similar to: asterisk 0 dial outgoing call

Dear all I have install asterisk 1.2.x and it is working fine my setup is like [*]-------[Mediant2k]------------[Avaya] Now i want to transfer call in internal extension i have read more document on www.voip-info.com but it is now so much clear so if u have any sample configuration file and doucment plz suggest me i have configure feature.conf and extention.conf for this task

Hi, Is there a way to send and receive SMS over SIP protocol with Asterisk ? I mean, between two SIP phones like below... SIP_phone "A" (sending sms) ====> Asterisk ========>SIP_phone "B" (receiving sms) ... Is it possible ? If so, how could I do it ? Thanks, Angel. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all anyone have idea about connect asterisk with mediant 2000 audiocode configuration ... anybody have configuration about it --------------------------------- Get your own web address. Have a HUGE year through Yahoo! Small Business. -------------- next part -------------- An HTML attachment was scrubbed... URL:

When I did install.packages("sqldf") on Windows and Mac OSX, it installed fine. However, when I did it on my Debian Squeeze box under R 2.14.0, it failed. I discovered that three of the dependent packages, chron, proto, and gsubfn, do not include a NAMESPACE file in their distribution tar.gz files. I contacted the developer, who told me that, for packages without a NAMESPACE file, R 2.14

Hello, I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum [sip_proxy-out] type=peer outboundproxy=QUINTUM_IP , and changed extensions.conf. When

I am on a rh 7.2 machinr running samba. My clients are windows 2000 and whn using locl profiles I experience quick log-ons and quicj log-offs. Whwn I flag the client for roaming profiles, it seems to take many minutes to log off. The users roaming directory IS created and is populated with their user inoformation, one user has 300M of stuff.Which I thought was alot. When the user goes to

I need to set up an office full of Cisco 7960 phones behind NAT with the server out in Colo. The first test phone registers fine, but the second one does not register. The first phone's registration looks like so: /SIP/Registry/3115552368 :64.169.xx.yyy:38836:3600:3115552368:sip:3115552368@64.169.xx.yyy:5060 When the second phone tries to register, it gets back a 404 not found. Not a

Hello list, I am trying to apply the paired t.test between diseased and not diseased patients to identify genes that are more expressed in the one situation under the other. In order to retrieve the genes that are more expressed in the positive disease state I do: p.values<-c() for(i in 1:length(Significant[,1])){ p.values[i]<-try(t.test(positive[i,],negative[i,],alternative

Dear all i need this feature in asterisk whn 2 party calling that time i pickup call and listen conversation of that party spoofing like is it possible in asterisk Rgds satish patel --------------------------------- Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool. -------------- next part -------------- An HTML attachment was

Hi List I have data in the following form: Gene    TFBS NUDC     PPARA(1) HNF4(20) HNF4(96) AHRARNT(104) CACBINDINGPROTEIN(149) T3R(167) HLF(191) RPA2     STAT4(57) HEB(251) TAF12     PAX3(53) YY1(92) BRCA(99) GLI(101) EIF3I     NERF(10) P300(10) TRAPPC3     HIC1(3) PAX5(17) PAX5(110) NRF1(119) HIC1(122) TRAPPC3     EGR(26) ZNF219(27) SP3(32) EGR(32) NFKAPPAB65(89) NFKAPPAB(89) RFX(121)

I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have

Hi, I am a new linux user.I jz setting up a mail server.Downloaded & Configured the components:CentOS,OpenLDAP,pam,dovecotMailServer,s endmial,squireelmail etc.... But wehn i want to check whether it works or not, so i open up my clientmail, wchi is Netscape 7.2. in Edit > Mail & Newsgroups Account Setting > server setting... i had put the ip address for the machine... but error pop

Hello, I''m a noobie to Rails and even more to mongrel. I have an issue with one server that I''ve installed mongrel. Its a Fedora Core 5 box, and it is having problems loading the Oracle OCI8 libraries. However, this only occurs when I use mongrel_cluster, or mongrel with out the - d switch. I have no problem using the library from the commandline and with Webrick or

Hi pros, Newbie to asterisk, need some help. My existing senerio is we have 6 analog quintums and 1 digital H323, and our gatekeeper is gnugk openh323 located in US. Our business is Call Center and our method of dial is using prefix and gateway IP provided my Carrier. I also brought two AudioCodes MP108 8 FXS gateways, as our gateway runs on h323 my friend suggested to go for Asterisk. If

Full_Name: Peter Perkins Version: 1.1.1 OS: LinuxPPC Submission from: (NULL) (24.4.89.36) the man page for grep mistakenly says "gsub" whn it should say "grep" in the Details section of grep.Rd. a diff -u with a corrected version follows: diff -u R-1.1.1/src/library/base/man/grep.Rd grep.Rd-new --- R-1.1.1/src/library/base/man/grep.Rd Wed Apr 19 08:51:19 2000 +++

Dear all I have asterisk 1.2 configuration and it is working fine but thing is that i have alread Avaya setup and i have intergrate my Linuxbox asterik with Avaya system avaya already use 4 digit dialplan (1644 example ) and in asterisk i have configure 2 digit dialplan ( 44 example ) now i want to configure 4 digit dialplan in asterik without any change in avaya or asterisk so

Hi, I just upgraded by 1.0.x home server to 1.2.2. Overall the upgrade went fine, but a strange problem has cropped up with the CALLERID name value of incoming calls from the X101P card. When an incoming call is presented (via Vonage ATA), the calledid value getting double quotes up from: -- Executing NoOp("Zap/1-1", """WIRELESS CALLE" <1404xxxxxxx>") in

"A Tale of TTL Troubles" I was hired to implement VPN for a subnet. The owner has a /27 at his home site, and he wanted to have the machines there answering BOTH on those IP addresses and some addresses at a remote colocation provider. Make sense? Not to me either. :( I think he''s trying to fool his customers into thinking he has a physical presence in the colocation city.

Hello - I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am having problems with routines that input long strings of numbers, in that I am getting more than a small number of double digit entries. As an example, I have a section that asks for the user to enter a call forwarding number, and then puts that number into a database. Almost always, there are double digits when the

Dear sir I have setup Avaya with mediant with asterisk [sip_phone]---[ * ]---[mediant]---E1-trunk--[Avaya]---[analog_phone] This is my configuration when i call from SIP phone i got ringing sound of phone but whn i call from analog_phone behind avaya i didn't get ring sound of ring but SIP phone speaker ring why i am not able to hear ring sound from analog phone Regards