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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='color:#1F497D'>Not the answer you are looking for, but some controlling factors are<o:p></o:p></span></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='color:#1F497D'><span style='mso-list:Ignore'>1.<span style='font:7.0pt "Times New Roman"'> </span></span></span><![endif]><span style='color:#1F497D'>The available bandwidth. Since a call takes 30-90K depending on the codec unless you are using a “compression” codec that can reduce this to 5K or so, your number of channels available will be limited by this.<o:p></o:p></span></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='color:#1F497D'><span style='mso-list:Ignore'>2.<span style='font:7.0pt "Times New Roman"'> </span></span></span><![endif]><span style='color:#1F497D'>Call-limit in sip.conf – if you set this low, you are limiting the number of trunks to that (found out hard way)<o:p></o:p></span></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='color:#1F497D'><span style='mso-list:Ignore'>3.<span style='font:7.0pt "Times New Roman"'> </span></span></span><![endif]><span style='color:#1F497D'>Your range in rtp.conf – you get one line for every 4 digits of range, IE 10001-10019 is 5 lines.<o:p></o:p></span></p><p class=MsoListParagraph><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoListParagraph><span style='color:#1F497D'>My best guess is that this isn’t controlled by Asterisk per se like PRI/DAHDI lines, but is determined by the provider and probably not offered as a query function.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Tobias Steen<br><b>Sent:</b> Friday, October 14, 2011 4:17 AM<br><b>To:</b> asterisk-users@lists.digium.com<br><b>Subject:</b> [asterisk-users] Get the total amount of lines/channels for a SIP-trunk?<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span lang=SV>Hello!<o:p></o:p></span></p><p class=MsoNormal><span lang=SV><o:p> </o:p></span></p><p class=MsoNormal>Is it possible for Asterisk to get the total amount of phone lines/channels on a SIP-trunk? Is there some kind of SIP-request to the provider or do I have to call the trunk provider and ask every time?<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>(I want to know the total amount available on the trunk, so “sip show channels” won’t help me out here)<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Best regards<o:p></o:p></p><p class=MsoNormal>Tobias Steen<o:p></o:p></p></div></body></html>