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brent at texascountryt... Guest
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Posted: Tue Jun 07, 2016 9:48 am Post subject: [asterisk-users] Delay after Answer |
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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 Guest
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Posted: Tue Jun 07, 2016 10:04 am Post subject: [asterisk-users] Delay after Answer |
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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
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Posted: Tue Jun 07, 2016 1:01 pm Post subject: [asterisk-users] Delay after Answer |
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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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brent at texascountryt... Guest
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Posted: Tue Jun 07, 2016 1:55 pm Post subject: [asterisk-users] Delay after Answer |
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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
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Posted: Wed Jun 08, 2016 1:54 am Post subject: [asterisk-users] Delay after Answer |
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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
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Posted: Wed Jun 08, 2016 2:08 am Post subject: [asterisk-users] Delay after Answer |
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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
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Posted: Wed Jun 08, 2016 2:11 am Post subject: [asterisk-users] Delay after Answer |
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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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