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</style><font face="arial" size="2">Guess I got where your confusiion lies.... you DON'T set up the ALL contexts the users will have acces to in sip.conf, in this file you will only set 1 single context. All other contexts you want the users to have acces to, you will have to add them in the extensions.conf<br /> <br /> So what you have to do is the following: <br /> -user 2092, set it the createmenu context in sip .conf<br /> - in extensions.conf under the "createmenu" context add as many "include" lines as you need the user to have acces to.<br /> <br /> it should look something like:<br /> ; sip.conf<br /> <br /> </font><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">[2092]<br /> <span class="GramE">type=</span>friend<br /> <span class="GramE">username=</span>2092<br /> <span class="GramE">canreinvite=</span>no<br /> <span class="GramE">context</span>= createmenu<span style=""></span></span></font><br /> <font face="arial" size="2"><br /> ; extensions.conf<br /> <br />[createmenu]<br /> ...<br /> ...<br /> include => outgoing<br /> include => some other context<br /> <br /> <br /> hope this helps<br /> <br />Alyed<br /><br /></font><br /><font face="Tahoma, Arial, Sans-Serif" size="2"><hr align="center" size="2" width="100%" /></font> <!--[if gte mso 9]><xml> Clean DocumentEmail </xml><![endif]--> <!--[if gte mso 10]> <![endif]--><!--[if gte mso 9]><xml> </xml><![endif]--><!--[if gte mso 9]><xml> </xml><![endif]--> <div class="Section1"> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">Hi,</span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">Thanks for the reply.</span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">What happens is that all users are registered with SER (a sip proxy). I have set SER up so when a user dials ‘0’ followed by a pstn number it will be forwarded to asterisk which will forward the call to a third party pstn gateway. I also use asterisk so that when a user who is registered with ser doesn’t answer (sending a 408 cancel response) or is busy (sending a 486 busy response) that the call is forwarded to asterisk voicemail. So therefore at the moment I have a user ‘2092’ which registers with ser and uses the ‘outgoing’ context in asterisk for pstn access and accesses their voicemail mailbox through the default context. </span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">Now I also set it up so that if a user registered with SER dials 20005 it should forwards to asterisk. This should call the context ‘createmenu’ which creates an IVR menu.</span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">What I’m confused about is this. I created a user 20005 in sip.conf using context=createmenu. This wasn’t working. After reading your post I realized my mistake was that the context that is being called is that of the caller i.e. 2092 as opposed to whom the call is directed at i.e. 20005. Therefore when I changed the context of 2092 to ‘createmenu’ it worked.</span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">BUT how can I set up my sip.conf so that 2092 can use the default, outgoing and createmenu contexts depending on the correct scenario? If someone who is also using SER has any comments, I’d also really appreciate it.</span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p> <p class="MsoNormal"><span class="GramE"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">i.e.</span></font></span><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"></span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">[300]<br /> <span class="GramE">type=</span>friend<br /> <span class="GramE">username=</span>300<br /> <span class="GramE">canreinvite=</span>no<br /> <span class="GramE">context</span>= WHAT GOES HERE??