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phil-asterisk at tinsl...
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 Posted: Sat Apr 23, 2016 6:19 am Post subject: [asterisk-users] Incoming calls from Andrews & Arnold fa |
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I have service with both VoIPtalk.org and Andrews & Arnold (aa.net.uk).
VoIPtalk calls are unauthenticated and reach me fine, but Andrews &
Arnold calls are authenticated. The last call I successfully received
was on Tuesday afternoon. Initially, A&A were for some odd reason not
sending calls to my server, but that has been resolved. The problem now
is that the calls fail to authenticate, and are therefore rejected -
error 403 is presented to them, and I see this in Asterisk's console:
[Apr 23 11:53:19] NOTICE[27398][C-00000004]: chan_sip.c:25535
handle_request_invite: Failed to authenticate device "XXXXX XXXXXX"
<sip:XXXXXXXXXXX@voiceless.aa.net.uk>;tag=2016042311531900001
I have checked that the username and password in my config agree both
ends, and have even tried changing them.
The bulk of my calls come in on A&A, so I am obviously trying to find
out what has gone wrong. No-one else is seeing any problem. What do I
need to do to track this down?
--
Phil Reynolds
mail: phil-asterisk@tinsleyviaduct.com
Web: http://phil.tinsleyviaduct.com/
--
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jb_soft at trink.co.uk
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| Posted: Sat Apr 23, 2016 4:45 pm Post subject: [asterisk-users] Incoming calls from Andrews & Arnold fa |
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Hello Phil,
On Saturday, April 23, 2016, 12:19:15 PM, you wrote:
| Quote: | I have checked that the username and password in my config agree both
ends, and have even tried changing them.
|
| Quote: | The bulk of my calls come in on A&A, so I am obviously trying to find
out what has gone wrong. No-one else is seeing any problem. What do I
need to do to track this down?
|
I have a couple of lines with A&A, and I have not been having any
problems recently. When I have had similar problems in the past, it
has been an issue with the SIP config. I originally had a number of
contexts set up in sip.conf to handle the lines coming in (such as
[aa-line1], [aa-line2]) each with their own username and password
settings. The type=user setting was critical, because all the calls
came from the same IP address, and using type=peer caused matching
problems which resulted in authentication failures. This got too
complex to manage once I added in all the IP addresses A&A calls might
come in from. so I simplified the setup.
I now have just one context in sip.conf to handle incoming A&A calls,
with the same username for all lines, and type=peer. Calls are then
sent to extensions.conf, where the calls are directed to the correct
call-handler for the line based on the CID. Here is the setup in
sip.conf for A&A calls:
-------------------------------------------
sip.conf
========
[aa-incoming](!)
type=peer
context=aa-incoming
insecure=invite
transport=udp
disallow=all
allow=alaw
trustrpid=yes
sendrpid=yes
; IPv4 hostnames
[voiceless-1](aa-incoming)
host=a4.voiceless.aa.net.uk
[voiceless-2](aa-incoming)
host=b4.voiceless.aa.net.uk
[voiceless-3](aa-incoming)
host=c4.voiceless.aa.net.uk
[voiceless-4](aa-incoming)
host=d4.voiceless.aa.net.uk
[voiceless-5](aa-incoming)
host=e4.voiceless.aa.net.uk
[voiceless-6](aa-incoming)
host=f4.voiceless.aa.net.uk
[voiceless-7](aa-incoming)
host=g4.voiceless.aa.net.uk
[voiceless-8](aa-incoming)
host=h4.voiceless.aa.net.uk
[voiceless-9](aa-incoming)
host=i4.voiceless.aa.net.uk
[voiceless-10](aa-incoming)
host=j4.voiceless.aa.net.uk
-------------------------------------------
The trustrpid and sendrpid settings were important.
-------------------------------------------
extensions.conf (DNIDs changed)
===============
[aa-incoming]
exten => 440000000001,1,Goto(from-aa-line1,s,1)
exten => 440000000002,1,Goto(from-aa-line2,s,1)
exten => 440000000003,1,Goto(from-aa-line3,s,1)
-------------------------------------------
Hope this helps.
Julian
--
Best regards,
Julian mailto:jb_soft@trink.co.uk
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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phil-asterisk at tinsl...
Guest
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| Posted: Sat Apr 23, 2016 5:11 pm Post subject: [asterisk-users] Incoming calls from Andrews & Arnold fa |
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On Sat, 23 Apr 2016 22:45:32 +0100
Julian Beach <jb_soft@trink.co.uk> wrote:
| Quote: | Hello Phil,
I have a couple of lines with A&A, and I have not been having any
problems recently. When I have had similar problems in the past, it
has been an issue with the SIP config. I originally had a number of
contexts set up in sip.conf to handle the lines coming in (such as
[aa-line1], [aa-line2]) each with their own username and password
settings. The type=user setting was critical, because all the calls
came from the same IP address, and using type=peer caused matching
problems which resulted in authentication failures. This got too
complex to manage once I added in all the IP addresses A&A calls might
come in from. so I simplified the setup.
I now have just one context in sip.conf to handle incoming A&A calls,
with the same username for all lines, and type=peer. Calls are then
sent to extensions.conf, where the calls are directed to the correct
call-handler for the line based on the CID. Here is the setup in
sip.conf for A&A calls:
|
Actually, this is now sorted. It turns out the latest recommended
configs on the A&A wiki had peer vs. user confusion. On correcting
this, all was well.
--
Phil Reynolds
mail: phil-asterisk@tinsleyviaduct.com
Web: http://phil.tinsleyviaduct.com/
--
_____________________________________________________________________
-- 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 |
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jb_soft at trink.co.uk
Guest
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| Posted: Sat Apr 23, 2016 6:53 pm Post subject: [asterisk-users] Incoming calls from Andrews & Arnold fa |
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|
Hello Phil,
On Saturday, April 23, 2016, 11:11:29 PM, you wrote:
| Quote: | Actually, this is now sorted. It turns out the latest recommended
configs on the A&A wiki had peer vs. user confusion. On correcting
this, all was well.
|
I'm glad you found it. It look me a while to track down that problem
when I had it.
The one that was hardest for me to track down was a slight mis-match
between the RTP ports in Asterisk and the corresponding ports open on
a firewall, which resulted in about 1 in 10 calls having no audio!
Doh!
--
Best regards,
Julian mailto:jb_soft@trink.co.uk
--
_____________________________________________________________________
-- 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 |
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