[asterisk-users] Problems during calls

[Aksel Celasun]

Thank you for replying also,

I will as you and Zeeshan suggest, look at the firewall issue first, i have been suspecting
network issue, because i cannot see anything in the log, so again thanks!


Best regards


Aksel

________________________________
Fra: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] på vegne av Sammy Govind [govoiper at gmail.com]
Sendt: 19. oktober 2011 08:48
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Problems during calls

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 <aksel at abacus-it.no<mailto:aksel at abacus-it.no>> 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: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com> [mailto:asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] 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<http://visionvoip.com>
Blog - ilovetovoip.com<http://ilovetovoip.com>

On 18/10/2011 10:02, Aksel Celasun wrote:
Thank you for replying


My sip.conf is set to no on canreinvite





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