Another thing i would check is encryption is disabled on the snom בתאריך 8 ביוני 2016 10:07, "Israel Gottlieb" <isr...@gmail.com> כתב:
> Are you using stun? I have seen that when using stun > בתאריך 8 ביוני 2016 09:54, "Faheem Muhammad" <faheem2...@gmail.com> 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 < >> br...@texascountrytitle.com> 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 < >>> br...@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> >>>> 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