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.g...@gmail.com
<mailto:jerry.g...@gmail.com>> wrote:
On Mon, Aug 9, 2021 at 8:11 AM Jerry Geis <jerry.g...@gmail.com
<mailto:jerry.g...@gmail.com>> wrote:
On Mon, Aug 9, 2021 at 7:57 AM Jerry Geis <jerry.g...@gmail.com
<mailto:jerry.g...@gmail.com>> wrote:
On Sun, Aug 8, 2021 at 3:18 PM Jerry Geis
<jerry.g...@gmail.com <mailto: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
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