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I am using asterisk 1.8 on ubuntu 9.10. I want to ask if it's possible to invite more than 2 sip clients to a conference room using call files? A sip client wants to create/ or has a conference room and would like to invite other 3 persons. how would the dialplan look like? When you invite persons to a conference room it means you call them?

How to do that with asterisk? I read about call files and extensions.conf dialplan. I have no ideea how to do this. I meed to be able to invite in a conference room more than 2 people or at least 2 people.

user1222905
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1 Answers1

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As per your scenario instead of .call file I suggest you to use originate command (introduced in asterisk 1.6.2). you have to create dialplan consisting on following two parts / extension

  1. conference this context / extension will be responsible to add caller into selected conference room

  2. incoming in incoming dialplan caller will be asked to enter number of required participants and dialplan will call each number one by one with originate command while providing conference extension as destination, like

    same => n,Originate(SIP/number,exten,conference_context,room_number,1)

and when done with invites the original caller will also be sent to conference with goto command,

Nasir Iqbal
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  • can you give me a complete example for the following: sip client1 has conference room 1234 and he wants to invite sip2 , sip 3 and sip 4. Thx. – user1222905 Mar 14 '12 at 10:49
  • I don't think that there is any complication, if you please mentioned the point where you need clarification then may i help – Nasir Iqbal Mar 14 '12 at 10:59
  • I don't know how to do it. in extension.conf I have [conference] exten => 1234,1,Meetme(1234); AFter that I'll have [internal] exten => ?,?,Originate(SIP/sip2 SIP/sip3 SIP/sip4,exten,conference,1234,1)? – user1222905 Mar 14 '12 at 11:54
  • in above message please note for **one by one** so it will be exten => ?,1,Originate(SIP/sip2,exten,conference,1234,1) exten => ?,2,Originate(SIP/sip3,exten,conference,1234,1) exten => ?,3,Originate(SIP/sip4,exten,conference,1234,1) – Nasir Iqbal Mar 14 '12 at 13:07
  • and after that where do i need the goto command? – user1222905 Mar 14 '12 at 13:25
  • exten => ?,4,Goto(conference,1234,1) – Nasir Iqbal Mar 14 '12 at 13:31
  • is not working. the other clients do not receive rining tones. – user1222905 Mar 14 '12 at 13:34
  • any error at cli? please check your dialplan and sip extensions config, registration etc.. and also verify if that extensions are reachable by dialing from other sip client or from asterisk cli – Nasir Iqbal Mar 14 '12 at 13:41
  • no . there are no errors on cli – user1222905 Mar 15 '12 at 09:08
  • you have not mentioned any thing about, if you have created / registered those SIP extension? and can you dial them? anyway it is simple if you can dial those client directly / CLI then you can also dial them from dialplan with originate and there is no science here – Nasir Iqbal Mar 17 '12 at 11:52
  • Cant the context [incoming] you suggested be included in a more wide range context such as a phone context that would contain incoming and outgoing rules? Just in case some of the invitees might be local or remote ? – OlaB Jun 20 '16 at 16:57
  • @OlaB I don't see any issue with it, only you should be clear about dialplan flow and logic. – Nasir Iqbal Jun 21 '16 at 19:07
  • @nasirbest Awesome, very intelligent logic you came up with here, really did help.I only had one issue which I don't know if its related to asterisk or my hardware,when the extensions are originated, the first one which gives a prompt of "you are the only one in this conference" and is internal to the manager that is sending the commands seems to hangup sooner after the other stations that are external are originated, I started by originating the internal as the first station to join the conference. – OlaB Jun 22 '16 at 11:52
  • I am thinking it has to do with the audio files not being able to be found by asterisk on that extension which causes a drop call according to the research I did, but there was no error logging in the debugs on why it drops that extension. Another question I would like to add if you might be of help is this, is there a way to set asynchronous mode for the manager that originates the extensions within the dial-plan?, that way it doesn't have to be sequential and the manger doesn't have to wait for responses before proceeding with the other originations. – OlaB Jun 22 '16 at 11:53
  • I know I can set asynchronous mode to yes if I were to be using telnet to process the requests via asterisk manager API, but I don't know how I would do that within a dial plan for example like embedding them into your solution strings as above.Thanks! – OlaB Jun 22 '16 at 11:53
  • @OlaB can't help with hangup, however regarding asynchronous, you can statisfy originate application by using local channels, which will use answer and then dial commands respectively to dial original contact. – Nasir Iqbal Jun 23 '16 at 04:05