On Mon, Aug 9, 2021 at 8:32 AM Jerry Geis <jerry.geis at gmail.com> wrote:> > > On Mon, Aug 9, 2021 at 8:11 AM Jerry Geis <jerry.geis at gmail.com> wrote: > >> >> >> On Mon, Aug 9, 2021 at 7:57 AM Jerry Geis <jerry.geis at gmail.com> wrote: >> >>> >>> >>> On Sun, Aug 8, 2021 at 3:18 PM Jerry Geis <jerry.geis at 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 at 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 at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20210809/26f0bba6/attachment.html>
On Monday 09 August 2021 at 17:05:42, Jerry Geis wrote:> Well my issue has changed now. I have dropped the 3382. Changed back to > 3381. So I am registering 3381 to the other server.That makes more sense to me, at least.> 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 OKI think eth1 and your routing table are not relevant to this.> 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?Commonly, because of firewalling and/or NAT. Given that your client 10.35.229.11/24 and the server 10.35.229.5/24 are both on the same subnet, it's not going to be a NAT problem, so I would look at the firewall rules, both on your machine and the one you are connecting to. (Please do tell us if the client and server are not connected directly through a switch as I have assumed here, and there's possibly something more complicated going on.) You want to look for firewall rules which will allow UDP in both directions on ports 10000 - 30000 (typically, may vary a bit, but something like that), or alternatively, look for any rules which would block this, and remove them. If that doesn't appear to be the problem, do a packet capture of your SIP traffic and look for the Invite and the reply, each with the SDP payloads, to find out what IP addresses and port numbers the client and server are advertising to each other. The only other thing I can think of right now is codec compatibility. Antony. -- I know I always wanted to be somebody, but I guess I should have been more specific. Please reply to the list; please *don't* CC me.
You could switch to PJSIP and avoid most of this silliness. I love Asterisk, but the peer/user/friend model in chan_sip is simply terrible. PJSIP is different so there is a learning curve, of course. On 8/9/21 11:05 AM, Jerry Geis wrote:> > > On Mon, Aug 9, 2021 at 8:32 AM Jerry Geis <jerry.geis at gmail.com > <mailto:jerry.geis at gmail.com>> wrote: > > > > On Mon, Aug 9, 2021 at 8:11 AM Jerry Geis <jerry.geis at gmail.com > <mailto:jerry.geis at gmail.com>> wrote: > > > > On Mon, Aug 9, 2021 at 7:57 AM Jerry Geis <jerry.geis at gmail.com > <mailto:jerry.geis at gmail.com>> wrote: > > > > On Sun, Aug 8, 2021 at 3:18 PM Jerry Geis > <jerry.geis at gmail.com <mailto:jerry.geis at 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 at 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 at 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 >-- http://help.nyigc.net/
On Mon, Aug 9, 2021 at 11:05 AM Jerry Geis <jerry.geis at gmail.com> wrote:> > > On Mon, Aug 9, 2021 at 8:32 AM Jerry Geis <jerry.geis at gmail.com> wrote: > >> >> >> On Mon, Aug 9, 2021 at 8:11 AM Jerry Geis <jerry.geis at gmail.com> wrote: >> >>> >>> >>> On Mon, Aug 9, 2021 at 7:57 AM Jerry Geis <jerry.geis at gmail.com> wrote: >>> >>>> >>>> >>>> On Sun, Aug 8, 2021 at 3:18 PM Jerry Geis <jerry.geis at 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 at 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 at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20210811/654cbec2/attachment.html>