<span style=""> </span><span style=""> </span>//createmenu calls the IVR but then outgoing pstn calls don’t work, outgoing allows pstn calls but then I can’t create a menu etc etc<br /> <span class="GramE">insecure=</span>very<br /> <span class="GramE">;callerid</span>= "voicemail user 1" <300><br /> <span class="GramE">host=</span>dynamic<br /> <span class="GramE">nat=</span>yes<br /> <span class="GramE">dtmfmode=</span>INFO<br /> <span class="GramE">mailbox=</span>300<br /> <span class="GramE">disallow=</span>all<br /> <span class="GramE">allow=</span>alaw<br /> <span class="GramE">allow=</span>ulaw<br /> <span class="GramE">allow=</span>g723.1</span></font></p> <p class="MsoNormal"><span class="GramE"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">allow=</span></font></span><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">g729</span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">Many thanks,</span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;">Aisling.</span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p> <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p> <p class="MsoNormal" style="margin-left: 36pt;"><font face="Tahoma" size="2"><span style="font-size: 10pt; font-family: Tahoma;">-----Original Message-----<br /> <b><span style="font-weight: bold;">From:</span></b> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b><span style="font-weight: bold;">On Behalf Of </span></b>Alyed Tzompa<br /> <b><span style="font-weight: bold;">Sent:</span></b> 04 January 2006 00:28<br /> <b><span style="font-weight: bold;">To:</span></b> asterisk-users@lists.digium.com<br /> <b><span style="font-weight: bold;">Subject:</span></b> re: [Asterisk-Users] confusion about contexts</span></font></p> <p class="MsoNormal" style="margin-left: 36pt;"><font face="Times New Roman" size="3"><span style="font-size: 12pt;"> </span></font></p> <p class="MsoNormal" style="margin-left: 36pt;"><font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;">I'm a bit confused on how you get your calls to Asterisk, what I mean is: are you phoning into asterisk via a sip user? in this <br /> case, which one?, if not is it iax or though a zap channel?<br /> <br /> anyway, here some tips:<br /> <br /> For your first problem it seems it has to do with what I pointed above, check that the user which is dialing into asterisk has <br /> the correct context (context=create-menu) with at least type= peer<br /> <br /> also don't have to retype the allow=codec, disallow=codec, dtmfmode=x for every user, just set it in the general context in <br /> sip.conf<br /> <br /> your second problem think it has to do once again with the firts thing above, and regarding the retyping, I'm afaid I don't know <br /> any other way than writing those lines again and again for everyuser. Maybe someone else out there knows someting else that can help.<br /> <br /> Don't set many "outgoing" context for every user in sip.conf!!!!! just set one and point all users to that one. If you need your <br /> user to have acces to other contexts just add <br /> include => your_context<br /> at the end of whatever context you want (btw can add more than one inlcude's )<br /> <br /> <br /> Alyed </span></font><br /> -------------------------------------------<br /> Hi,<br /> <br /> Hope someone can help me-Asterisk isn’t behaving as I would expect<br /> and I think it’s down to my contexts.<br /> <br /> There are two things I can’t fathom.<br /> <br /> Firstly I want to record an IVR and so have created a user 20005 and<br /> a context called createmenu. I am using SER in front of asterisk so I<br /> changed the ser.cfg so that if the user dialled this number it<br /> forwards to asterisk. This works fine. The problem is when the invite<br /> reaches my asterisk box, asterisk uses the wrong context. It appears<br /> to call the “outgoing” context which is the context used to route<br /> calls to my pstn gateway provider. It then trys to execute a “Dial”<br /> command for 20005 which isn’t supposed to happen.<br /> <br /> Secondly SER uses Asterisk for voicemail if a phone doesn’t answer<br /> after a certain period of time or is busy. This works fine but I have<br /> to create an entry for every user in extensions.conf under the<br /> [default] context. Can I create a generic entry which would also work<br /> to shorten the config file?...Also if I change this and out all the<br /> entries under a context “voicemail” it doesn’t work….I have to keep<br /> it in default…This must obviously be something got to do with<br /> Asterisk finding the contexts.<br /> <br /> I am confused as to how you apply multiple contexts to one user. At<br /> the moment nearly each user (besides 20005 and 1234) has a context of<br /> ‘outgoing’ in sip.conf. This is so that they can make outgoing pstn<br /> calls…But what if I needed them to use another context in other<br /> situations?...I’m just confused as to what context should be applied.