similar to: Asterisk and DMS100 Channelized T-1

I'm developing asterisk to work in a small Call Center for a Mobile Communications Provider. I'm looking for references for Agent Logins and Huntgroups. If any of you have already configured Asterisk for such a Task and would be willing to let me take a look at your config files could you please email them to me at mailto:jhelmich@bluesky.as Thanks, Jason Helmich MIS, Blue Sky

I'm running FreeBSD 4.6 release and Samba 2.2.4_1. Recently I've been having trouble managing access rights on my shared directories. scenario: I have a shared directory "bluesky" which sits in path /fileserv/bluesky directory bluesky has a mode of drwxrwx--- the owner of the directory is john and the group is bluesky group bluesky has members: john, jack, jenny, jake my

Hello there, I have a habit of compiling R-patched and R-devel every few days. Recently, I noticed the following problem: I downloaded both versions with Revision: 40421, Last Changed Date: 2007-01-09. R-patched compiles with no problem. R-devel compiles fine but when it starts to make PDF documentation it generates the following (tail of the "make distribution" output): ! pdfTeX

hi,all: I come here to consult that whether FFmpeg could be replaced with Theora in our project. Here the thing. Our project, Bluesky( http://incubator.apache.org/projects/bluesky.html) is podling in Apache incubation. However, core function of our system, the encode/decode part which implemented by FFmpeg(GPL licensed)(we mainly use libavcodec and libavformat library), conflicts with Apache

Hi all I am trying to build an R package, which I have successfully done many times before, but have an error I cannot trace. I hope someone can help me. Here's is some edited output (full output below if it is useful): pdunn2 at PDunnUbuntu:~/DSdata$ R CMD build GLMsData * checking for file 'GLMsData/DESCRIPTION' ... OK * preparing 'GLMsData': * checking DESCRIPTION

I don't mean to be a troll in any way shape or form. I was on IRC last night and I observed the following convo. below. What do you guys make of it ? [02:14] <bkw__> Let me tell you how chidlish digium and Mark Spencer is. I walk into a restaurant with them all here at Astricon wearing my sangoma shirt and he asked me to leave. [02:15] <Dovid> u serious ? [02:15] *** mog

hi all: i downloaded libtheora1.0.tar.gz from xiph.org, and when i compile it, such error message popped. i tried to run player_example.c under /libtheora_1.0/examples/. error message: ivysummer at ivysummer-desktop:~/??/libtheora-1.0/examples$ gcc -o player player_example.c /tmp/ccpyle3c.o: In function `buffer_data': player_example.c:(.text+0x15): undefined reference to

Dear all, I want to start my post by expressing my sincere gratitude for all the help this group has given me in Sweave/Latex/R. The tools are excellent and so is the community! On to the question. My Sweave code is intended to make lots of plots and create a *.pdf document. Sweave is perfect for this. The only problem is that I find myself using the same R code, within my Sweave input file

hi using sipfriends from mysql from asterisk 1.0 branch, how can I make asterisk show the true channel's current codec with SIP SHOW CHANNELS? This does not seem to work, and bkw_ said sipfriends from mysql didn't have that info at all. For what it may seem, asterisk uses G.726 as told, giving me a -- Format for call is g726 at the start of the call, but in SIP SHOW CHANNELS all these

Hi, I have been making an RPM of FLAC to bundle with GStreamer. In order to get it working I had to make some rather hackish solutions in the SPEC file. The flac Makefile does to build into the correct directories while creating an RPM for some reason. I have attached the SPEC file I ended up with if it is of interest. Of course it didn't help me much cause it turned up we had a bug in the

If you have been paying attention, you already know this, but this weekend I have spent time ironing out the various details with my chan_skinny code that has been out there, if you knew where to look. I believe I now have all basic features operational and am going to be working on getting the class 5 (hold, transfers, call waiting and caller*id, etc) operational in the comming week(s).

Today class we are going to be talking about the wonderful line of GrandStream products. Or should I say BarbieTone phones? Who else is having MAJOR issues with the grandstream products? How many times have you been told upgrade upgrade upgrade? How many of you have paperweights, granted the phone is light as a feather and couldn't weight papers down in the first place? How about that

Hi Folks, I'm trying to send CallerID Name information out to the PSTN via a PRI with Bell Canada with no success. With inbound calls (originating from the PSTN) CNAM is received successfully, and we've not had any similar problems with other Telco PRIs, so I'm stumped.

----- Original Message ----- > Message: 3 > Date: Thu, 16 Sep 2004 07:57:15 -0400 (EDT) > From: David Troy <dave@popvox.com> > Subject: Re: [Asterisk-Users] No Caller Name sent from Asterisk over > National or DMS100 PRI to a Norstar MICS? > snip> > > I have a PRI link up and running between Asterisk and a Nortel Norstar MICS > > v4.1 . I'm having a

All good information, thanks. However this is private network between Asterisk and a Norstar MICS about six feet away. So I'm holding both ends of the link. :-) > -----Original Message----- > From: David Troy [mailto:dave@popvox.com] > Sent: September 16, 2004 4:57 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] No Caller Name

I have not been able to get two B-channel transfer to work on DMS100 PRI. I consistently get the following errors: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ROSE RETURN ERROR: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: OPERATION: RLT_OPERATION_IND [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ERROR: RLT Not Allowed I have tried on two

Has anyone really looked at the costs for Astricon. But the hotel costs. $111.00 USD per night.. come on guys give me a break. I will not be staying at that hotel. I can rent a car and stay near the air port for almost half that. In addition from what I have been told their will be no shuttle service from the Airport to the hotel. Anyone else have any input on this? bkw_ PS: I'm going

Another issue of Asterisk Summer News, delivered right to your mailbox! Back here in Sweden, it's finally summer weather. Sunshine and some heat. It's good for our ice bears and the snow houses to get some sunshine :-) Asterisk development and IRC chat has gone into a lazy summer mode, but the mailing list is still cooking. It's impossible to keep up with it, for both gurus and

I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)

Hi folks, I've just put CentOS 4.1 on my webserver, displacing OpenBSD (1 up for Linux :-)) I found that the Virtual Hosts function doesn't appear to work. All virtual hosts accesses default to the main www directory instead of serving from their respective directories. Has anyone else found the same problem? Thanks. Ben