Re: [asterisk-users] Troubleshooting TDMs (Packet capture like debugging)
Hi All, Many thanks for the info, I am now finding the time to look back into this again. I have seen this page, which indicates that dahdi_pcap was pushed into the dahdi driver from version 2.4.0; https://issues.asterisk.org/jira/browse/DAHTOOL-49?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel I have also seen this page regarding installing dahdi with dahdi_pcap; http://lists.digium.com/pipermail/asterisk-ss7/2011-September/004519.html All seems simple enough. I have no test systems with E1s/PRIs connected. I have a particular production box though which is very quite and I can schedule a maintenance window in which I can fiddle with it. If I killed the box it's not really a problem as other boxes can pick up the load. So my question is this; Looking on this quiet box I see /usr/src/dahdi-linux-2.4.1.1 which doesn't include dahdi_pcap. I am thinking of checking out the latest version and compiling it to test dahdi_pcap. This box has Asterisk 11.3.0 running on Ubuntu Server 12.04.2 LTS. If I re-route traffic away from this box (having never installed DAHDI before) can I "simply" shutdown Asterisk, checkout the latest SVN, compile, reload the kernel module and restart Asterisk? What can go wrong here? Is this risky, or is it really that simple? Cheers, James, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk RTP Questions
Heh, should have guessed it would be you that replied Gareth ;) Sorry yes, this box is on public IP with no NAT as is the upstream providers box (or so they say). So we have had audio cease outbound towards the provider. We have a couple of volunteer customers who are being routed via this new test upstream. It's very difficult (basically impossible!) to replicate the failure it's so infrequent. Looking at PCAPs between us and the upstream we stop sending them audio for example and then a little while later the call drops. Without PCAPs between us and the customer at the same time I can't say why we stopped sending audio (where we receiving any from the customer, did their connection drop for example). We have also had the reverse where we stop receiving audio then a short period later, SIP BYE from us to them! I have read up on rtpkeepalive and rtptimeout. I will put this to one side for now until we have a direct connect to the new test provider there are to many variables in the equation. Thanks for your input though Gareth!. Kind regards, James. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk RTP Questions
Hi All, I have some questions regarding RTP and Asterisk; I am trialling a new SIP upstream provider. We connect to them over the Internet at present which I know is not ideal, but we are just testing at present. During the trials we have had an issue where we have had one way audio between us and the provider after the call was successfully set up and bidirectional audio has been already flowing (so at some point during an existing call, two way audio has dropped to one way audio). I am running a constant PCAP which I sent back to the provider. They have said that the latency has increased or fluctuated to the extent that RTP as stopped sending audio in one direction (because of our test peering over the Internet). Weather this is true or not is a separate issue, what I want to know is; What is the maximum delay RTP will tolerate one way (Does Asterisk have a limit too)? Can this be tuned (increased or decreased) within Asterisk (I'm thinking of DSL customers where we may have this issue between our PBXs and the customer)? How can I monitor for such an effect? Does anyone else have any / or had any issue like this? Kind regards, James. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP from pcap file
Hi all, Many thanks to all for your helpful suggestions. I have compiled rtpbreak. That is exactly what I needed, easily scripted with sox, for auto conver PCAPs to WAV files. Many thanks all, especially Gianluca! Cheers, James. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP from pcap file
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. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Troubleshooting TDMs (Packet capture like debugging)
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 I can see the signalling (like the SIP message) and the audio stream (like the RTP stream) in my packet captures? If it helps, the end goal is to create something like the packet captures I am making so I can see the control signals and audio streams (in and out) for troubleshooting one way audio issues for example. So, am I sending audio to the TDM provider, are they sending it to me, have we both signalled correctly to start/stop sending audio, etc. Many thanks, James. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
