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33 posts

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Topic # 22871 10-Jun-2008 19:57
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Hi,

I have Trixbox 2.6.0.0 installed on a server (this is my first time playing with Trixbox) with the VFX voip service configured. It does not want to work with call forward or Follow Me (effectively the same as call forward).

In the first instance, can anyone confirm for me that Trixbox 2.6.0.0 call forward is working for them? I seeing some weird behaviour and am starting to think that perhaps this version is just broke!

Second, can anyone help me with my problem? Here's the problem, and some things I have tried to establish what is going on.

My system: Trixbox 2.6.0.0, VFX trunk installed, two extensions - 200 and 201, inbound and outbound routes setup. Call forward turned on to ring an external number (my cell) when a call comes into ext 200.

1. I can make an outgoing call, and receive incoming calls on both extensions. All works fine.

2. I can make a call from ext 201 to ext 200, and it gets forwarded to the external number perfectly.

3. I can make an outgoing call on extns 200 and 201 simultaneously and it works (although audio quality degrades)

4. I can make an incoming call to extn 201 and an outgoing call from ext 200 to my external cellphone number.

5. HOWEVER - when I make an incoming call to ext 200, the call gets forwarded to my external number but when I answer it I get no audio on either end!

Signalling seems to be working fine, ie the external phone rings. Audio is working given the other tests I have done (to the best of my knowledge, I'm no expert on this.) It just will not work, which is really frustrating.

I started out testing on 2Talk, with the plan to switch to VFX to test calls, then Fusion once I was confident it was all working. I first observed the problem with 2Talk trunk configured, and hoped that switching to VFX would fix it. But it hasn't. Any thoughts or ideas? Other things I could try?

Many thanks
Aaron

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33 posts

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  Reply # 137154 10-Jun-2008 20:53
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Well, I've been looking for a solution to this for days, and within 30 minutes of posting my message I found a work around that appears to work - for me at least. This appears to be a known issue (well at least it affects a fair few people) on Trixbox CE for the last few versions.

The answer is to set it up as a Follow Me, with your forwarded phone number in the follow me list, and then turn on confirmation. On your external number, when you asnwer you'll be prompted with a prompt to press '1' to accept incoming call. Press 1 and then you will have audio.

It worked for me, but is not really ideal. I would still be interested in any better solution, and an explanation of why it happens, from anyone.

Hope this helps others.

Aaron

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  Reply # 137218 11-Jun-2008 05:49
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The issue here will be in the signaling actually and the negotiation of the Media Ports used for the Audio, it will depend on how you are passing the transferred call of as well, using refer or invite, we would need to have some call details to offer an explanation.




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  Reply # 137227 11-Jun-2008 08:29
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Also check that you are not trying to be the media bridge, I'm not sure what the commands are, if I stumble across them I will let you know.

Cheers

Josh




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  Reply # 137229 11-Jun-2008 08:40
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Did you try adding progressinband=yes which seems to have solved the issue for others? This issue only affects calls forwarded to a SIP channel and doesn't occur when forwared via a Zaptel channel. This will generate ringing as described here

sip.conf: progressinband


 progressinband=yes

When "RING" event is requested, always send 180 Ringing (if it hasn't been sent yet) followed by 183 Session Progress and in-band audio

 progressinband=no

Send 180 Ringing if 183 has not yet been sent establishing audio path. If audio path is established already (with 183) then send in-band ringing (this is the way asterisk historically behaved because of buggy phones like polycom)

 progressinband=never

Whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behaviour of Asterisk.



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  Reply # 137265 11-Jun-2008 11:21
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Actually, I was just going to post a new message this morning about the progressinband=yes command. I found that late last night.

What I did was add the progressinband command into sip.conf and confirmed that this solved the problem. Was feeling a little nervous about leaving it in there, since the file says that FreePBX can overwrite it at any time. So then I added it into the peer section of the VFX trunk and it also appears to work fine. So that is good to know.

Now the down side - maybe minor, but from what I can tell this means that when you make a call, Asterisk is generating the ring tone you hear on the line itself rather than allowing the remote end to generate the sound. It puts more load on my Asterisk box, and I noticed that when I was forwarding calls and therefore have two active calls the ringing sound started to skip a little and degraded in audio quality.

Maverick, I would be interested in finding a better solution. If you're able to offer any advice, or structions on what call info you would require to help me, please let me know.

I'm also still interested in hearing whether this is affecting any others of you here in NZ. I found some posts about it on the Trixbox forums, so I assume some of you SHOULD be affected. Hey, misery shared is misery halved Cry

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  Reply # 137267 11-Jun-2008 11:25
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I have never used call forward so it's not something I have experienced. I might have a play myself when I get some time, probably won't be for a few days.



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  Reply # 137269 11-Jun-2008 11:31
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Why would you not? I think it is one of the coolest features of running your own home PBX. Laughing Have my home phone ring my cellphone and my wifes cellphone as well, and choose where I want to pick it up. The follow me feature is great!

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  Reply # 137282 11-Jun-2008 11:58
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Probably because I get maybe 2-3 calls per week maximum on my 04 number vs probably 60-70 on my mobile. If I am at work my E65 is connected via Wifi to my Asterisk box anyway which means I can answer the call if somebody did ring.

Everybody I know uses mobiles these days, PSTN numbers are largely irrelevant.

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