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[asterisk-users] Delay after Answer


 
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brent at texascountryt...
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PostPosted: Tue Jun 07, 2016 9:48 am    Post subject: [asterisk-users] Delay after Answer Reply with quote

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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darryl at moores.ca
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PostPosted: Tue Jun 07, 2016 10:04 am    Post subject: [asterisk-users] Delay after Answer Reply with quote

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:

Quote:

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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faheem2084 at gmail.com
Guest





PostPosted: Tue Jun 07, 2016 1:01 pm    Post subject: [asterisk-users] Delay after Answer Reply with quote

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@texascountrytitle.com (brent@texascountrytitle.com)> wrote:
Quote:

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

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To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
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brent at texascountryt...
Guest





PostPosted: Tue Jun 07, 2016 1:55 pm    Post subject: [asterisk-users] Delay after Answer Reply with quote

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:

Quote:
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@texascountrytitle.com (brent@texascountrytitle.com)> wrote:
Quote:

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



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faheem2084 at gmail.com
Guest





PostPosted: Wed Jun 08, 2016 1:54 am    Post subject: [asterisk-users] Delay after Answer Reply with quote

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@texascountrytitle.com (brent@texascountrytitle.com)> wrote:
Quote:

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:

Quote:
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@texascountrytitle.com (brent@texascountrytitle.com)> wrote:
Quote:

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 [/url][url=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
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isrlgb at gmail.com
Guest





PostPosted: Wed Jun 08, 2016 2:08 am    Post subject: [asterisk-users] Delay after Answer Reply with quote

Are you using stun? I have seen that when using stun בתאריך 8 ביוני 2016 09:54,‏ "Faheem Muhammad" <faheem2084@gmail.com (faheem2084@gmail.com)> כתב:
Quote:



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@texascountrytitle.com (brent@texascountrytitle.com)> wrote:
Quote:

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:

Quote:
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@texascountrytitle.com (brent@texascountrytitle.com)> wrote:
Quote:

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 [/url][url=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
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isrlgb at gmail.com
Guest





PostPosted: Wed Jun 08, 2016 2:11 am    Post subject: [asterisk-users] Delay after Answer Reply with quote

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

Are you using stun? I have seen that when using stun בתאריך 8 ביוני 2016 09:54,‏ "Faheem Muhammad" <faheem2084@gmail.com (faheem2084@gmail.com)> כתב:
Quote:



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@texascountrytitle.com (brent@texascountrytitle.com)> wrote:
Quote:

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:

Quote:
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@texascountrytitle.com (brent@texascountrytitle.com)> wrote:
Quote:

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 [/url][url=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

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