similar to: asking for some help

hi folks, i hope that i will get some help about this issue,so my configuration is : X100P card ,with FXO port ,the problem is that when i send a call using Originate,every thing goes well but what i realy need is to know how can i detect the status of this channel till the called person hung up or pick up the phone or may be busy .so how can i know that those dahdi show status and the like

Hi, I started playing around with a function for using StatTransfer (version 10) for importing data. This started as a simple task but it's not working and so now I'm very frustrated. I'm using R version 2.13 on Windows 7. The function, called fn.importData, is: function(file = NULL, type = NULL){ ## ## create statTransfer command file -

Hi every body; I want to intall some softwars working with my Asterisk server and I get these erreurs : * error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. /usr/sbin/grub-probe: error: cannot seek `/dev/sda'. dpkg: error processing grub-pc (--configure): subprocess installed post-installation script returned error exit status 1 *

Hi list, I'm trying to create an asterisk 1.8 rpm with SRTP. I found mention of a libsrtp rpm, <http://qutecom.ipex.cz/RPMS/srtp-1.4.4-1.i386.rpm > in these instructions, <http://www.voip-info.org/wiki/view/Asterisk+SRTP> but it is unreachable (by me, anyway). The libSRTP source is here, <http://srtp.sourceforge.net/download.html>. Has this already been packaged for

Hi Bilal, if you're looking for Asterisk, CRM and even Google Contacts integration, do check out Aptus FonB (www.aptus.com). I believe that's the exact solution you're looking for. Br, Maax __________________________________________________________________ Hi; I used vicidial for call center and I would like to try elastix. Can someone advise about the advantages? Does Elastix

what is the difference between FreeSwitch and Asterisk , whitch one is more scalable and reliable? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi everyone, actually i want to release an IVR system using PHPAGI API , in this IVR i want to get value from the user. I already used get_data defined in phpagi but they are not able to get the value given by the user and store it in a php variable. i tested this : $result = $agi->get_data('beep', 3000, 20); $keys = $result['result']; but every time i found in $keys variable

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. Regards

Hi I am trying to deploy freeswitch with Digium TE121 card for my office setup, but it is continuously showing Signaling is up and channels are down except D channel. Our Architecture is like We have freeswitch installed with libpri1.4 and Dahdi. I am from India and here we are having E1 trunk. Dahdi Configuration is cat system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 7

I am trying to install the binary version of xen 3.1. When I boot, after some console output, I get the following: :: Loading Initramfs scsi_mod: no version for "struct_module" found: kernel tainted. SCSI subsystem intialized :: Running Hook [udev] :: Loading udev...input: AT Translated Set 2 keyboard as /class/inpt/input0 done. :: Running Hook [keymap] :: Loading keymap...Cannot find

Hi all, Today I got problem below and my domU become unresponsive and I should restart the pc to make it running properly again. [ 240.172092] INFO: task kworker/u:0:5 blocked for more than 120 seconds. [ 240.172110] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [ 240.172376] INFO: task jbd2/xvda1-8:153 blocked for more than 120 seconds. [ 240.172388]

Hi all, I'm new to VoIP, now we have a project that needs a PBX with client APPs. In our team we have argument for choosing PBX. By so far, we have following candidates: A: Open source 1) Asterisk PBX (http://www.asterisk.org) (with longest history that almost every one knows it, now the last version using the PJSIP stack) 2) FreeSwitch (http://www.freeswitch.org) (A lot people

Hi, i have a samba 4.0 server with bind DNS. I has two configured interfaces 192.168.1.1 and 192.168.2.1. The second nic is just for backup purposes i don't want it to be accessed by clients. At the moment samba dnsupdate push the .1.1 and the .2.1 address as server addresses. Windows clients in the 1.0/24 try to access 2.1 which does not work. How can i tell samba not to puch the second

Hi Please check out this product http://www.sangoma.com/products/hardware_products/data_networking/a301.html Does it work on Asterisk or Freeswitch ? Do Telcos provide an E3 connection ? One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does

This is slightly OT in that it isn't specifically *-related, but I was wondering what the members of the * user community felt about these two subjects. I've been perusing the OpenPBX.org mail list and the current hot topic is the fact that their project has come to a grinding halt. They are concerned that they don't have enough people working on their project. They feel that * has

I am using AMI to Originate a call. I have been able to get the caller id number to be passed through. However, I can't get the name to be passed through. A person I'm working with has a Freeswitch that is able to pass the caller id name and number through for their call. Comparing the Asterisk SIP messages to the Freeswitch SIP messages, I have narrowed the problem down to a single

I have a list of txt files that I want to convert into .rdata R data object. filenames 1. "./file1.txt" 2. "./file2.txt" 3. "./file3.txt" 4. "./file4.txt" 5. "./file5.txt" 6. "./file6.txt" 7. "./file7.txt" 8. "./file8.txt" 9. "./file9.txt" 10. "./file10.txt" I saved these files as for ( i in

All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and

Hi I've been trying to install RBGL and several other packages in gR task view on Fedora11 64 bit without success My guess is that it is due to boost (fedora is at boost 1.37 which is a few minor versions behind 1.40) and very recent g++ Has anyone managed to install the gR task view or RBGL in particular? Thanks Cheers Peter ======== [root at SPH-PH-428-04P ~]# gcc --version gcc (GCC)