similar to: RTP traffic not being forwarded

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster debugging ? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090629/5e160c92/attachment.htm

I have an Asterisk 11.5 system, using SIP Realtime and operating as a ITSP. One of my customer's endpoints is a NetVanta 7100 PBX system that has a SIP trunk connection to my Asterisk box. The NV 7100 has a public IP on it that doesn't have any NAT between it and my Asterisk system. When the customer transfers a call from one handset to a voicemail box, the NV 7100 sends a RE-INVITE to

Hi, I've been working on migrating my asterisk from zap to sip (due to compatibility issues between my TDM400P and my Hauppauge PVR500). I've purchased a Linksys SPA-3102 and a Siemens Gigaset SL75 WLAN (wireless SIP phone). I managed to get it all working with my asterisk 1.4.0 installation, but I'm seeing some interesting things with the canreinvite option that I can't explain,

Hi, Is it possible to put some kind of bridge which will do traffic shaping/prioritising between my 6 external IP addresses and my PPPoA modem interface? My other option is to put some kind of device at the edge of all my networks to shape the traffic in/out. I'd rather do it in one box if possible? thanks Mike

> I don't think it's an endpoint issue. I think the SIP packet headers get > over-written by the tunnel (openvpn) protocol. I'd be rather astonished if OpenVPN itself were responsible for this. As far as I know, OpenVPN doesn't do higher-level-protocol rewriting of any sort. It just provides the "bit pipe" through the tunnel. I'd suggest several other

Hi everybody, I am operating a tinc network with nearly 200 peers connected over the internet. Some peers are permanently connected and offer a public, fixed IP ("servers") while others are behind NAT firewalls ("clients") and connect to the former primarily. Unfortunately, sometimes (~ once a day) the traffic on the ethernet links seems to explode way beyond whats normal

We have a situation here that is a real mystery. Our MRTG on our outgoing router and a firewall server that protects our web servers is showing a spike every six hours. I can't find the server behind the firewall that is generating such an extreme amount of packets, even though I've looked through the crontabs of nearly all servers, performed "ps" variations, and other

On my new CentOS 6, KDE 4, running WireShark I see what appears to be frequent nonsensical DNS queries, for example: "settings-personal.desktop" and "settings-system.desktop". The DNS response is always:"No such name". Do tell! These appear especially when I click on things on the KDE menus. On my old CentOS 5 box, on the same LAN, I see no such thing. I note

Hi everyone, I am sending out a multicast page using the following in my dialplan: Page(MulticastRTP/linksys/${MULTICAST_IP_ADDR}:6061/${MULTICAST_IP_ADDR}:6061,q) Everything works great, but I had a question about SIP and SDP: Should I be seeing a SIP/SDP message from the asterisk server containing media information and the multicast IP address? On wireshark, I see SIP/SDP from the admin

I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port >= 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm

Hey all! I recently tried the dtmf_mode "auto_info" on my setup to support endpoints that only understand SIP INFO as a fallback. My setup is the following: Endpoint A (RFC4733) --> Asterisk <-- Endpoint B (SIP INFO) Both are configured with "auto_info" dtmf_mode in pjsip.conf. What I ran into is, that DTMF sent from endpoint A to endpoint B is additionally sent via

Hello. Is someone here using VoIPmonitor? I am using just the sniffer and I found some pcap files that contain some odd streams. For example, I have a file with 3 streams, but the weird stuff is that 2 streams are the same (e.g., have the same source address and port and same destination address and port). Example: "Source Address","Source Port","Destination

with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN

hi, i have following topology PSTN - Asterisk ---- internet -----  router - jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router - public IP/private IP (NAT) jssip client - private IP - sip over websocket to Asterisk PJSIP ~30% of calls has problem with no audio. reason is that Asterisk is sending RTP to private IP of jssip

Hi everyone, I'm wondering I would accomplish the following: I want to dial several SIP extensions simultaneously, HOWEVER, for different times (say ext 10 for 15 sec and ext 11 for 30 sec), and potentially with different headers (such as ALERT_INFO) and codecs for each extension. Obviously whoever picks up first gets the call. After the longest timeout expires (30 sec in this example) I want

Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone

I am trying to determine the root of an issue I am having. How can I watch traffic destined to a specific port on my CentOS 5.1 box to see if its even hitting it? It would be udp traffic. Thanks! jlc -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20080520/d5066957/attachment-0005.html>

Hi Guys, My Server at Mosso has more than doubled in bandwidth over the last week. Mosso recommended getting an analyzer tool to tell me why. What can you guys recommend, easy to install and informative? Thanks! -Jason

Dear Asterisk friends, Please help me to clarify my doubt. After monitor SIP and RTP traffic with Wireshark. I found that both SIP and RTP traffic between 2 sip clients must be passed through Asterisk. Is it possible that 2 sip clients connect with each other directly for RTP session after sip session completed ? Thank you, Kamonwat -------------- next part -------------- An HTML