similar to: pick sip channel whn two party talking

Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-----[*]----[mediant2k]-----[Avaya_PBX]------e1-----[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give

Dear all I have one more question about codec what codec i use for LAN setup G.729 or Alaw which is best for LAN setup caz some people told me G.729 is use for wan link not for lan caz it is cost effective so can anyone suggest me best codec for asterisk and SIP phone Rgds satish patel --------------------------------- Don't pick lemons. See all the new 2007 cars at

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 to all i think this is not a new problem for this forum....but its newest for me as i m a new linux lerner. Even if it is new plzz....reply me ur answer..n if its already asked n have solution..plzz forward the solution. My problem is here mentioned: I have fedora core 4 as a linux server. there r two external links connected to this. the settings are as: eth0 ->for internal (that is

Author: wyllys Repository: /hg/zfs-crypto/gate Revision: 1b97a96daa581c4f53b4fd8acceb76a27a9fe324 Log message: 6310540 6290437 causes gss_accept_sec_context not to output ret_flags whn no deleg cred; breaks ssh Files: update: usr/src/lib/libgss/g_accept_sec_context.c

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

Dear all I have asterisk 1.4.11 i am new in asterisk i want to see online call list how it is possible to see how man call currently active is there any command or tool to see online call ?? from --- to Regards --------------------------------- Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. -------------- next part

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

just recently I have finished implementing support for saving non-audio chunks in RIFF WAVE and AIFF with the FLAC file in application metadata blocks, and restoring them when decoding. you can read about more about it here: http://www.hydrogenaudio.org/forums/index.php?showtopic=56968 the zip file there has windows binaries; CVS HEAD has everything checked in if you need a different platform.

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

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)

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 +++

Hi, Here is a newbie to Yum on Centos 5. I got an interesting yet misterious problem here. When I rolled a mod_jk package and tried to install it onto a bunch of Centos 5 boxes with comand 'yum -e 10 -d 10 install mod_jk', it fails with the messages at end; and so mod_jk just ended up ignored. But when I manually run 'rpm -ivh --test mod_jk...rpm' on command line it reports

Hi all, If I have code that does this: while (FLAC__stream_decoder_process_single (decoder)) /* Do something. */ ; I get an infinite loop. Shouldn't FLAC__stream_decoder_process_single return false if it gets to FLAC__STREAM_DECODER_END_OF_STREAM? If so, here's a patch. Cheers, Erik ------------------8<------------------8<------------------8<------------------

Dear all I have beginer in Voip and i have configured Asterisk server with 100 IP SIP phone ( SNOM ) everything is fine but problem is how to transfer call from one user to other means i call to some one and then someone want to transfer call to another person how it is possible i have also try with feartue.conf but it is now working i have also read document on voip-info website

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

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

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