similar to: Calls directed via queue to unavailable device result in call acceptance

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my

Hi, Is there a way to add another extension to a live dial, for example Dial(PJSIP/1000,,) and after 20 secondes change it to Dial(PJSIP/1000&PJSIP/1001,,) I am open to suggestions such as using manager or stasis. Thanks in advance. Best regards, Kkh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have configured a call queue as follows: [sales] music = default strategy = roundrobin timeout = 20 retry = 5 reportholdtime = yes announce = queue-sales member => Agent/9021 member => Agent/5901 member => Agent/9020 member => Agent/1114 Now, I would expect all calls to this queue to be delivered first to 9021, then to 5901, then to 9020, then to 1114 (all agents are logged in

I would like to perform partial mantel tests on only within group values, with "between group" values assigned to NA. This is possible in package ncf partial.mantel.test, however this sues a different permutation to that used in ecodist.ecodist will not accept data with NA values, returning a "matrix is not square error. is it possible to perform this test in ecodist? many thanks

I'm trying to get my Cisco 7960 configured to work with Asterisk, with no luck. I'm sure I'm missing something very easy... since I know others have this working. I've stepped through Andy Powell's excellent "Getting Started with Asterisk", and it works for my X-Lite softphone. My sip.conf entry for the cisco looks like this: [cisco] type=friend username=cisco

Hi all, My latest Asterisk system is based on Debian squeeze with Asterisk 1.6.2.6-1 and SIP only. One of my favorite features that I had working with Asterisk 1.4 is the PrivacyManager. However, this was not straightforward, because anonymous SIP calls arrive with ${CALLERID(num)} = "anonymous", instead of being blank. So, to get it to work I added the first three rules to

Hi all. I'm trying to get a simple configuration working so I can later expand it to something more interesting. I'm using kphone to call an extension on the * server. When I try to connect, I get this error: DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all!

I am having one frustrating time trying to decipher why ov_read is failing in my MFC/C++ application, and I see in Google results that others have had difficulty as well. These google discussions generally involve libraries, dll's and linking options, as though there are perhaps some basic incompatibilities between ogg vorbis and win32 and/or MFC/C++. Very confusing. If I pass fopen a

Hello al, Is it possible [indeed is this the right place] to add iptables to force all internet traffic to go thru a particular computer on a LAN? I have a 4 port Router/modem that contains a Busybox v0.61 Linux system. I am able to add entries to the iptables tho'' I don''t really know what it does yet. I want to be able to use Ethereal on this one computer to check what web

Has anyone successfully made this scenario work in 1.4. I found info at http://www.voip-info.org/wiki/view/Asterisk+presence indicating that this does not work with 1.4 implementations. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100903/13ed20ba/attachment.htm

Hello All, I went through lot of searching for papers to do clustering on Co-Authroship.I ran SNA metrics on my Undirected Graph(betweenness,Closeness,Eigen Vector) using R. After running the sna Metrics I found important nodes and did clustering on those nodes based on Binding strength .This clustering is done through if they are connected.I would like to do clustering based on Weight of the

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, is there a pass for profile-directed optimization or transformation, which doesn't use ProfileInfo- and ProfileLoader-class? Perhaps in a project based on LLVM that uses profile data generated 'outside' of the standard LLVM-chain shown in 'profile.pl'? Is there an example for such a pass I've slipped? Kind regards, Nico

Hello, everyone. How can I implement profile-directed if-conversion for an specific architecture on LLVM? I know if-conversion pass of LLVM. Thank you for your help! I am new LLVMer, your presents do me a favor!

Hello, everyone. How can I implement profile directed if conversion for an specific architecture on LLVM? I know if conversion pass of LLVM. Thank you for your help! I am new LLVMer, your presents do me a favor!

I've tried the directed pickup like explained in http://www.voip-info.org/wiki-PBX+Call+Pickup but I've got no way doing it work. When I call the ringing phone it use the second line of that phone Group call pickup works perfectly. Is there some special configuration to do to make it works? Best regards, ?ric

Hi! i have a problem with the PICKUPMARK of the Pickup() application. E.g. A calls B. B is ringing. C wants to pickup B. To make this work with PICKUPMARK I have to add the variable PICKUPMARK to B. But how can I do this? B is just created inside the Dial() application. thanks klaus PS: If you wonder why I use the PICKUPMARK - Pickup(exten) does not work for me as the incoming calls will

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