similar to: Pickup some else's call

I'm trying to duplicate a behavior we had with our old avaya system, and I've come across Auto Answer (Ring Answer). However, its not quite the same yet. Right now, when I dial **5053, it will add the SIP header for Ring Answer and it will call 5053. The phone auto pickups just fine. However, we need that call to be muted. If you were to call into a meeting, we wouldn't want them to

Hi all. I've made a patch for chan_oss.c to enable callgroups and pickupgroups in it (since wasn't enabled). I needed it for a special use of the console (pickup calls arriving to the console from another phone) btw, If someone is interested, I can submit a patch to the bugtracker. I won't do it until that's usefult for someone... since is a very special features that probably no

Hi, I have been trying to workout how to use the call pickup. So Far, I have the following in zapata.conf [channels] signalling => fxo_ks context => local pickupgroup=1 callgroup=1 channel => 1-3 When I dial *8# all I hear is busy tone. What have I missed? thanks Jay.

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

Hello all, I was wondering if anyone has had much experience with Realtime Asterisk. I like the ability to setup my extensions and voicemail boxes in MySQL, but I have a huge worry. What if MySQL crashes. I played with rtcachefriends, but can't seem to find a way to have asterisk store the extension information to ensure the phones will continue to work even if MySQL has a hiccup. Any

Hi all, at time i am trying to get a better idea of callgroups and pickupgroups (especially within the SIP Channel) A Pickupgroup is relative clear - everyone in the same pickupgroup may pickup a call And a callgroup does what ? - The same ? I thought that a callgroup would act like the ZAP groups - so that you then can dial SIP/g1 - and every SIP Client which is in the callgroup 1 does then

Should setting a namedcallgroup & namedpickupgroup supersede numeric callgroups and pickupgroup ? I've got 5 peers on my 13.7.0 box, Three of them have a namedcallgroup & namedpickupgroup of 'kiniston' and Two of them have a namedcallgroup & namedpickupgroup of 'sanday'. I'm not specifying a numeric callgroup or pickupgroup so all the peers are defaulting to

Hi All, I need a function that I believe isn't available in Asterisk, but I don't know if I'm correct about this. I have a queue and I want agents that are in that queue to have the ability to answer a call in the queue with calling an extention. For example, if I'm an agent and my colleague forgot to logout I could take the call when his phone is still ringing without

So I might have a bit of a more narrow question from my earlier one. Previous, I had been wondering what would cause a phone dialing into a DID that connects to the asterisk box to get a fast busy. I've noticed the following message: chan_zap.c: Ring requested on unconfigured channel 0/1 span 2 Any idea what would give me this error? And would this cause a fast busy? Thanks again everyone

callgroups & pickupgroups greater than 31 are not working for sip calls with 1.2.14 tarball. Anyone know which branches support 64? John

Hi, I'm working with this issue for a while, Now I already solve the dialplan issues, but I still have a question about the Callgroups, I read at www.voip-info.org that , there is a 63 limit of callgroups. And I'm wondering why?? and if the 1.2.0beta version supported more than 63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any unoficial patch for that ? Thanks

We have a location that is having a really odd issue. We have a sangoma POTs card. We are running software echo cancellation with the card (through asterisk) to try to eliminate some major echoing problems. I've turned on both EC and echotrain, which seemed to have gotten rid of the echo for the most part. However, we are now running into an issue where the outside caller hears a star wars

I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system

Hey all, I had a user complaining of calls which were dropping mid-conversation. I looked into the time of one of the calls, and saw the following: Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82b8430', 10 retries! Apr 4 12:13:05 WARNING[6660]

We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching

Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working for the secretaries that monitor their bosses' phones. The BLF and the speed dial works great on the Linksys phones. Call pickup is the problem. My features.conf

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. I have several Asterisk 1.4.21 machines, each with ISDN cards in them, and Polycom SIP phones on people's desks. Making / receiving internal / external calls works fine - no problems. We're using 4-digit extension numbers in the offices. I'm trying to work out how to provide a remote pickup facility along the following lines: - phone extension 2233 rings - person at

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

hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? thanks -- --------------------------------------- Marek