Hi, Call getting silenced in the middle definitely point to RTP but I think the redialling part should be considered as well. I think that Phones are loosing registrations or like Zeeshan mentioned could be getting blocked by firewall - Asterisk server's firewall as well as any other firewall in front of server should be inspected for sessions/connections limit etc.
-- Regards, Sammy On Wed, Oct 19, 2011 at 12:27 AM, Aksel Celasun <[email protected]> wrote: > Thank you for the reply.**** > > ** ** > > ** ** > > The Asterisk is behind a firewall, but not in a dmz, been thinking of > placing it in a dmz soon, maybe that will solve the problem.**** > > Or else, I will try your guide with wireshark.**** > > ** ** > > Thank you very much.**** > > ** ** > > ** ** > > Best regards**** > > ** ** > > Aksel**** > > ** ** > > *Fra:* [email protected] [mailto: > [email protected]] *På vegne av* VisionVoIP > *Sendt:* 18. oktober 2011 16:31 > > *Til:* Asterisk Users Mailing List - Non-Commercial Discussion > *Emne:* Re: [asterisk-users] Problems during calls**** > > ** ** > > I can only make another guess. If your system is behind a firewall, try > adding 'insecure=invite' in your sip.conf's general section. > > > To troubleshoot such cases, do a tcpdump trace like this: > > 1. Run tcpdump on your server before making a call. Use command "tcpdump > port 5060 -s0 -w dumpfile.pcap". > 2. When you notice the silence problem, hangup, and stop the trace using > CTRL+C. > 3. Copy the dumpfile.pcap to a computer with Wireshark installed. > 4. Open this file in Wireshark and follow my blog at > http://ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/ > 5. Given that you know some basics of how VoIP works over SIP, the > wireshark graph will tell you if RTP was still flowing when it was silent. > It probably is, but to which IP address. > > My guess is your RTP, i.e. voice date, starts flowing towards some wrong IP > address, or stop flowing, or is blocked by the router. > > A good solution is to put your Asterisk server in DMZ mode. > > There can be many other guesses, but the above is a good start. > -- > > Zeeshan A Zakaria > > PBX - visionvoip.com > Blog - ilovetovoip.com > > On 18/10/2011 10:02, Aksel Celasun wrote: **** > > Thank you for replying**** > > **** > > **** > > My sip.conf is set to no on canreinvite**** > > **** > > **** > > **** > > ** ** > > -- > _____________________________________________________________________ > -- 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 >
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