similar to: one doubt

Can someone offer a hint on what I'm doing wrong with the basic * config? Just implemented * for the first time using yesterday's cvs. The initial configs are based on John Todd's article at http://www.onlamp.com/lpt/a/3956, and using two 7960's for initial testing. When one 7960 calls the other, I get the following and the call is dropped: -- Executing

Using the following defnitions with a C7960: exten => 3001,1,Dial(SIP/3001,15,r) exten => 3001,2,Voicemail2(u3001) exten => 3001,102,Voicemail2(b3001) exten => 3001,103,Hangup If someone is on this phone (real conversation) and another call comes in, the second call goes through the 15 second timeout and is dropped into the 2-priority as "unavailable" (not the 102 busy as

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu -------------------------------------------------------------------------------- SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw

I have a SwissVoice IP10S but can not seem to get it to have dialtone or dial on *. Calls to or from 3001 don't work. Any ideas are appreciated. Robert mgcp.conf is: [general] port = 2427 bindaddr = 192.168.0.110 [ip10] host = 192.168.0.5 context = from-sip line => aaln/1 The portion of extensions.conf is: exten => 3001,1,Dial(MGCP/aaln1,20) exten => 3001,103,Hangup

Has anyone used the H option for Queue() with Callback queues? I want customers in my queues to be able to jump out to voicemail when they get tired of waiting, but in my setup when I pretend to be a customer and press '*' [when I am waiting in the queue] I see the message 'User hit * to disconnect call.' but then just jump out to the outer loop where queued callers wait to

Hello, I am quite new in this, and I have not been able to figure out what the problem can be after looking at the documentation for quite a long time. I guess that it will be a stupid mistake, though. I have RedHat9.0 and have installed wine with the .rpm for this distribution available at sourceforge.net. When I try to execute the binaries 'wine' or 'winecfg' I get the

Hi, I'm trying to run Derive5 in Redhat9.0 with wine-20030508 i get the following err's when i try to run the app: fixme:ole:CoRegisterMessageFilter stub fixme:font:WineEngCreateFontInstance just using first face for now fixme:font:WineEngCreateFontInstance Untranslated charset 255 fixme:font:WineEngCreateFontInstance Untranslated charset 255 fixme:font:WineEngCreateFontInstance just using

Hi Everybody, I Created a small application in VC++ MFC. The application displays a messagebox on clicking a buttton. This application I want to port on Linux version. I am using Linux RedHat9.0 and kernel version is 2.4.20-8. with WineLib I am following the guide lines given in Wine LIB User guide. I copied all the source code from CD and I change the mode read /write than in the current

Hi, I am trying to synchronise two directories on separate machines across a network using rsync, but I want to apply rsync whenever the contents of the directories are altered, rather than using a cron job. Does anyone have any ideas? We are running Redhat9.0. Thanks Henry

I have a T100P card connected to an Adtran and then a T1. I have added the following configurations to Asterisk...but, when I dial 9 and then a local phone number, it bounces between the dial tone and silence and the *error* light on the Adtran blinks. zaptel.conf span=1,0,0,esf,b8zs fxsks=1-8 loadzone=us defaultzone=us zapata.conf [channels] context=from-sip signalling=fxs_ks

Hi. I have 2 PC's connected with 1GBit NIC's. When I transfer a file from my File-Server(Redhat9.0, 256 SD-RAM, 300MHz PII, RTL8169 NIC, 2x Western Digital WD200JB RAID 0) to my Windows-PC(AMD Athlon XP 1800+, 1024 MB DDR-RAM, WINXP PRO, RTL8169 NIC, 2x Western Digital WD080JB RAID 0) with Samba, i get Speeds around 8-9MB/sec. I think this is too low for an GBit Network, so i tested the

hello to all. I am working on macroeconomic data series of India, which in a yearly basis. I am unable to convert my data frame into time series. kindly help me. also using zoo and xts packages. but they take only monthly observations. 'data.frame': 30 obs. of 4 variables: $ year: int 1980 1981 1982 1983 1984 1985 1986 1987 1988 1989 ... $ cnsm: num 174 175 175 172 173 ... $ incm:

I use the winbind to map the w2k PDC users and groups to the redhat9.0, but a few problem appears,when I run "wbinfo -u" can list the user but "getent passwd" can't list the user,new created users in the w2k PDC also appear this problem.why? example: [root@honet5 lib]# wbinfo -u | more GZ+xiangmao GZ+xiaobinl GZ+XiaoHongChen GZ+xiaoliangc

I'm having some problems with valid field names when using RMySQL to interface R (version 1.7.0, under RedHat9.0), to MySQL (4.1.0-alpha). I think I've spotted the problem and a solution (which is working for me), but I wanted to share this with you as I may be missing something. (Note: I'm aware that this is an old R version, but I've checked the code of the lastest version of

Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18 kernel. I am using kernel 2.6.18 but facing a very serious problem of modutils and iptable. Can anybody help me out. Thanx and Regards sanchal singh

Respected sir, hoping that you are well.sir, i am trying to run Tado-Yamamoto causality test with my data. I have three variables. but in running wal.test in R, I have faced problems (especially in 'terms' arguments). my results have shown as... Error in L %*% V : non-conformable arguments -- kindly help me in solving this issue. I have also attached my codes and data to this email.

Hello, I'm trying to set up a conference room. When I dial it's extension, I get an audible error saying "Not a valid conference room, please try again" followed by a disconnect. I've got debug sip peer 1001 (my X-Lite client) and I see this in the logs: (I'm pretty sure it has something to do with ztdummy, but I dunno... I have the port installed, but I

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

Hello all. Now I aleady installed the Asterisk. I could make communication between 2 XLite client through Asterisk. I tryed to test the voicemail function as follow. 1, I make a call to 1001 from 1002 2, Start ringing 3, Wait untill time out for ringing If no problem, 1001 go to voicemail and unavailable message will be played. But 1001 receive a 403 forbidden massage and connection go