridge_channel_internal_push_full: Channel<br>> SIP/Trunkname-00002b55 joined 'simple_bridge' basic-bridge <7e260e93-<br>> abd4-48ea-96f1-33601165dba2><br>> <br>> [2022-09-07 09:37:57.310937] pbx VERBOSE[29149][C-00001265]:<br>> bridge_channel.c:2252 in bridge_channel_internal_push_full: Channel<br>> Local/number@context-00000ce9;2 joined 'simple_bridge' basic-bridge<br>> <7e260e93- abd4-48ea-96f1-33601165dba2><br>> <br>> <br>> So, when the channel Local/number@context-00000ce9;1 gets answered, the<br>> result is to bridge the channels Local/number@context-00000ce9;2 and<br>> SIP/Trunkname-00002b55<br><br>I see something very similar in the documentation about local channels at <br><a href="https://wiki.asterisk.org/wiki/display/AST/Using+Callfiles+and+Local+Channels" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Using+Callfiles+and+Local+Channels</a> - <br>there are examples of both devices-ecf0;1 and devices-ecf0;2 but no mention of <br>what the final digit means.<br><br>Can anyone enlighten me please?<o:p></o:p></p></blockquote><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>A single channel can't do two things at once (you can't have a channel talking to Alice while also executing the Voicemail dialplan application for example) - so Local channels solve this by having <o:p></o:p></p><p class=MsoNormal>two independent channels that exchange things back and forth internally. The ;2 leg is the one that gets sent into the dialplan, while the ;1 leg is doing whatever dialed it decides to do with it. If you send audio to ;1 it then pops out of ;2, and vice versa.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>--- <o:p></o:p></p><p class=MsoNormal>Oh I had the channel numbers reversed in my previous example. So yeah, this.<o:p></o:p></p></div></div><div><p class=MsoNormal><o:p> </o:p></p></div></div><p class=MsoNormal><o:p> </o:p></p></div></body></html>