<br /> <br /> I have included the relevant parts of my sip.conf and extensions.conf<br /> below. I would appreciate any advice as to why these issues are<br /> occurring.<br /> <br /> Many thanks,<br /> Aisling.<br /> <br /> ;sip.conf<br /> [general]<br /> <br /> bindport=5064<br /> bindaddr=0.0.0.0<br /> disallow=all<br /> allow=ulaw<br /> allow=alaw<br /> allow=gsm<br /> srvlookup=yes<br /> canreinvite=no;<br /> autocreatepeer=yes<br /> nat=yes<br /> dtmfmode=info<br /> insecure=very<br /> registerattempts=0<br /> <br /> register => username:password@sip.blueface.ie/1234<br /> <br /> ;To receive incoming calls specify this and replace<br /> "yourcontext-pstn" for your dial plan<br /> <br /> [blueface-in]<br /> type=peer<br /> host=sip.blueface.ie<br /> context=pstn<br /> <br /> [1234]<br /> type=friend<br /> username=1234<br /> canreinvite=no<br /> context=pstn<br /> insecure=very<br /> ;callerid= "Ais" <1234><br /> host=dynamic<br /> nat=yes<br /> dtmfmode=INFO<br /> mailbox=1234<br /> disallow=all<br /> allow=alaw<br /> allow=ulaw<br /> allow=g723.1<br /> allow=g729<br /> <br /> ;added below line(s) for BLUEFACE conf<br /> ;To make outgoing calls specify this block<br /> <br /> [blueface-out]<br /> type=peer<br /> host=sip.blueface.ie<br /> username=username<br /> secret=password<br /> <br /> [20005]<br /> type=friend<br /> username=20005<br /> canreinvite=no<br /> context=createmenu<br /> insecure=very<br /> ;callerid= "Ais" <20005><br /> host=dynamic<br /> nat=yes<br /> dtmfmode=INFO<br /> mailbox=20005<br /> disallow=all<br /> allow=alaw<br /> allow=ulaw<br /> allow=g723.1<br /> allow=g729<br /> <br /> [300]<br /> type=friend<br /> username=300<br /> canreinvite=no<br /> context=outgoing<br /> insecure=very<br /> ;callerid= "voicemail user 1" <300><br /> host=dynamic<br /> nat=yes<br /> dtmfmode=INFO<br /> mailbox=300<br /> disallow=all<br /> allow=alaw<br /> allow=ulaw<br /> allow=g723.1<br /> allow=g729<br /> <br /> extensions.conf<br /> [general]<br /> static=yes<br /> writeprotect = yes<br /> <br /> [createmenu]<br /> exten => 20005,1,Wait(2)<br /> exten => 20005,2,Record(/tmp/asterisk-recording:gsm)<br /> exten => 20005,3,Wait(2)<br /> exten => 20005,4,Playback(/tmp/asterisk-recording)<br /> exten => 20005,5,wait92)<br /> exten => 20005,6,Hangup<br /> <br /> ;specify context for receiving incoming calls<br /> [pstn]<br /> ;Note this is just an example there are infinite different ways to<br /> handle the incoming call.<br /> ;exten => 1234, 1,Wait(1)<br /> ;exten => 1234, 2,Playback(beep)<br /> ;exten => 1234, 3,Hangup<br /> exten => 1234, 1, Dial<br /> <br /> (SIP/2092@seraddress) ; 1234 is the contact extension, default<br /> contact extension is "s"<br /> <br /> ;exten => 2092,1,Answer()<br /> ;exten => 2092,2,Playback(welcome)<br /> ;exten => 2092,3,Background(menu)<br /> ;exten => 1,1,Dial($316)<br /> ;exten => 2,1,Dial($314)<br /> <br /> [outgoing]<br /> ; Dial the Blue Face Speaking Clock<br /> exten => 300,1,Dial(SIP/300@blueface-out)<br /> exten => 300,2,Hangup<br /> <br /> ;Send PSTN calls to Blue Face<br /> exten => _X.,1,Dial(SIP/${EXTEN}@blueface-out)<br /> exten => _X.,2,Hangup<br /> <br /> [default]<br /> <br /> exten => 300, 1,Dial(SIP/300,20)<br /> exten => 300, 2,Voicemail(u300)<br /> exten => 300, 102,Voicemail(b300)<br /> exten => 300, 103,Hangup<br /> <br /> exten => 301, 1,Dial(SIP/301,20)<br /> exten => 301, 2,Voicemail(u301)<br /> exten => 301, 102,Voicemail(b301)<br /> exten => 301, 103,Hangup <br /> <br /> etc etc<br /> <br /> <br /> <br /> -------------------Legal Disclaimer---------------------------------------<br /> <br /> The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.<br /> <br /> _______________________________________________<br /> --Bandwidth and Colocation provided by Easynews.com --<br /> <br /> Asterisk-Users mailing list<br /> To UNSUBSCRIBE or update options visit:<br /> http://lists.digium.com/mailman/listinfo/asterisk-users<br /> <br /> -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. </p> </div>-------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.<br />