Re: [asterisk-users] Delay after Answer

2016-06-08 Thread Israel Gottlieb
Another thing i would check is encryption is disabled on the snom
בתאריך 8 ביוני 2016 10:07,‏ "Israel Gottlieb"  כתב:

> Are you using stun? I have seen that when using stun
> בתאריך 8 ביוני 2016 09:54,‏ "Faheem Muhammad"  כתב:
>
>>
>>
>> Are you sure *nslookup  *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 --
 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

Re: [asterisk-users] Delay after Answer

2016-06-08 Thread Israel Gottlieb
Are you using stun? I have seen that when using stun
בתאריך 8 ביוני 2016 09:54,‏ "Faheem Muhammad"  כתב:

>
>
> Are you sure *nslookup  *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 --
>>> 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

Re: [asterisk-users] Delay after Answer

2016-06-08 Thread Faheem Muhammad
Are you sure *nslookup  *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  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 --
>> 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

Re: [asterisk-users] Delay after Answer

2016-06-07 Thread Brent Davidson
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 
> 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




-- 
_
-- 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

Re: [asterisk-users] Delay after Answer

2016-06-07 Thread Faheem Muhammad
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 
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
>
-- 
_
-- 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

Re: [asterisk-users] Delay after Answer

2016-06-07 Thread Darryl Moore
I've seen this sort of thing where a DNS server is programmed in 
resolv.conf but is not accessible over the network. Threads get blocked, 
and you have to wait for the DNS query to timeout.



On 16-06-07 10:48 AM, Brent Davidson 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

[asterisk-users] Delay after Answer

2016-06-07 Thread Brent Davidson
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