Are you sure *nslookup <hostname> *command is returning as expected? Also check the output of the below command. >> hostname && hostname -s && hostname -f
On Tue, Jun 7, 2016 at 11:54 PM, Brent Davidson <[email protected] > wrote: > 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 < > [email protected]> 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> >> 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 >> > > > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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
