[asterisk-users] Delay after Answer

[Israel Gottlieb]

Another thing i would check is encryption is disabled on the snom
בתאריך 8 ביוני 2016 10:07,‏ "Israel Gottlieb" <isrlgb at gmail.com> כתב:

> Are you using stun? I have seen that when using stun
> בתאריך 8 ביוני 2016 09:54,‏ "Faheem Muhammad" <faheem2084 at 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 <
>> brent at 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 <
>>> 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
>>>>
>>>> --
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>>>
>>>
>>>
>>> --
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>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>                http://www.asterisk.org/hello
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>
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