HELLO & WELCOME

[SimonB]

Hi everyone,

1st post here so I thought a Hello would be a good start. I've been a definite "voip end user" for that last 5 years and have set up UK DIDs for my family who live abroad and saved them and us hundreds of pounds in calls over that time. I've used a SPA3102 and a Gigaset N300a - both of which are great bits of kit.

In the next 6 months we will be giving up our BT landline and moving to super duper FTTH broadband. I'm here so I can learn and set up a home voip setup that rocks.

I registered here so I could ask (and hopefully help) questions about freepbx on the pi, but I realy need to understand some underlying SIP concepts first. I'm a IT sysadmin (once upon a time I was a unix admin) so you can probably get as tecnical as you need to without issue.

So my 1st question is this: Once a incoming call is connected through an asterisk server can you then "break out" to another extension? I know the receiveing user on an asterisk extenstion can transfer (blind or attendeded) but can you configure it so the asterisk server will listen for DTMF on an incoming trunk and act upon it?

Alternatively can you get a SPA3102 or OBi110 to do the same trick? i.e. can you allow an incoming (authenticated) user to transfer themselves away from the middle of a VMail recording to another extension - possibly an asterisk server?

Regards

SimonB

[WelshPaul]

Hi everyone,

1st post here so I thought a Hello would be a good start. I've been a definite "voip end user" for that last 5 years and have set up UK DIDs for my family who live abroad and saved them and us hundreds of pounds in calls over that time. I've used a SPA3102 and a Gigaset N300a - both of which are great bits of kit.

Hello and welcome to the forums Simon. :)

It sure is great to hear that just like me, you're saving huge amounts of money using VoIP and the SPA3102 sure is a great bit's of kit. Unfortunately for me I have not had the privilege to try out the Gigaset N300a but it has some fantastic reviews. :-D

In the next 6 months we will be giving up our BT landline and moving to super duper FTTH broadband. I'm here so I can learn and set up a home voip setup that rocks.

FTTH or FTTP isn't available in my area even though I live 3/4 of a mile from the exchange (as the crow flies).

The best I can get is 17Mb which sucks! Thank goodness for Virgin Media's 152MB cable broadband. :laugh:

I registered here so I could ask (and hopefully help) questions about freepbx on the pi, but I realy need to understand some underlying SIP concepts first. I'm a IT sysadmin (once upon a time I was a unix admin) so you can probably get as tecnical as you need to without issue.

No worries, still learning a lot myself. Anyone who has any first hand experience or technical knowledge is a great asset to any VoIP forum. :)

So my 1st question is this: Once a incoming call is connected through an asterisk server can you then "break out" to another extension? I know the receiveing user on an asterisk extenstion can transfer (blind or attendeded) but can you configure it so the asterisk server will listen for DTMF on an incoming trunk and act upon it?

Am I correct in thinking you want the person who actually calls in to a line/extension hosted on the PBX to be able to dial 200 while in an active call and have that current call terminated and in turn have extension 200 ring? (simply by detecting DTMF tones during the live call)

Alternatively can you get a SPA3102 or OBi110 to do the same trick? i.e. can you allow an incoming (authenticated) user to transfer themselves away from the middle of a VMail recording to another extension - possibly an asterisk server?

Surely this would be done by the PBX and not an ATA... What do you want to happen to the original call? Have it placed on hold, terminated or initiate a three way call?

[SimonB]

So my 1st question is this: Once a incoming call is connected through an asterisk server can you then "break out" to another extension? I know the receiveing user on an asterisk extenstion can transfer (blind or attendeded) but can you configure it so the asterisk server will listen for DTMF on an incoming trunk and act upon it?

Am I correct in thinking you want the person who actually calls in to a line/extension hosted on the PBX to be able to dial 200 while in an active call and have that current call terminated and in turn have extension 200 ring? (simply by detecting DTMF tones during the live call)

Yes. I want the asterisk server to normally pass through the call to the DECT base station and it's built-in answermachine. But for advanced users who don't want to leave a message be able to break out into an asterisk IVR during the built in answer machine message playback and go do clever stuff like call/sms a mobile phone instead.

Alternatively can you get a SPA3102 or OBi110 to do the same trick? i.e. can you allow an incoming (authenticated) user to transfer themselves away from the middle of a VMail recording to another extension - possibly an asterisk server?

Surely this would be done by the PBX and not an ATA... What do you want to happen to the original call? Have it placed on hold, terminated or initiate a three way call?

This is an alternative to the above idea that doesn't have the asterisk server (Pi model B+) in the call path, only the ATA, but have the ATA do the transfer mid call. I'm a little nervous of depending on asterisk 100%. It's a bit too easy to break when I mess with it.

The critical question which I don't understand is this: Is it possible in the SIP/asterisk world to divert calls mid call if you are the incoming user? Or once an endpoint has answered is that the end of the dialplan and you can't interact with the channel any longer?

The other aspect of this is where are the DTMF tones interpreted once a call has connected? in the channel or on the endpoint. I have to consider any * codes from my hosting provider as well I think as the asterisk ones and the endpoint (answermachine) codes to control playback and retreival.

Thanks

Simon

[WelshPaul]

So my 1st question is this: Once a incoming call is connected through an asterisk server can you then "break out" to another extension? I know the receiveing user on an asterisk extenstion can transfer (blind or attendeded) but can you configure it so the asterisk server will listen for DTMF on an incoming trunk and act upon it?

Am I correct in thinking you want the person who actually calls in to a line/extension hosted on the PBX to be able to dial 200 while in an active call and have that current call terminated and in turn have extension 200 ring? (simply by detecting DTMF tones during the live call)

Yes. I want the asterisk server to normally pass through the call to the DECT base station and it's built-in answermachine. But for advanced users who don't want to leave a message be able to break out into an asterisk IVR during the built in answer machine message playback and go do clever stuff like call/sms a mobile phone instead.

Yes, that certainly should be possible.

There's a whole host of 'dial commands' and apparently T and t are the ones of interest.

• T: Allow the calling user to transfer the call by hitting the blind xfer keys (features.conf). Does not affect transfers initiated through other methods.
  If you have set the variable GOTO_ON_BLINDXFR then the transferrer will be sent to the context|exten|pri (you can use ^ to represent | to avoid escapes), example: SetVar(GOTO_ON_BLINDXFR=woohoo^s^1); works with both t and T
• t: Allow the called user to transfer the call by hitting the blind xfer keys (features.conf) Does not affect transfers initiated through other methods.
  If you have set the variable GOTO_ON_BLINDXFR then the transferrer will be sent to the context|exten|pri (you can use ^ to represent | to avoid escapes), example: SetVar(GOTO_ON_BLINDXFR=woohoo^s^1); works with both t and T

A friend of mine has setup attended transfer by code, so, when a call comes from an onsite PBX he is a member of, he can dial #7, then make another call. He can then dial ** to cancel the xfer or hangup to complete.

I'm sure it can work the other way around too.

Alternatively can you get a SPA3102 or OBi110 to do the same trick? i.e. can you allow an incoming (authenticated) user to transfer themselves away from the middle of a VMail recording to another extension - possibly an asterisk server?

Surely this would be done by the PBX and not an ATA... What do you want to happen to the original call? Have it placed on hold, terminated or initiate a three way call?

This is an alternative to the above idea that doesn't have the asterisk server (Pi model B+) in the call path, only the ATA, but have the ATA do the transfer mid call. I'm a little nervous of depending on asterisk 100%. It's a bit too easy to break when I mess with it.

The critical question which I don't understand is this: Is it possible in the SIP/asterisk world to divert calls mid call if you are the incoming user? Or once an endpoint has answered is that the end of the dialplan and you can't interact with the channel any longer?

The other aspect of this is where are the DTMF tones interpreted once a call has connected? in the channel or on the endpoint. I have to consider any * codes from my hosting provider as well I think as the asterisk ones and the endpoint (answermachine) codes to control playback and retreival.

Thanks

Simon

Neither the SPA3102, OBixxx or any other ATA that I know of is capable of doing what you require on it's own, not without help from a PBX.

I am digging a little deeper on this so will post back once I have clarified things. :)

[WelshPaul]

Yes. I want the asterisk server to normally pass through the call to the DECT base station and it's built-in answermachine. But for advanced users who don't want to leave a message be able to break out into an asterisk IVR during the built in answer machine message playback and go do clever stuff like call/sms a mobile phone instead.

Ok an update...

Looks like this isn't possible after all but there maybe a work around...

Instead of using a DECT phones built in answering machine could you not use the PBX's Voicemail? That way you could enable operator assistance on the voicemail system and set the operator number to be an IVR (potentially a hidden one).

You could then hit zero during voicemail and then use the IVR to go somewhere else.

[SimonB]

Thanks Paul,

This is what I though was happening as well. I don't think I can use the t and T dial commands as they allow the callee to transfer the caller and the caller to transfer the callee respectively, not the caller to transfer themselves.

I suspect your plan may be the neatest and bypass the phone's answerphone and use voicemail instead. I could hack the phones to speed dial the voicemail access if I was getting advanced. the idea is that non office users like physical buttons and voicemail generally doesn't work like this.

Simon