similar to: Manager logic to pickup a ringing extension

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

From: "Nik Martin" <nmartin@radiancetech.com>> To: <asterisk-users@lists.digium.com>> Subject: RE: [Asterisk-Users] asterisk compile problem Date: Wed, 14 Jul 2004 09:22:38 -0500 Organization: Radiance Technologies, Inc. Reply-To: asterisk-users@lists.digium.com Fletcher Bonds wrote: >> Hello all >> >> As of 5pm PST today (7/13), I pulled

I'm having a tough time getting call pickup to work on *. Here's my configuration: X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image A call comes in, and * picks up and presents a menu. Caller chooses extension, (in this case ext 103, SIP/wsmith) Wsmith is sitting in my office, hears his phone ringing, picks up my phone, gets dial tone, and presses

If I have an IAX client (Firefly softphone in this example), and the client is not registered at the moment because they are not connected to the network and someone dial that extension, they get the user's "I'm on the phone at the moment" message vs. the "I'm unavailable" message. Is this by design? Here's the extension in question's dialplan:

Of course, big brother wants his say in the matter. http://www.wired.com/news/politics/0,1283,63884,00.html?tw=wn_2polihead

Nik Engel wrote: > Hi list ! > > I implemented *8 to pickup any call on my asterisk system. But after the > pickup callerid is missing, so there is no way to see from where the > call originated. How can this callerid be passed on. > > Nik > Hi Nik, I'm after the same question as I would like to keep callerID data after pickuping up the call. Maybe using a

Hi all, I've just setup a sip line with Telasip and when they route the calls to my asterisk box, they include an extension along with the context that is defined in sip.conf for that DID. At first, I couldn't figure why they were getting 404 error from my asterisk box, but then figured out that they are sending the call to an extension that matches my number with them, in the

Hi, I'm having alot of trouble understanding how to use dialplans for outgoing calls on Dahdi. Context : I have 3 TI spans, so 69 voice channels and three D channels (24,48,72). This is on a TE420B from Digium, if it matters. Here are my (apparently simple) questions in no particular order: 1) Dial(DAHDI/5555555555|20) doesn't work. But Dial(DAHDI/42/5555555555|20) does

Hello, I am having something of an odd problem: about every 100 calls or so, when a call comes in via an external mISDN interface and I route it to an internal mISDN interface by dialing an internal msn that is programmed for multiple phones on the internal bus, somtimes the other phones continue ringing for several minutes after the call has already been picked up by one (or even eventually

Hello. I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : */wget 'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug' -O - | patch -p0/* ?? Does this mean I have a "patched" asterisk ? (I ask this because some applications require a

Hi, I've run into a minor snag trying to pass on CALLERID presence from one Asterisk to another via SIP (both running 13.16.0) I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP. PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has its own callerid values and presence. I pass on those calls to PBX_B via SI, and I'm trying to pass on this

Ok, I've been running applications on 1.4 for quite some time using meetme to hold a person, while the person on the other end of the call accepts, etc. I was playing status messages to the calling party using a context like this: [status-one-en] exten => 100,1,Playback(my_status_message) exten => 100,1,Hangup() and then creating a call file like this: Channel: Local/100 at

Hi All, I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would like to be able to pickup ringing extention from any SIP phone using Pickup() application. from my dial plan: [incoming] exten => s,1,Dial(SIP/somebody1|60|tTrR) [internal] include => outbound-local include => parkedcalls

Hi everyone, Is there a simple way of answering a different ringing extension from a sip phone using AAH? I have absolutely zero technical know-how when it comes to modifying conf files etc. Still working on figuring it all out. ;) That brings me to my second question... where the hell does one find an extensive manual of sorts that explains all conf files and what the strings all mean etc?

Hello, I'm using asterisk queues, for reception phone, and I have small problem: I have only one phone as queue member, and the problem is, that ALL channels waiting in queue are ringing on it. And if there are too many people ringing on it, it's not possible to use attended transfer then... Is it possible to limit maximum ringing calls from queue? or some other tip? thanks a lot in

Hi, I've been experiencing trouble with my DAHDI channels for some time and have finally decided to try and resolve the issue. Essentially, the problem I am having is that when a call comes in, and my DAHDI phones therefore ring, Asterisk thinks that one of the handsets has picked up to answer the incoming call - whereas in actual fact it is still on hook. The call then gets instantly

Hello everyone, I use winbind against a Samba DC for nsswich, and on one client it works perfectly (Samba 3.5.15 on all systems). On another client, everything works except SID_TO_UID (i.e. wbinfo -i, -S ... which breaks directory listings, too). I've now tried to narrow down the problem in a level 10 log, but I need some help interpreting. In log.winbindd, I see the following when

Hi, I was trying out call file just to see how they worked and my system does not seem to do anything with them, although asterisk *is* deleting the files that I put into /var/spool/asterisk/outgoing. 1. I nano'd a quick call file like so: Channel: SIP/axVoice/9105555555 CallerID : Leebo <5555555555> MaxRetries: 2 RetryTime: 30 WaitTime: 10 Context: main_menu Extension: s Priority:

Hi, What do I need to do to get asterisk to NOT pickup a Zap channel when it rings? The channel in question is used for outbound calls only, and all incoming calls are answered by an analog phone elsewhere in the building that does not run through asterisk... so.. either make it not answer.. or make it delay for like 90 seconds.. I've tried wait's.. but it still seems to pickup the

Hello fellow asterisk users, I'm trying to use feature application defined in application map. it's defined as follows: lbxvml => 1,self/caller,Macro,Jump2Voicemail It's working properly when called party answers the call, but I'd like to have feature usable while call is still ringing in queue but this just does not work.. Is this a bug or feature? Is there a way to have