similar to: RTP traffic not being forwarded

Displaying 20 results from an estimated 10000 matches similar to: "RTP traffic not being forwarded"

2007 Nov 08
2
asterisk and installing chan_h323.so rpm
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
2009 Jun 29
4
how to sniff RTP and SIP traffic only
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
2013 Aug 26
1
Asterisk 11.5 not honoring RTP port change in RE-INVITE
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
2007 Feb 10
1
canreinvite problems
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,
2005 May 18
2
Traffic shaping for IAX and SIP calls through Asterisk?
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
2010 Sep 23
1
OpenVPN tunnel and one-way audio - Do I still need a SIP proxy? (bruce bruce)
> 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
2020 Mar 19
2
High tinc traffic on ethernet without tinc load
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
2012 Jun 14
4
OT - Is there a package to monitor network traffic
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
2011 Aug 31
3
CentOS 6, KDE 4: bad DNS traffic
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
2016 Apr 27
2
SIP/SDP for MulticastRTP page
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
2007 May 01
5
OT: Capture Asterisk traffic
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
2018 Sep 26
2
chan_pjsip: DTMF mode "auto_info" on endpoints
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
2017 Aug 14
2
VoIP monitor and multiple RTP streams
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
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
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
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
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
2005 Mar 16
2
Dial multiple extensions, but different variables/timeouts
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
2010 Oct 26
2
No media being sent in SIP call
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
2008 May 21
4
Show IP Traffic on a port
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>
2009 May 23
1
Traffic Analyzer
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
2005 Jul 01
4
asterisk showing more than once on ps
Guys. Anybody know why sometimes on some servers Asterisk shows more than once while doing a ps? [root@server2 akrall]# ps -ax|grep asterisk 20555 ? S 0:00 /bin/sh /usr/sbin/safe_asterisk 20557 ? S 0:00 asterisk -vvvg -c 20558 ? S 0:00 asterisk -vvvg -c 20560 ? S 0:00 asterisk -vvvg -c 20561 ? S 0:00 asterisk -vvvg -c 20562 ? S