similar to: Asterisk RTP Questions

Howdy all, Does anyone know of a niffty CLI tool for Linux that can take a PCAP file that was created on a SIP PBX for example, and then dump the payload of the various RTP streams in there into seperate files so I can listen to them? I can go this graphically with Wireshark, but I'd like to script it for automation. Cheers, James.

So I went to System > Administration > Server Settings > Samba and added a folder to share and initially set it to guest access to get things going. I couldn't access this one and only share on this server with guest access from either my Windows laptop (XP Pro SP3) or a Leopard server (10.5.8) so I changed the settings to use user authentication and added my local user details

So on a virtual server the root password was no longer working (as in I couldn't ssh in anymore). Only I and one other know it and neither of us have changed it. No other account had the correct privileges to correct this so I'm wondering, if I had mounted that vdi as a secondary device on another VM, browsed the file system and delete /etc/shadow would this have wiped all users passwords

Hi List, We have a CentOS VPS running a web site in a DC far away. The chap that dev's this site told me he couldn't SFTP in yesterday, his password was being rejected (I went to his desk to confirm and saw it was telling him the password was incorrect but neither him nor me had changed it and we are the only two with access to this VPS). So I logged in as root and reset his password, be

Hi,all I configured two Asterisk PBXs: 1.4.X and 1.6.X. All relevant ports are opened, externip is configured in sip.conf. I think, that all relevant configurations are checked. But, no voice hear between local and remote extension. What I need to check, configure in router and PBX for resolving this issue ? How I can to install and configure STUN server ? Thanks, Oleg . -------------- next part

I have some questions on how I can perform the following actions from the terminal in CentOS 5.5 final. My Google searches aren't helping and I keep seeing references to the command 'useradd' but this isn't present on my box; -How to add a new user? $ useradd -bash: useradd: command not found (This is the same for my normal user and when logged in as root) -How to set/change an

Dears; I am facing a problem in disconnecting the calls, it is related to the rtptimeout (disconnecting if there is no RTP packets from both sides). My Asterisk Box is behind NAT but there is a static real IP address at the ADSL router. We call from the Mobile to the PSTN analogue numbers which are connected to Asterisk Analogue card (the telephone lines are analoge), and then we dial the

Listee's... I have a 5.5 box and I have formatted a partition as ext4 however I can't mount it, the mount command is telling me its an unknown file system type. fsck tells me the partition is a clean ext4 partition but I can't mount it. Some reasearch has lead me to believe the problem is that ext4 isn't enabled in the kernel by default in 5.5 (this was originally a 5.2 box that

Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both boxes. I need to connect both PBXs with E1/R2 and UTP cable. What are the requirements to deploy the UTP cable ??? Straight-through or crossover ??? What are the pinouts in both peers ??? Thanks a lot, Alejandro

Shadies and Mentlemen; I am trying to be green and put our backup servers to sleep during the day and have them wake on LAN and fire back up at night for our nightly backups as "sleep" is a sort of low power usage mode. (At this point I would be curious to know the different levels of sleep, what can I achieve? Does my server just drop into a low power state, or can I stop the hard

I'm stuck trying to work this one out and my Google skills are apparently lacking today. This is a test bash script; #!/bin/bash do something | tee a.log if [ $? -ne 0]; then echo broken fi The problem with this script is $? contains the exit value of the tee command, but I want to check the exit value of whatever command I put in place of 'do something'. How can I achieve this

Hello. I am going through the documentation and trying to find if asterisk can help me in my case. It is quite difficult to find answer because I do not know the exact question. I have two location. Each in different country. Both locations have Siemens HiPath - different type and software. I can not use card that would allow me to connect those PBXs using SIP. But I have some free ISDN and

Hi All, I am looking for a way to troubleshoot issues with TDM (E1) trunks with a provider. Currently with SIP trunks I am using tcpdump to perform packet captures between our gateways and the SIP providers IPs, capturing traffic on all ports, to include both the SIP messages and the RTP stream. How can I achieve a similar result on TDM links connected to TDM cards in Asterisk servers, where by

Hi, I'm buying an interface for my Toshiba DKT424 to connect to *. Initially it is a trial to route internal calls over our wan. Think of it as a tie trunk between 2 pbxs. I expect to get PRI from the telco our layout will be like this: telco_T1_PRI to * to various PBX or TDM or IAX to * to other pbxs My question is, is there benefits to connecting with PRI over a standard T1 within my

Hi List, I suppose this isn't strictly CentOS but I'm talking about some 5.5 servers so I thought someone else here may have had the same issues; We power on servers at night via IPMI and shut them down the same way for automation. The problem I'm facing is that the servers are shutdown in about 2 seconds. IPMI provides remote access to the power features as we know so I don't

Hey Listees, I have a question (hopefully quite simple) about the system V init scripts in /etc/init.d. I have an app installed and for some reason it had no system V init script. It has been installed a long while now so I can't quite remember whats going on but basically I remember I wrote the init script my self and the app is enabled as a service which chkconfig to run at system levels 3,

Hi to all, i already searched the archive without finding a solution to my problem. I have asterisk installation 1.2.18 to support multiple virtiual PBXs. I use SIP peer in the format <ID>-<EXT> to let every virtual PBX to share the same numbers of EXT. Ex. (PBX ID 10 Extensions) 10-101 10-102 10-103 (PBX ID 20 Extensions) 20-101 20-102 20-103 I use some rules in the dialplan to

We're getting ready to ditch our hosted virtual PBX for an Asterisk solution and I've got a couple questions that probably come from experience; we're looking to host two PBXs on the same box if the PBXs are identified by different DID numbers, and I was curious if: 1. Is it possible to have duplicate extensions between the two PBXs? Eg: 555-1111x100 and 555-2222x100 on same

I must be doing something stupid, but I can't figure it out. I have three PRI lines connected to Asterisk, one from the phone company, and two more connected to PBXs. Asterisk uses the incoming DID information to decide which PBX to route the call to. Should be simple. Asterisk is clearly getting the caller id info from the phone company: -- Accepting call from '512345xxxx'

I have several Linux machines some running on really old hardware and some on brand new, some run old distros (RedHat 6) and some new (FC3 or CentOS). All of them experienced power failure more then once, none of them has failed to load after a reboot. BUT, Asterisk is running your PBX. Your PBX isn't your proxy server, it isn't your web server, mail server, firewall, or whatever