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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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* > * > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160607/1acb1d28/attachment.html>
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 > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160607/709ab7e5/attachment.html>
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 <mailto: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* > * > > -- > _____________________________________________________________________ > -- 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 > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160607/b8d9b71d/attachment.html>