similar to: No subject

Hello may situation is the next: Asterisk <--> NAT1 (router)<---> internet <--> NAT2 (router) <--> x-lite ^ | ip phone (cisco) Asterisk and de cisco phone are in the same LAN. I want to make a call between the x-lite and the ip phone. I can do the call but there is only audio from de ip-phone

After upgrading from 1.2.x to 1.4.x call detail records are not being written to /var/log/asterisk/cdr-csv/Master.csv In cdr_manager.conf I have [general] Enabled = yes Apparently there is something else that needs to be configured for call detail records in 1.4.x. Can someone point me in the right direction? Don Pobanz

How to read a xml file? I have this XML source: ------------------------------------------------------- <?xml version="1.0" encoding="UTF-8" standalone="yes"?> <fichas> <ficha> <nombre>Gabriel</nombre> <apellido>Molina</apellido> <direccion>Alfredo Vargas #36</direccion> </ficha>

I have a TE210P digium card that has 2 E1/T1 ports. the code in my extensions.conf file for span 1 is : [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=Zap/g1 ; Trunk interface TRUNKX=Zap/g2 ; 2nd trunk interface ... ... ; dial a long distance outbound number to SPAIN ; This

I have the following line in the sshd_config file: GatewayPorts no If I launch the ssh client as this: ssh -l user host -R 9000:otherHost:25 the port forwarding is successful! :-( As you can see, the 'netstat -na' command shows the Secure Shell daemon listening to the port 9000. Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address

Hey, all. I was just wondering if there were any tools/utilities/what-have-you out there that would allow a user to click on a contact in Outlook, and have their phone dial it? (Or, I guess, have Asterisk dial both their phone and the destination number, and put the two into a conference.) Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is

Buenas Tardes, Les escribo para solicitarles una ayuda dado que tengo 2 tablas, una con los campos: cedula | nombre | direccion y la otra con la tabla: cedula | barrio | municipio Lo que necesito es hacer una comparación del campo cedula de las dos tablas y si son iguales, agregarle los campos barrio y municipio de la segunda tabla a la fila correspondiente de esa cedula de la primera

Hello!! I am new user to Wine. I have downloaded Wine-0.9.15 sources on solaris box and tried to compile a simple "Hello World" program using WineLib. I am getting following error: winegcc: -Wl,-G,-B,symbolic failed. *** Error code 2 make: Fatal error: Command failed for target `helloworld.so' Can anybody guide me as to what went wrong? I have following environment setup

Right now I've been working on setting up an Trixbox server on our internal network. Its behind the firewall, but I'd like to open up the firewall to it because we sometimes have developers working off site and I'd like them to be able to connect. Is this safe to do? I've got the "Allow Anonymous Inbound SIP Calls" box unchecked in freePBX. Is there anything else

Hi, I have a requirement, whenever a user comes into the conference, it has to announce the user name to all the person who are all available in the conference. I have used Meetme(,di) where i is to announce the user leave/join with review. I user used I also, which is to announce the user leave/join with out review. In both the above cases, it is prompting the user to say their

I have the following models: class Comandancia < ActiveRecord::Base has_many :elementos end class Elemento < ActiveRecord::Base belongs_to :comandancia validates_presence_of :nombre, :apellido_paterno, :apellido_materno, :comandancia end In the view for New Elemento I''m using <%= start_form_tag :action => ''create'' %> <%= render :partial =>

After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports

My application fires several calls thru AMI ORIGINATE command. For example if I have 3 operators I do 3 ORIGINATEs. My trouble is when one operator quit for some reason, I should kill the corresponding ORIGINATE. Of course, I could let the call ring and hangup after the customer pick-up. But this is not the case, I do have to kill the corresponding ORIGINATE. I could execute a soft hangup,

Hi!! I am not able to compile a test program on solaris using winegcc tool. Can anybody help? Here's the program source code: // file test.cpp #include <stdio.h> #include <windows.h> int main () { SYSTEMTIME lpSystemTime; GetSystemTime(&lpSystemTime); printf("Today is: %d/%d/%d\n", lpSystemTime.wYear,

We want to setup Samba as a domain controllser on our 3 sites , we'll be highly appreciated if you solve our queries : 1. Can Samba act as a backup domain server if incase Master Samba server is down then Backup Samba server can take place . Same like Windows concept if PDC is down then BDC will up automatically. 2. Can we configure SAMBA in that way that user can login from any location in

Hello, using Asterisk 1.4.28, I encountered a problem with SIP RTP port allocation. I found some entries in mailinglist and bugtracker regarding this issue, but only old ones. My rtp.conf has [general] rtpstart=30000 rtpend=30100 so 100 ports available. I know that up to 4 ports per channel can be used and so up to 25 channels are possible. But even earlier I often get the error about

Hi, I'm bumping on this : cd /usr/src wget http://www.misdn.org/downloads/mISDN.tar.gz tar xvf mISDN.tar.gz cd mISDN-1_1_18 make <snip> In file included from /usr/src/mISDN-1_1_8/drivers/isdn/hardware/mISDN/core.h:9, from /usr/src/mISDN-1_1_8/drivers/isdn/hardware/mISDN/avm_fritz.c:20: /usr/src/mISDN-1_1_8/include/linux/mISDNif.h:791: error: field

Hello, excuse me for my English. I have several samba installations in different places and in all of them I have the same problem. Therefore, something I am doing wrong or I understand wrong. When I want to manage the share from Windows, it returns the error: "Rpc server is unavailable" and on the other hand in the edition of permissions (also from Windows) in advanced is the legend:

Hi, I have two asterisks that are connected to each other via a back-to-back E1 link using a pair of sangoma cards. With the following scenario: SIP-PHONE <-> Asterisk <-> E1 <-> Asterisk <-> SIP-PHONE, the sound quality degrades significantly. I can't understand why as the amound of packet lost should be very minimum. Does anyone know why? Does it have anything

Hi folks, Is it possible (asterisk 1.6) to trigger the playback of an audio file in the middle of a call using the Manager Interface? I'm looking for something like AMI PlayDTMF command but for audio files. Thanks a lot, G. -------------- next part -------------- An HTML attachment was scrubbed... URL: