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>
Are you sure *nslookup <hostname> *command is returning as expected? Also check the output of the below command.>> hostname && hostname -s && hostname -fOn 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 >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by <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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160608/cf491042/attachment.html>
Are you using stun? I have seen that when using stun ?????? 8 ????? 2016 09:54,? "Faheem Muhammad" <faheem2084 at gmail.com> ???:> > > 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 >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by <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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160608/eb30891c/attachment.html>