Thomas Peters
2019-Feb-27 14:16 UTC
[asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM
Thanks for the reply John.
About 85-90% of what this box has to do is just handle calls, but it also has
options to transfer calls to the main phone system, which up to now has been
another asterisk box. For example, you can hit 6 to be transferred to the Lost
& Found Department.
I do have allowguest set to “yes” already, but of course I also have type=peer
and the other stuff for a sip trunk.
The Avaya engineer is telling me that I need to change my “From” header, and I
don’t know how to do that.
I have to figure this out in the next few days, or I’m in deep doo-doo.
-T
Thomas M. Peters | Sr. Systems Administrator | tpeters at
mcts.org<mailto:tpeters at mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org<mailto:helpdesk
at mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>
1942 N 17th Street | Milwaukee, WI 53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter
<https://twitter.com/RideMCTS>
From: asterisk-users <asterisk-users-bounces at lists.digium.com> On
Behalf Of John Kiniston
Sent: Tuesday, February 26, 2019 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users
at lists.digium.com>
Subject: Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM
Thomas,
Does the Asterisk box need to do anything other than handle calls for this one
specific IVR? IE does it ever originate calls?
If it's only recieving calls then I'd turn on guest access and not even
bother with a peer.
Just set
[general]
context=transit-ivr
allowguest=yes
On Tue, Feb 26, 2019 at 3:13 PM Thomas Peters <TPeters at
mcts.org<mailto:TPeters at mcts.org>> wrote:
Hello all, I hope someone can help me with this old Asterisk version. I have to
run this version because of a custom IVR written on it. Porting it would take
much too long and we’d have to hire a consultant because of all the hooks it has
into Oracle databases and real-time information.
We have a brand-new Avaya phone system in place and we will be cutting over to
it in late March 2019.
Presently:
* We have an Asterisk 13.3.2 server with no phones registered to it, acting
as a PSTN gateway. Calls come into it and get distributed to other Asterisk
boxes with phones.
* If a call comes in from the provider marked as having been dialed as
xxx-xxx-6711 (those are digits, not a pattern) it gets routed to the IVR box
* The IVR box runs Asterisk 1.8.7.0 and a custom IVR.
Where we have to get to:
* The new Avaya Session Manager has to have a working SIP trunk to the IVR
so it can pass calls that come into xxx-xxx-6711 to it.
What the problem is:
* I don’t fully understand what’s going on here, neither how it works now,
nor what I need to do to make Avaya’s SM happy.
* When I do sip show peers on my IVR box, I see the Avaya session manager:
jerec*CLI> sip show peers
Name/username Host Dyn
Forcerport ACL Port Status
sessionmgr1 10.10.0.17
5060 OK (1 ms)
* The Avaya engineer says he is seeing “SIP/2.0 400 Bad FROM header” in his
trace screen, and his SM status screen shows “500 NOT REACHABLE” as the status
for our IVR.
* He says we are sending
“asterisk” sip:asterisk@(null):0;tag=as682f2c53
as the “From” in the SIP header.
* He wants us to send
10.10.0.103 at mcts.org<mailto:10.10.0.103 at mcts.org>
or more likely
<sip:10.10.0.103 at mcts.org<mailto:10.10.0.103 at mcts.org>>
instead.
* Pings from either end to the other work just fine.
* nmap doesn’t show port 5060 open. It shows only port 22/tcp open. But then
again, my main asterisk PBX doesn’t show that port open either. So I don’t think
that means anything.
The IVR machine (Asterisk 1.8.7.0) sip.conf file has an old section for the old
PSTN gateway, and a new section I just added for the session manager.
Old section for existing connections to the IVR:
[general]
;context=transit-ivr
context=incoming
disallow=all
allow=ulaw
canreinvite=no
[sipivr]
host=dynamic
secret=1NA6oZjTg1rjhZN8lArDgzLI7z8V2fxV
type=peer
;context=transit-ivr
context=incoming
dtmfmode=inband
The new section, with many failed experiments commented out, is after the
[sipivr] section:
[sessionmgr1]
type=peer
;type=friend
port=5060
host=10.90.0.17
dtmfmode=inband
allowguest=yes
qualify=yes
realm=mcts.org<http://mcts.org>
promiscredir=yes
;Some have suggested using canreinvite=no with Avaya- didn't try that yet
;canreinvite=no
canreinvite=yes
transport=tcp
;context=incoming
context=from-internal
;username=10.90.0.103
fromdomain=mcts.org<http://mcts.org>
disallow=all
allow=ulaw
allow=alaw
tcpenable=yes
tcpbindaddr=0.0.0.0:5060<http://0.0.0.0:5060>
Nothing I tried seems to make it stop sending asterisk@(null) in the header.
This is supposed to be a sip trunk, not an extension, so I think I should NOT be
user a username or secret. I’m not even sure what promiscredir does, or if it’s
helping or harming me.
There’s virtually nothing in the logs about this connection, other than this:
[Feb 26 16:05:42] NOTICE[32142] chan_sip.c: Peer 'sessionmgr1' is now
Reachable. (1ms / 2000ms)
Can anyone help?
Thomas M. Peters | Sr. Systems Administrator | tpeters at
mcts.org<mailto:tpeters at mcts.org>
Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org<mailto:helpdesk
at mcts.org>
Milwaukee County Transit System <http://www.ridemcts.com/>
1942 N 17th Street | Milwaukee, WI 53205
Check us out on Facebook<https://www.facebook.com/mcts> & Twitter
<https://twitter.com/RideMCTS>
--
_____________________________________________________________________
-- 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
--
A human being should be able to change a diaper, plan an invasion, butcher a
hog, conn a ship, design a building, write a sonnet, balance accounts, build a
wall, set a bone, comfort the dying, take orders, give orders, cooperate, act
alone, solve equations, analyze a new problem, pitch manure, program a computer,
cook a tasty meal, fight efficiently, die gallantly. Specialization is for
insects.
---Heinlein
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Thomas Peters
2019-Feb-27 16:09 UTC
[asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM
Thanks for the reply John. About 85-90% of what this box has to do is just handle calls, but it also has options to transfer calls to the main phone system, which up to now has been another asterisk box. For example, you can hit 6 to be transferred to the Lost & