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Asterisk conference calling (w/ Trixbox - freePBX)
Posted: Tue Feb 06, 2007 7:11 pm
by DaveQB
Hi all,
I finally got my Trixbox set up and loving Asterisk so far. I am keen to do conference calling. The page on the web GU looks straight forward, but how does one ring in to the conference ? It seems to get an extension, so if someone is external, how do they get access to it ?
Googling didn't want to reveal much, but I will persevere.
Posted: Fri Feb 09, 2007 9:36 pm
by DaveQB
bump
I thought the show used Asterisks to create conferences ?
Do you use parking to move incoming calls to the conference ?
Posted: Sat Feb 10, 2007 7:26 pm
by Karl
Under inbound routes conference should be a choice.
Or you could have a phone ring then transfer the person into the conference.
Also you could setup voice prompts. Like press 3 to enter the confernce. You need to enable the IVR module to configure that.
Posted: Sat Feb 10, 2007 10:38 pm
by DaveQB
Thanx for that Karl
It makes sense.
I will give your suggestions a run today.
Posted: Sun Feb 11, 2007 12:16 am
by DaveQB
Ok, I see now that if a conference is set up its an option in Incoming Routes.
So looking at it and from what you said Karl I need to either a) set up a menu system for all incoming calls. Not really what I want. Or b) transfer incoming calls to the conference number. Which sounds better as they wont be that common.
So to do the latter...
Take the first call, transfer it to the conference and let them sit there talking to them self while I wait for other callers. then when I have caught and transferred all the expecting calls to the conference, I then dial the conference ext number myself ?
Posted: Mon Feb 12, 2007 12:56 am
by DaveQB
I guess a third way would be to have people register their ATA/SIP/soft phone straight into my server. And they would just be another extension and cost nothing.
Not practical for relatives that know nothing about technology
Posted: Mon Feb 12, 2007 12:57 am
by DaveQB
Oh PS.
One would just have to open and forward port 5060 to get external access to Asterisk ?
Posted: Mon Feb 12, 2007 8:05 am
by Karl
I believe it is port 5060 UDP and ports 10000 - 20000. I think those are UDP also. 10000 - 20000 is used for the audio streams.
Posted: Thu Feb 15, 2007 3:55 am
by davijordan
Is there a good generic voice modem that will work as an fxo modem?
Posted: Thu Feb 15, 2007 5:25 am
by DaveQB
Can I tack onto this thread...
How do you transfer a call ? Googling that doesn't provide much, strange.
Is it # and then the extension ? Not working here....
Posted: Thu Feb 15, 2007 8:17 am
by Linc
DaveQB wrote:Can I tack onto this thread...
How do you transfer a call ? Googling that doesn't provide much, strange.
Is it # and then the extension ? Not working here....
Yup! Someone calls you. Answer and while you are on the phone press # and the extension and that's all she wrote.
Posted: Thu Feb 15, 2007 8:34 am
by Karl
DaveQB wrote:Can I tack onto this thread...
How do you transfer a call ? Googling that doesn't provide much, strange.
Is it # and then the extension ? Not working here....
By default it will only work for a incoming call. If you want it to work when you call out you can try going under General Settings. Where it says dial command add a T. If you mouse over "Asterisk Dial command options" it will give you a list of options that you can put in.
Posted: Thu Feb 15, 2007 8:40 am
by Karl
davijordan wrote:Is there a good generic voice modem that will work as an fxo modem?
If your looking for something cheap, look for a X100P on ebay. They're usually around $10. A lot of people have echo issues with those cards though.
Your best bet is to get something like a Sipura SPA-3000.
Posted: Thu Feb 15, 2007 1:51 pm
by DaveQB
Karl wrote:DaveQB wrote:Can I tack onto this thread...
How do you transfer a call ? Googling that doesn't provide much, strange.
Is it # and then the extension ? Not working here....
By default it will only work for a incoming call. If you want it to work when you call out you can try going under General Settings. Where it says dial command add a T. If you mouse over "Asterisk Dial command options" it will give you a list of options that you can put in.
Thanks alot Karl. Just added that. I'll give it a spin when someone else is awake here. [5:50am atm
]
Posted: Fri Feb 16, 2007 7:34 pm
by DaveQB
Hmmm notworkng
Here is the full log output while trying to hit #211 on an extension to transfer the call to extension 211
Code: Select all
Feb 18 01:27:28 DEBUG[23251] channel.c: Got DTMF on channel (SIP/201-08e6dff8)
Feb 18 01:27:28 DEBUG[23251] channel.c: Bridge stops bridging channels SIP/09436
711-08e68ab8 and SIP/201-08e6dff8
Feb 18 01:27:28 DEBUG[23251] res_features.c: Feature interpret: chan=SIP/0943671
1-08e68ab8, peer=SIP/201-08e6dff8, sense=2, features=2
Feb 18 01:27:28 DEBUG[23251] res_features.c: Set time limit to 500
Feb 18 01:27:28 DEBUG[23251] channel.c: Nobody there, continuing...
Feb 18 01:27:28 DEBUG[23251] channel.c: Bridge stops bridging channels SIP/09436
711-08e68ab8 and SIP/201-08e6dff8
Feb 18 01:27:28 DEBUG[23251] res_features.c: Timed out for feature!
Feb 18 01:27:29 DEBUG[23251] channel.c: Got DTMF on channel (SIP/201-08e6dff8)
Feb 18 01:27:29 DEBUG[23251] channel.c: Bridge stops bridging channels SIP/09436
711-08e68ab8 and SIP/201-08e6dff8
Feb 18 01:27:29 DEBUG[23251] res_features.c: Feature interpret: chan=SIP/0943671
1-08e68ab8, peer=SIP/201-08e6dff8, sense=2, features=2
Feb 18 01:27:29 DEBUG[9040] chan_sip.c: Stopping retransmission on '650191b10240
da9f23bc6937371a8fee@10.1.1.127' of Request 102: Match Found
Feb 18 01:27:29 DEBUG[23251] channel.c: Got DTMF on channel (SIP/201-08e6dff8)
Feb 18 01:27:29 DEBUG[23251] channel.c: Bridge stops bridging channels SIP/09436
711-08e68ab8 and SIP/201-08e6dff8
Feb 18 01:27:29 DEBUG[23251] res_features.c: Feature interpret: chan=SIP/0943671
1-08e68ab8, peer=SIP/201-08e6dff8, sense=2, features=2
Feb 18 01:27:30 DEBUG[23251] channel.c: Got DTMF on channel (SIP/201-08e6dff8)
Feb 18 01:27:30 DEBUG[23251] channel.c: Bridge stops bridging channels SIP/09436
711-08e68ab8 and SIP/201-08e6dff8
Feb 18 01:27:30 DEBUG[23251] res_features.c: Feature interpret: chan=SIP/0943671
1-08e68ab8, peer=SIP/201-08e6dff8, sense=2, features=2
Feb 18 01:27:32 DEBUG[9040] chan_sip.c: Stopping retransmission on '027891d9653b
014b164b2a023441c0c2@10.1.1.127' of Request 102: Match Found
Feb 18 01:27:41 DEBUG[23251] channel.c: Didn't get a frame from channel: SIP/094
User didn't get a dial tone once they hit #