Are you using stun? I have seen that when using stun בתאריך 8 ביוני 2016 09:54, "Faheem Muhammad" <[email protected]> כתב:
> > > 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 >
-- _____________________________________________________________________ -- 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
