search for: app_pickup

Hi All, I'm using Asterisk 1.2.14 with Wildcard TDM400P. I need app_pickup.so application so I can pickup channel-independent calls from any IP Phone headset. How to compile and install only this application from bristufff package? Thank you in advance, Dominik

...s Zap and Sip. The only snag is that having picked up a call with it, you can't then transfer it on. Taking a dive into app_dial, it looks like when you specify the T option, it does: ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT); Since there's nothing like that in app_pickup, I guess that's why you can't do the transfer (nothing has enabled the flag) Is there any easy way to allow transfers on calls picked up using PickUp? Failing that, is there a way to call ast_set_flag on the caller features from within the Dialplan, or am I going to have to start hacki...

Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm using a Snom 320 in a CAP position and the receptionist wants to do blind transfers. OK, no problem so far. Now she has asked me how to UN-transfer a call, as in, she transfers a call and wants to hook the call back before it connects (she wanted to tell the caller additional information for example) I don't think that this is possible as once my dialplan starts using Dial()

Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add

...gt; (Silly MP3 Application) == Registered application 'MP3Player' [app_nbscat.so] => (Silly NBS Stream Application) == Registered application 'NBScat' [app_parkandannounce.so] => (Call Parking and Announce Application) == Registered application 'ParkAndAnnounce' [app_pickup.so] => (PickUp/PickDown/Steal) == Registered application 'PickDown' == Registered application 'Steal' == Registered application 'PickUp' [app_playback.so] => (Trivial Playback Application) == Registered application 'Playback' [app_privacy.so] => (Requi...

I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When I call from a zap channel or a SIP phone to another SIP phone, then dial *8 from a third SIP phone, I get 503 Service Unavailable on the third phone and I get this at the Asterisk console: Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible... Jun 28 09:01:24 NOTICE[16774]: