similar to: Asterisk stops sending RTP packets to ethernet interface

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Hello All, Currently my setup uses Xlite and Asterisk and i found that all the RTP voice packets are transfered via the asterisk server from one xlite to another. Is there any possibility that we can make all the RTP Packets to be transfered directly between the two clients once the connection is established?. Any one please help me. Thanks and Regards, Senthil Murugan.V

Hi there, i did some tests with two Cisco 7912G phones (SIP stack) yesterday. With both ethereal and tcpdump listening on the Asterisk-Server's NIC, it came up that all RTP packets were doubled, with some small but almost constant delay (~460 us). The setup is 7912G <--> ASTERISK <--> 7912G The tcpdump output shows RTP traffic ASTERISK --> 7912G: 000000 IP $ASTERISK.17944

Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where the phone did not send rtp keepalives when on mute (resulting in disconnect from tech support hold and concalls) A side effect seems to be that Asterisk pops the following warning on the console... Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short Grandstream say they are not sure what it is but

Hi, I am try to configure Asterisk as PBX system with two interfaces as shown below. One interface pointing to the local subnet with a SIP phone and another interface pointing to the external ISP SIP Sever. SJPhone(X.X.141.32)<--------->(Y.Y.47.149)local-intf-|Asterisk|external- intf(Z.Z.247.106)<-------->(w.w.158.26)ISP-SIP-Server----OutsideWorld I am able to setup a call from the

Hello, What does it mean when "rtp set debug ip" shows RTP packets that have been send, but there is no audio ? There was no audio on my call in both directions, but "rtp set debug" shows that there were RTP packets send. There is no firewall active on my Asterisk server : [root at sip asterisk]# /sbin/service iptables status iptables: Firewall not running. Kind

Hello, When a call is starting, Asterisk starts sending and receiving RTP packets. Each packet has a sequence number. 1. Instead of logging everything as rtp set debug is currently doing, is there a way to only log: - the sequence number of the first received packet, - any missing or misplaced sequence number. 2. Is there a way to log RTP debug information in a specific file or send the same

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=498 Summary: RTP packets are not hitting NAT table Product: netfilter/iptables Version: linux-2.6.x Platform: All OS/Version: Fedora Status: NEW Severity: major Priority: P2 Component: NAT AssignedTo: laforge@netfilter.org ReportedBy:

Hello, My target system is : PSTN <---> Sip Provider <---IP/ADSL---> Router with fw/NAT <--- SIP/IP/Eth --> Asterisk <--- SIP/IP/Eth --> SIP Phones Asterisk is configured to keep NAT connection with SIP provider open (with qualifyfreq) so I don't have any problem (yet) with either casual incoming or outgoing calls. To work around a possible No Audio when an incoming

Hi, I'm connecting Asterisk v. 1.4.10 to Zanzibar Open IVR that acts as a SIP trunk. Since recognition didn't work correctly, I've troubleshot with Wireshark and saw that RTP stream is first send to one port on SIP trunk and then when first RTP packet arrives in opposite direction (from TTS part of Zanzibar - it's a prompt) Asterisk starts sending to the same RTP port -

Been having problems with CVS lately. I get incoming calls from an iaxcomm user, using a windows system. Asterisk stops sending data after about 30 seconds. I view this with tcpdump on the same computer. I still receive data and can hear the remote party. This problem starting sometime mid last month. I regularly build cvs and have run into this issue before. Now it's been like this for some

Hi All, I am doing some testing with the asterisk server and have been monitoring the packets exchanged during a SIP-ZAPTEL phone call. I see that the IP-ID in all of the RTP/UDP/IP packets are set to zero. After some googling, I have learnt that some of the linux implementations set the IP-ID to 0 (if the DF bit is set in the IP header) if the two hosts exchanging data are on the same subnet.

Hello, I have my provider dropping the calls after 41 seconds of not receiving any RTP from my asterisk. Obviously there is no RTP back when the caller is leaving a message in the voicemail. Is it possible to have asterisk generate some RTP packet back? Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am working on voice application. I want to identify RTP packets and set DSCP for those. Is there anyway to accomplish this task either using tc or iptables. Please help me.. Thank you -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi, I am facing a problem where for legal obligations (LI) I have to copy/mirror/forward the RTP streams for some selected call to an external address/port and I have not found a way to do it with built-in functionality. Do I miss something? The basic requirements are: * Raw RTP (no transcoding, header and payload as is) * Direction (did it arrive at asterisk or was it sent) * End

Hello All, I have installed Asterisk 1.6 with openVPN in the same machine. I have set up a VPN connection between 2 SIP clients and Asterisk using x-lite. The 2 clients connects to Asterisk. SIP signaling goes ok over the vpn tunnel. When attempting to make a call between the clients, the siganling part of the call goes well. But, when the call is set up, some RTP packets are exchanged at

> What machine are you running (namely what OS)? Debian. > I dont understand your approach. > Why running a 'streamer' behind a nat? > Not enough 'resources' to rent/ rent to buy a ded. Server? > Mean, can't expect to satisfy a lot of listeners this way. :-) I am listening the Muazkhan indications XD: > > Hi Muaz Khan, >> We are adtlantida.tv and

https://bugzilla.netfilter.org/show_bug.cgi?id=900 Summary: Bridging issue: IP packets with Multicast Ethernet Address Product: netfilter/iptables Version: unspecified Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P5 Component: bridging AssignedTo:

My phone is located behind a NAT, 172.16.0.0/21. Asterisk 16 is on a public IP. PJSIP has the config below: force_rport=yes direct_media=yes disable_direct_media_on_nat = yes direct_media_method=invite But when I send a call I see the RTP being sent to my private address, vs the public IP. This only happens when Asterisk has dialed the call to another carrier. If instead of Dial I choose

While not strictly Asterisk related this issue could certainly affect some of you: http://blog.krisk.org/2013/02/packets-of-death.html -- Kristian Kielhofner