search for: callpickup

So I'm having trouble getting call-pickup working. Got a few different SIP phones (cisco 7940's and SPA-841s) all with pickupgroup=0 in sip.conf. I can't seem to get it working. This *is* possible from SIP phones, right? Do I need to add anything to my dial-plan? -- Paul A. Dugas Dugas Enterprises, LLC paul@dugasenterprises.com 1711 Indian Ridge Drive

hi folks, Im pretty sure this has been covered before but I just wasnt able to find any answer. Im having troubles with the call pickup feature, is just not working for me. whenever I press *8 or 200 or anyother. nothing happens and sometimes I also get" nothing to pickup". I have read this might be a bug although I havent found any patch for it. does anyone have any ideas? Im using

Does anyone have use for a broadcast/annouce app? I wrote SystemAnnounce which will play a specified file to all active channels (in an UP or bridged state). This was originally to tell users to get off the system, but there are several other uses... I also wrote a new CallPickup and CallPark app, both of which work more as expected (supply actual extension numbers, etc). Let me know if there is any interest and I'll post the code. Justin ____________________________________________________________________________________ Looking for last minute shopping deals...

...son ST2030S) is configured to send an INVITE message whenever a BLF is pressed while blinking. The INVITE is build with the extension number (attached to the BLF that was blinking and pressed) prepended by a customizable Starcode sequence. For instance, when extension 142 is ringing (and *8 is the CallPickup Starcode), pressing the associated BLF would make the hardphone send an INVITE to extension *8142. With 1.6.1.11, this INVITE is treated as if "call picker" had dialed *8142 : it enters dialplan with appropriate extension and context. With 1.6.2.0, this INVITE fails with message like: c...

...you, but I need you guys to help me. Here are my questions: 1. Music On Hold, it doesn't play any sound on the parked call or hold call. But if I do ps-ax, it shows mpg123 .....( I forgot the exact line). I'm using slackware 9.1 2. I have fxs 3 port, and in my zapata.conf I have included callpickup=1-4, but I can't pick up the call. 3. Supposed I have 2 fxo cards (right now I have one already) and 3 fxs, and one of the fxo will have two phone (running pararell), is there any way for * to: a. It always dial the first fxo, if the fxo is busy or is being used (have other people conversat...

Hello, all. I've got a problem where we set up call pickup for a customer. If the Bob's extension rings and Bob is in Jim's office, Bob can press the button on his Snom 320 that says "Bob" and pick up his line. It works great for calls coming in from the outside but does not work for internal calls. Internal calls generate a app_directed_pickup.c:204 pickup_exec: No

...ed *8, This worked in 1.4. Now, Running 1.6 (I know it's old I had to load it quickly, And that's what I got working first. It'll get upgraded to 1.8 soon). The strange part is *8 no longer works.The only CLI feedback I get is "== Using SIP RTP CoS mark 5" In features.conf, Callpickup *8 is commented out, But just incase I also changed it to *7 (We don't use that feature). It appears to be something completely SIP based, As if the call originates from DAHDI, It works fine.. If anyone has any ideas, Please let me know. Thanks! SIP Trace Below <--- SIP read from UDP:208....

Anyone know how to specify a call group (as specified in the example zapata.conf) when using Dial? No other reference I can find. Jim Friedeck

Hi, I'm having trouble dialling the codes for blacklist & un-hide caller ID (*80 & *82) using a TDM400. I get reorder tone as soon as I dial *8, and in fact I can pick up calls with just *8 rather than *8# when there is one to be picked up. Ideas? Thanks, brad -- Brad Bergman <bradley@bergman.ca>

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?