On Mon, Aug 9, 2021 at 11:05 AM Jerry Geis <jerry.g...@gmail.com> wrote:
> > > On Mon, Aug 9, 2021 at 8:32 AM Jerry Geis <jerry.g...@gmail.com> wrote: > >> >> >> On Mon, Aug 9, 2021 at 8:11 AM Jerry Geis <jerry.g...@gmail.com> wrote: >> >>> >>> >>> On Mon, Aug 9, 2021 at 7:57 AM Jerry Geis <jerry.g...@gmail.com> wrote: >>> >>>> >>>> >>>> On Sun, Aug 8, 2021 at 3:18 PM Jerry Geis <jerry.g...@gmail.com> wrote: >>>> >>>>> I am not using a SIP trunk as I normally do. >>>>> >>>>> I have an extensions 3382 setup that my server registers to the other >>>>> SIP system. >>>>> When the other system calls 3381 on my system I am getting this error: >>>>> >>>>> [Jul 27 10:08:50] WARNING[89791][C-00000068] chan_sip.c: username >>>>> mismatch, have <3381>, digest has <8124> >>>>> [Jul 27 10:08:50] NOTICE[89791][C-00000068] chan_sip.c: Failed to >>>>> authenticate device "USCOL TEST" <sip:XXXX@IP>;tag=1c1947164290 for >>>>> INVITE, code = -2 >>>>> >>>>> How I allow this ? I want to allow any SIP call to 3381. >>>>> Using Astering 18.4.0 >>>>> >>>>> Thanks, >>>>> >>>>> Jerry >>>>> >>>> >>>> Sure here it is: >>>> [general](+) >>>> register => 3382:XX@IP/3382 >>>> >>>> ; Description: Connection to PBX >>>> [3382] >>>> type=friend >>>> defaultname=3382 >>>> defaultuser=3382 >>>> secret=XX >>>> dtmfmode=RFC2833 >>>> host=IP >>>> description=Connection to PBX >>>> context=incoming >>>> rtptimeout=60 >>>> rtpholdtimeout=60 >>>> rtpkeepalive=60 >>>> callerid=3382 >>>> qualify=no >>>> canreinvite=no >>>> nat=never >>>> disallow=all >>>> allow=ulaw >>>> allow=alaw >>>> allow=gsm >>>> >>>> Thanks >>>> Jerry >>>> >>>> >>> > What's the association between 3381 and 3382? >>> >>> 3381 is the number they want to dial into my asterisk. 3382 is the >>> registered extension to their system. >>> >>> Jerry >>> >>> >>> >>>> >>>> >>> >> >You register as 3382. That means that if someone on their system dials >> 3382, >> >your Asterisk server gets the call. >> >> >> I think at first I was only using 3381. That was the extension I >> registered. There was no 3382. Something was going wrong there also. >> (Might have been a similar error), >> and I could not get that to work either. >> >> Jerry >> > > > Well my issue has changed now. I have dropped the 3382. Changed back to > 3381. So I am registering 3381 to the other server. > The other server is 10.35.229.5. My IP is 10.35.229.11. > I have two network cards. > > 10.35.229.11 is Eth0 > 192.168.1.60 is Eth1 > > route looks OK > route -n > Kernel IP routing table > Destination Gateway Genmask Flags Metric Ref Use > Iface > 0.0.0.0 192.168.1.1 0.0.0.0 UG 0 0 0 > eth1 > 10.35.229.0 0.0.0.0 255.255.255.0 U 0 0 0 > eth0 > 169.254.0.0 0.0.0.0 255.255.0.0 U 1002 0 0 > eth0 > 169.254.0.0 0.0.0.0 255.255.0.0 U 1003 0 0 > eth1 > 192.168.1.0 0.0.0.0 255.255.255.0 U 0 0 0 > eth1 > > The issue is that the call comes in but the user hears no audio. > There is any crazy networking going on - why would the user not hear audio > ? > Thanks > > Jerry > Hello All, I got more information about the "no audio". The incoming call is from 10.37.229.5 - I have two network cards in the box. 10.35.229.11 eth0 192.168.1.60 eth1 When I noticed the incoming address was 10.37.229.5 I thought the audio packets are sending out the default route of eth1. SO I tried to add a route: route -n Kernel IP routing table Destination Gateway Genmask Flags Metric Ref Use Iface 0.0.0.0 192.168.1.1 0.0.0.0 UG 0 0 0 eth1 10.35.229.0 0.0.0.0 255.255.255.0 U 0 0 0 eth0 10.37.229.0 0.0.0.0 255.255.255.0 U 0 0 0 eth0 169.254.0.0 0.0.0.0 255.255.0.0 U 1002 0 0 eth0 169.254.0.0 0.0.0.0 255.255.0.0 U 1003 0 0 eth1 192.168.1.0 0.0.0.0 255.255.255.0 U 0 0 0 eth1 But I am still not getting audio. Anything else I might try ? Thanks Jerry
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users