similar to: asterisk 1.4.X, T.38 and NAT

Hi, I looked for asterisk in /usr/sbin using the commands ls and find and whereis and it was not there. I know that the process is killed because when I start asterisk using the command asterisk -vvvvc it starts and then it exits and the word killed is wrote on the console. Ever time I copy a new executable to /usr/sbin either using cp command or make install it gets deleted too. Now I used

Hi, I am trying to implement call forwarding on the event that my ATA was not reachable to Asterisk, whether due to registration timeout, network interruptions between the ATA and Asterisk, or simply because the network on which the ATA resides in unreachable for any reason. I there a way of implementing such a feature in Asterisk? I have implemented CF unconditional, and CF on busy, CF on

Hi Thank you for your support. The server is actually compromised, I discovered that after making a deep trace using the audit daemon and looking for the kill signal (SIGKILL) that terminates asterisk. I discovered that there is an executable with a random name in the /boot folder that is killing and deleting asterisk !!! This executable is launched by a service in /etc/rc.d/ with the same

Hello I am going to try the new Digium Fax for Asterisk product. I'm planning to connect fax machines to Asterisk (currently 1.6.0.9) via T.38 ATAs. I'm looking at Grandstream HT502 or Linksys SPA2102 ATAs. If anyone has any experience with these devices, or other recommendations, I would be grateful if you could share your experiences. Regards Ian

Hello Asterisk users, We are having MANY but MANY problems configuring an analog fax machine to work properly on Asterisk, the first thing we do was to plug in the Fax analog machine to the FXS port of the Digium TDM410P, we set echocancel=no on zapata.conf and also faxdetect=yes on general section, but our Asterisk CRASH every time we try to send/receive fax! We are using Asterisk 1.4.21 and

Hi, I am struggling with ?a very strange issue I have been facing for the past week;I have a fresh install of CENTOS 5.11 and I have installed asterisk 1.8.32 form sources.The asterisk installation went fine but as soon as I start asterisk executable it loads everything and then after the "Ready" line the process gets killed and when I try to run it again i get: /usr/sbin/asterisk :

So far these people let me know there are going to be there, who else is going and wants to do some networking ==== Joachim Vanheuverzwijn (zoachien AT securax.org) - Attractel.com - wednesday / thursday. Tan Aksoy - Telappliant - wednesday / thursday Antoine Megalla - SAND - wednesday / thursday loic didelot - wednesday / thursday Olle E. Johansson - edvina - wednesday / thursday Marius

Anyone have any testing experience with T38 and HT-502 Grandstream? I just want to confirm that t.38 is working on this device. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110801/fea18d8c/attachment.htm>

Hi, Did anyone try to install Asterisk on the Mini-Box M300 with a Versa mini-ITX board 1GHz VIA x86 CPU? The box looks promissing, but I am not sure if Digium cards are compatible with the mother board (Versa mini-ITX) Also I am not sure if the 1GHz VIA processor can handle a Digium 24 port analog board, or an E1 digital board. If anyone had tried the Mini-Box, the processor, of the mother

Sending works but on receive it keeps failing - reporting back 'training' failure. I am using Asterisk 1.6 with T38. What should I post to the list to assist diagnoses? Michael

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

Hi All, I'm in the lab with Asterisk 1.6.1.12 and several ATA's testing T38. I hit a snag with the Grandstream HT502. It only seems to nail up a session at 9600bps. The Grandstream GXW4104 nails up consistently at 14400bps. I'm using the same equipment in the same configuration, just switching out the ATA. I have the latest firmware on each unit. Any ideas on what could cause

What I have is: * Asterisk 1.8.10.1~dfsg-1ubuntu1, * SPA112 ATA with analog fax in 1-st FXS port connected, * SIP trunk with provider supporting T.38. My network looks like this: * spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in neighbouring LANs, * Asterisk connects to the provider (80.75.130.136) via router (82.200.7.184). Router has full DNAT to Asterisk server. What happens?

Did you take a look at /var/log/syslog? Am 26.11.2014 21:08, schrieb Antoine Megalla: > Hi, > > I looked for asterisk in /usr/sbin using the commands ls and find and > whereis and it was not there. > > I know that the process is killed because when I start asterisk using > the command asterisk -vvvvc it starts and then it exits and the word > killed is wrote on the

Hello all, i have a problem on incoming call's from SIP Provider that ist going through the Asterisk to a Grandstream HT502. The first ring is executed on the HT502 propperly, but no more ring will follow. But the call can nevertheless be answered by a phone on the gateway. If i call the same Gateway through a connected second Asterisk the ringing is done well. If a call is coming

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Hello everyone. I'm trying to set up a SIP DID on a customer, which uses T.38 for faxing. Voice is working great, but I never configured anything using T.38 in Asterisk so I'm kinda lost. On my googling I found out that would be best letting the Linksys SPA8000 (for those that don't know, it's identical to the PAP2 but with 8 ports instead of 2) and the telco negotiate with each

I want to obtained the called number (aka DID) as a variable. How is this done? (Upline connectivity is via SIP provider). With verbose = 100, the called number is clearly shown as follows "Wait("SIP/[called number]-64e6", "1")" for example, but I can't find the info as to how to get it as a variable. Michael

I have a block of DID's that I ported to Vitelity about 7 days ago. The problem is if a POTS caller dials into the system, his dtmf is not heard at READ() or Background() while a prompt is played. After the prompt is finished, then dtmf is heard. I've been working with their support, but it still not resolved. SIP callers are not effected. Yesterday, I purchased a DID from

Hi all, Our network is suspected to be infected by malware by the detector in upline network. Turns out that some of our developers use 1.1.1.1 as a "pinging testing". Google comes to my knowledge that 1.1.1.1 is not a private IP anymore? Since when? Also Google says 1.1.1.1 is well-known to be used by botnet command and control host?? I've blocked it in the local gateway. Just