[asterisk-users] Delay after Answer

Brent Davidson brent at texascountrytitle.com
Tue Jun 7 13:54:33 CDT 2016

Well, I thought I had the problem solved.  Ported everything over to 
PJSip and build RDNS records for the phones and the server, but I am 
still experiencing the problem on incoming calls.


On 6/7/2016 1:00 PM, Faheem Muhammad wrote:
> I've faced the same issue. The issue was related to DNS, the reverse 
> lookup query failure caused the delay around(7-9 seconds). The purpose 
> of reverse lookup is to block IP Spoofing attacks.
> Regards,
> Faheem
> On Tue, Jun 7, 2016 at 7:48 PM, Brent Davidson 
> <brent at texascountrytitle.com <mailto:brent at texascountrytitle.com>> wrote:
>     I am having an issue with a couple of phones where they ring, but
>     there is a long delay after the phone is picked up before the
>     audio starts.
>     My setup:
>       * Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC
>       * Server is CentOS 7
>       * Quad core CPU with 16GB Ram
>       * 2 Snom 300 phones.
>       * NO NAT.  Server and phone are on the same subnet with only a
>         gigabit switch between them.
>       * Digium TDM400 analog card with 2 incoming analog PSTN lines
>     When a call comes in, the system answers, IVR plays, caller dials
>     an extension, Snom 300 rings, handset picked up.  Caller continues
>     to hear ringing for another 7 to 10 seconds.  Answerer hears a
>     click, a quick burst of audio, then silence, then another click
>     and audio is engaged.
>     I have tried both SIP and RTP debugging and there are absolutely
>     no messages indicating any timeout or retransmit.  I am at a total
>     loss.  In the past I've always been able to find an answer to
>     issues like this on my own, but this time I just don't know.  I
>     was even beginning to suspect the network switch might be bad, but
>     pinging between the server and the phones shows no packet loss and
>     0.969ms average response time.
>     What am I missing*?*
>     Thanks,
>     Brent Davidson*
>     *
>     --
>     _____________________________________________________________________
>     -- 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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160607/b8d9b71d/attachment.html>

More information about the asterisk-users mailing list