similar to: call transfer with consultation

I am trying to find an example of how to set up my FXS Station Card in my Asterisk. I have (1) XP100P I have (1) tdm20B (2 Port FXS) Could someone tell me if this is correct? /etc/zaptel.conf fxsks=1 fxoks=2 fxoks=3 loadzone=us defaultzone=us /etc/asterisk/zapata.conf [channels] ; language=en ; ;X100P Port 1 context=inbound-analog signalling=fxs_ks usecallerid=yes echocancel=yes

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

I tried adding wwww to my dial string... It appears to not made any differnece. I see by watching show channels that the w's are there in the dial. I have this box connected to bell south down in georgia. I have an identical box connected in indiana. The box in indiana works fine when dialing out. When in GA dialout is getting "you must first dial a 1 to place this call". show

hi i need help in connecting a X100P-clone in germany. Basic questions: a)what configuration do I need ? what is the difference (maybe explain in german? via email?) fxs_ls: FXS (Loop Start) fxs_gs: FXS (Ground Start) fxs_ks: FXS (Kewl Start) b) can I use one card for connecting to the telephone network (in this case my analog pbx) and the same card for connecting an analog phone

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

I have turned off call progress in zapata.conf and things seem to be working better. Jerry ---------- >/ I tried adding wwww to my dial string... It appears to not made any />/ differnece. />/ I see by watching show channels that the w's are there in the dial. />/ />/ I have this box connected to bell south down in georgia. />/ I have an identical box connected in

Hey all, posted this the other day, but re-read it & realized I didn't give enough info to be useful.. so I thought I'd try again.. I'm using AAH 2.0 (* 1.2.0) and am unable to transfer a call when I initiate the outgoing call. In AMPs general settings, I've tried changing the Dial command using tT but transfers are only available when I'm the recipient of the call, not

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

Hi List, I have a Multitech H323 Gateway MVP400 box with 1 phone on port FXO 1. I have Asterisk ruuning in Fedora Core 3. Both are in the same network. But I can't figure what I have to do in Asterisk to make that box work. What files I have to configure? Can anyone help me? I will really appreaciate youu help. Luis. _________________________________________________________ Do You Yahoo!?

See: http://lists.digium.com/pipermail/asterisk-users/2004-September/063348.h tml -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kumara Jayaweera Sent: Tuesday, June 14, 2005 1:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help meplease

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

hi all, Where i can found docs about install Xen under CentOS 4 ?? (I only find docs for Fedora). Do I need a precompiled kernel or can I use binary kernel that comes with xen binaries??? And the last question, can I a fresh install solaris 10 under xen?? Thank you very much for your help. _________________________________________________________ Do You Yahoo!? Informaci?n de Estados Unidos y

Hello to all. I have a couple of budgetones connected to Asterisk server. I can establish calls using budgetone with no problem, but when I hang up a Budgetone, Asterisk does not detect the hangup. It seems that the communication goes on in spite of budgetone's hangup. My sip.conf: [general] disallow=all allow=ulaw bindaddr=172.16.60.21 [sip1] callgroup=1 pickupgroup=1 type=friend

Hi all, Somebody knows when centos 3.5 will be released?? Thank you very much. _________________________________________________________ Do You Yahoo!? Informaci?n de Estados Unidos y Am?rica Latina, en Yahoo! Noticias. Vis?tanos en http://noticias.espanol.yahoo.com

Hi, i need move 20 GB of data from a old computer to a new server and I need than the permiss of user, group, other and symlink be the same in the new server. Y try with rsync -avzpogl -e ssh archivoOrigen usuario@ipDestion:/dir2/ but when a see in the new server, the data dont have the same permiss of owner, group than the old server. How i can to have the same permiss? Note: I dont

Nota: Se adjunt? el mensaje reenviado. _________________________________________________________ Do You Yahoo!? Informaci?n de Estados Unidos y Am?rica Latina, en Yahoo! Noticias. Vis?tanos en http://noticias.espanol.yahoo.com -------------- next part -------------- An embedded message was scrubbed... From: michael mendoza <mike114x@yahoo.com> Subject: Re: rsync + ssh -o -p -g -l Date:

Hi all, I am trying to compile madwifi for CentOS-4 without success. Error is: Checking if all requirements are met... ok. mkdir -p ./symbols for i in ./ath_hal ath_rate/onoe ./net80211 ./ath; do \ (cd $i; make) || exit 1; \ done make[1]: Entering directory `/usr/src/madwifi/ath_hal'' make -C /lib/modules/2.6.9-5.0.3.ELsmp/build SUBDIRS=/usr/src/madwifi/ath_hal

Dear R users, I would really appreciate some help, I have a plain text file which contains info like NANATGGGGGGGCCCAGCTTGAATTCCCNTCCCCNTTCCCATCCAGGCC I would like to read it as a vector, for example, (N,A,N,A,T,G,G,G,G,G,G,G,C,C,C,........) I've tried commands such as read.table and scan but I don't get what I want. Thank you in advanced. Sandra

I have a TDM20B and asterisk compiled fine. The drivers have been loaded (wcfxs and run ztcfg -vv, I see the drivers loaded with lsmod). Asterisk starts up fine. I am using the default configuration files that are made when you do a "make samples". I was wondering if someone had a link or website that stepped someone through this kind of setup. What I want to do right now, is use a

Hi, I am trying to configure * 1.0.7 box with a Digium Wildcard TE110P T1/E1 and libunicall latest code. All libs compiled successfully and the E1 have a green light! I am able to receive a call (or at least) testcall shows some information when an incoming call is received so the drivers and basic configuration is working. My * box has 2 cards, one TDM400B (2 fxs and 2 fxo) and one TE110P