similar to: Can you monitor a call via the asterisk speaker system and do a call pickup if you wish

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

Hello. Is this normal. Or does it mean there is a problem ? ------------------------- stop now Beginning asterisk shutdown.... Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Yuck! Error in buffer handling...: Broken pipe Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (0).

Address lookup canonical name asterisk.org. aliases addresses 216.27.40.102 Service scan FTP - 21 Error: TimedOut SMTP - 25 Error: ConnectionRefused HTTP - 80 Error: ConnectionRefused POP3 - 110 Error: TimedOut NNTP - 119 Error: TimedOut digium.com is ok though Address lookup canonical name digium.com. aliases addresses 216.207.245.1 Service scan FTP - 21 Error: TimedOut SMTP - 25

Hello. I can't find a gsm plugin for XMMS. How do Unix, Linux, BSD users listen to gsm samples ? Regards...Martin -- While you don't greatly need the outside world, it's still very reassuring to know that it's still there.

Hello. I'm seeing this at the console. NOTICE[81926]: File chan_sip.c, Line 5119 (handle_request): Registration from '<sip:marrandy@192.168.1.1>' failed for '192.168.1.70' What's this all about ? Regards...Martin -- Osborn's Law: Variables won't; constants aren't.

Hi list, i'm trying to do that iax channels can acces the pickup feature(normaly *8 dialing). But always the iax channel when dial *8, search for the extensi?n *8 on its context. I know i can program the *8 extension with the pickup applicati?n. But its doesn't works for me, becouse i need to pickup some calls comming from IVR's o Queues. And there de exten is no the same as the

Hello all Sorry for the repost, but I believe this is essential for Aastra phones to use directed pickup with BLF in 1.2 (and it's still experimental in 1.4 I think). pickup-mgernoth-2006-07-28.patch.txt If anyone knows otherwise than that this is still required in latest 1.2 or 1.4 please reply with success/failure stories. Any info would be appreciated. We can get directed BLF pickup

Hello. I've seen some mentions of asterisk possibly being used as an inexpensive voicemail attachment to a commercial PBX etc. Does anyone here, have experience of using it in this fashion ? What commercial systems have been successfully attached too ? How is the attachment made ? Analog, digital ? If anyone has successfully accomplished this, I would like to hear the make and model of

Hello. I had grandstream working fine to FWD through my firewall. Now I want it to talk to the asterisk server. Did lots of reading, attempts but I keep getting registration errors even though I can call to/from the sip phone from an analog phone on a tdm400 card. Basically. grandstream = 192.168.1.70 asterisk = 192.168.1.1 The error I see is ;- -- Executing Dial("Zap/2-1",

Hello, ? I'm trying to configure my system such that I have a TCP port listening which will send all data it receives to a serial port. ie:?? tcp/2112? -->? /dev/ttyS0 My goal would be to be able to: Use netcat to create the listen and redirect to a serial port: ? $ nc -l 2112 > /dev/ttyS0 Then in another window, run minicom at /dev/ttyS0 Then in a 3rd window, ? $ telnet localhost

I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible

I'm having trouble with call pickups. Suppose ring group is 100 and has extensions 101 and 102. Someone calls 100, 101 rings and 102 wants to pick the call up. If 102 dials **100, call pickup works. If 102 dials **101, call pickup fails. In my dialplan I have: exten => **101,1,NoOp(pickup extension) exten => **101,n,Pickup(101) exten => **101,n,NoOp(pickup group) exten =>

Hey there list ! I'm receiving negative feedback when people try to pickup another ringing phone by pressing *8 on there own Grandstream device. These are my setting that should make pickup possible : all my sip-clients (Grandstream) have this in their config (sip.conf) : callgroup=1 pickupgroup=1 canreinvite=no qualify=yes So they are all in the same pickupgroup... This the

Hi all, I have a GXP2000 with BLF configured. I follow the configuration guide to enable the pickup cmd as follow and include it under corresponding content. [BLF_group_pickup] exten => _**1XX,1,Pickup(${EXTEN:2}) exten => _**1XX,n,Hangup The I press the single key to pickup the call to extension 100 when there is a call to it. From CLI, I can see it issue **100 to asterisk but failed

Hi, according to the description of Pickup() on page http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup I can use this command to pickup a call at a certain extensions. When I try this with e.g. exten => *8200,1,Pickup(200) Asterisk tells me that the highest value for the Pickup command is 63. Wenn I enter the number of a callgroup instead of an extension, I can pickup the call.

We are installing Aastra phones (480's and 57i's) into a fairly simple asterisk setup. Although call park & pickup work fine using xfer to 700 (to park), dial 701 (to pickup), we are unable to make the park/pickup softkey feature work on the aastra's. Although we've programmed the softkeys per the manuals, they seem to have no effect (just dead). For example, our 57i is

What's the proper syntax for implementing directed call pickup? Running cvs-head from today (9/24/05 including Mark's fixes), and tried: exten => *99,1,Pickup(${EXTEN:3}) but that does not seem to work, and there isn't an example in the configs directory. 'show application pickup' suggests the above should work with our sip phones, but apparently I'm missing

Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first rings at my phone and in the second i can see the callers number. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib

I'm trying to set the Pickup feature. I'm setting my extensions.conf as: exten => _*.,1,Pickup(SIP/${EXTEN:1}) but if, for example, extension 03 is ringing by a call made from extension 01, and I try to pick it up from extension 02 (by dialing *03 from extension 02), I can see in the Asterisk console (Verbosity set to 10): -- Executing Dial("SIP/01-512c",

I have a TDM400 in the server. I want to press **1XX to pickup a call. It is ok if I pickup a call dialled from 1XX to 1YY (internal network call). However, it is failed to pick up a call from PSTN thro' TDM400 card. It seems I can't guess the correct context of it. How can I know the context of the call using CLI? The default context of the TDM400 is from-pstn but pickup still