[HELP] Call transfer & hold on analog phones


#1

[BACKGROUND]
I’m going to be setting up an asterisk system for one of our remote sales branches, and I just found out that the salesmen will need to use cordless telephones.

I’ve looked at Wi-Fi phones (such as UT Starcomm F1000), but they don’t seem as if they’re really anything much more than amusing gadgets at this point. The best alternative that I’ve found so far is to pick up some ATA devices, and use analog cordless phones.

[QUESTION]
I need to find out how to get Asterisk to let an analog telephone transfer a call to another line and put calls on hold, using the dialing pad or the ‘Flash’ button.


#2

voip-info.org/wiki/view/Aste … p+transfer


#3

It can’t be as simple as that, can it? I’ll try it out in a few miniutes.

Thanks

[quote]From voip-info.org/wiki/view/Aste … p+transfer : How to transfer a call on a phone connected to a ZAP channel

* hook flash (On some phones, press the R button), this puts call 1 on hold (You can try # as well instead of flash)
* dial tone is played
* dial another end point
* talk to that extension
* hook flash again 

Now we are on a 3-way call. At this point you can stay on the call, or if your telephone line allows call transfer, simply hang up to transfer the call. If you do not have Call Transfer on your telephone line, all parties will be disconnected when you hang up. [/quote]


#4

[quote=“mismanccc”]It can’t be as simple as that, can it? I’ll try it out in a few miniutes.

Thanks

[quote]From voip-info.org/wiki/view/Aste … p+transfer : How to transfer a call on a phone connected to a ZAP channel

* hook flash (On some phones, press the R button), this puts call 1 on hold (You can try # as well instead of flash)
* dial tone is played
* dial another end point
* talk to that extension
* hook flash again 

[/quote][/quote]

One thing doesn’t work for me.
When I’m on analog phone (AP) and person from SIP phone (S1) calls me, I pick up and I can talk with him. Then when I decaide to transfer that call to another person on SIP phone (S2) i press flash bottun (S1 goes on hold and I get free line signal) and dial S2 number. When S2 picks up I can hear him but he can’t hear me! Then I press flash again and all three (AP, S1, S2) can hear evrybody.

So, why S2 can’t hear me when S1 is on hold?

Tomislav


#5

The instructions above worked fine for me (Thanks for the help rusty), but I don’t think that my situation is directly comparable to yours. I’m actually not using a zap channel as far as asterisk is concerned. I’m using a SIP extension with an analog telephone on a Granstream Handytone 286 ATA.

They’re not too expensive (around $50 a piece). For me it’s actually cheaper and easier to use the adapters on the analog phones, instead of getting a TDM400P card with FXS modules.

I don’t have the expertise to offer a better suggestion that to try it this way.

Post if you find the answer, though.