Question

Asterisk (Trixbox) 1.4.2.4 n-way calling / 3-way calling / conferencing

Asked by: jetli87

Hi Gurus,

I need help configuring n-way calling or 3-way calling.

I'm using Linksys SPA941 phones...When I'm talking to a user and try to conference in another caller, I press "conference" on the phone, dial the number and when the caller picks up and i press "join," instead of conferencing, it's transfering the call.

What do I need to do to get this working?

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Asked On
2009-09-18 at 19:43:06ID24745034
Tags

Asterisk 1.4.2.4 Trixbox voip

Topics

Asterisk Open Source Telephony

,

Voice Over IP

,

IP Telephony

Participating Experts
2
Points
500
Comments
21

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Answers

 

by: grbladesPosted on 2009-09-19 at 08:20:58ID: 25373302

Have you tried pressing the CONF button again after the 2nd person answers as this is what the instruction manual sais to do.
If that doesnt work then I would try a firmware upgrade on the phone as it is a feature built into the phone itself and doesnt rely on any specific trixbox configuration.

 

by: koszegiPosted on 2009-09-21 at 06:29:05ID: 25382595

Try updated the firmware.

Checkout
http://www.voip-info.org/wiki/view/SPA-941

Good Luck!

 

by: jetli87Posted on 2009-09-22 at 15:24:34ID: 25398157

I'm running firmware version:  5.1.8

I did exactly as you mentioned and from the manual, but when i click "join" the call just gets transferred and it hangs up my extension.

 

by: grbladesPosted on 2009-09-23 at 09:59:39ID: 25405355

The manual I have seen makes no mention of the join. It sais to press the CONF button again to join the conference.

 

by: jetli87Posted on 2009-09-23 at 10:39:18ID: 25405771

I tried both ways...

this is my exact steps:

1) I call "bill" and "bill" pickups
2) I press "conf" and get dial tone and dial "julie"
3) "julie" pickups and then i press "conf" again
4) now "bill" hears MOH.
5) I also see a "join" option which i press since nothing else happens
6) the phone transfers "bill" to "julie" and I'm disconnected.

???

 

by: grbladesPosted on 2009-09-23 at 10:57:07ID: 25405947

Sorry no furthur ideas. All the copies of the manuals and instructions independant websites have posted say that your procedure is correct and as soon as you press conf the second time you should be in a 3 way conference.

I would contact linksys technical support as it may be a fault with the phone.

 

by: jetli87Posted on 2009-09-23 at 11:10:06ID: 25406094

Well i tried with multiple Linksys phones (running latest firmware) and Polycom phones with no luck.

The thing is, I'm using Trixbox 2.4, which is my first time.

When i was running Asterisk 1.2.4 a few years back, I never had an issue with this feature which is way i thought it might be a setting with Trixbox.

 

by: koszegiPosted on 2009-09-23 at 11:16:19ID: 25406162

I would reset the phone to factory default.  if the problem still exist I would update the firmware.  If that doesn't fix the problem, then it is a problem with the way your Asterisk box is configure.  Check you sip.conf to make sure your account for the phone is configured correctly.  Make sure it has the correct options. example canreinvite=yes, type=peer

Checkout:
http://www.voip-info.org/wiki/view/SPA-941

http://www.voip-info.org/wiki/view/PBX+Three+Way+Calling

http://www.voip-info.org/wiki/view/Asterisk+PBX+functions

 

by: grbladesPosted on 2009-09-23 at 11:27:25ID: 25406271

3 way calling is entirely a phone feature. There are two lines active and it just just performs a mini conference internally.
It is entirely independant of any conference feature in trixbox.

The only possible thing I can think of that could possibly interfere is any configuration related to the maximum number of calls permitted in trixbox (which might be related to disabling call waiting). If this is set to 1 or call waiting disabled then trixbox may deny the phone from switching the original line from MOH back to itself while the 2nd line is active.

 

by: koszegiPosted on 2009-09-23 at 11:32:03ID: 25406326

that is why I gave him the link on top and told him to check the type and canreinvite.  I don't believe the can reinvite will help but being a peer will allow the phones to conferrence multiple call legs.

 

by: jetli87Posted on 2009-09-23 at 13:11:01ID: 25407383

Per Koszegi,

I changed settings for SIP to canreinvite=yes & type=peer but still no luck.

Also, call waiting has been enabled.

 

by: koszegiPosted on 2009-09-23 at 13:13:03ID: 25407405

can you post the sip.conf setting for that phone?

 

by: jetli87Posted on 2009-09-23 at 13:15:49ID: 25407434

[203]
deny=0.0.0.0/0.0.0.0
disallow=all
type=peer
secret=203
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=203@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/203
context=from-internal
canreinvite=yes
callgroup=
callerid=device <203>
allow=g729
accountcode=
call-limit=50
                                              
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by: koszegiPosted on 2009-09-23 at 13:23:43ID: 25407518

I see your problem.  You are using g729 and this can cause calls to fail if the user attempts to use three-way calling, or place simultaneous calls on both lines of a two-line device, and G.729 is the only allowed codec. Try changing to alaw or ulaw.

Let me know what happen.

 

by: koszegiPosted on 2009-09-23 at 13:25:16ID: 25407529

This is a license codec and may allow a limited amout of channel per device.  This could be your problem.

 

by: koszegiPosted on 2009-09-23 at 13:28:20ID: 25407571

also refer to this link for more info

http://www.voip-info.org/wiki/view/ITU+G.729

 

by: koszegiPosted on 2009-09-23 at 13:31:24ID: 25407603

Here your answer from the link above about G729.

It offers toll quality speech at a reasonably low bit rate of 8Kbps. However, it is a rather "costly" codec in terms of CPU processing time, therefore some VoIP phones and adapters (notably the Linksys/Sipura/Cisco brands) can only handle one G.729 call (channel) at a time. This can cause calls to fail if the user attempts to use three-way calling, or place simultaneous calls on both lines of a two-line device, and G.729 is the only allowed codec.

 

by: koszegiPosted on 2009-09-23 at 13:46:54ID: 25407764

What I would do is either use another codec or use g729 with another codec like ulaw or alaw.

 

by: jetli87Posted on 2009-09-23 at 13:53:54ID: 25407833

thanks, that was the issue.


But now what should i do if i want to use 3-way calling and g729 codec since it works so well with remote phones.

 

by: koszegiPosted on 2009-09-23 at 14:07:45ID: 25407962

By the way that's another question altogether.

you need to buy a license for g729 and install it.  
goto  http://www.voip-info.org/wiki/view/Asterisk+G.729+Licensing

Before you just go out and buy a new license, I would find out if the processor on the linksys phone can support multple calls with g729.  I ould recommend enable another codec with g729 for that device instead of spending the money for a solution that is hard to implement and at the end may not provide you a acceptable preformance.

 

by: jetli87Posted on 2009-09-23 at 14:12:21ID: 25407997

I'm actually using the open-source version of the codec.

I'll repost for my follow-up question.

20120131-EE-VQP-002

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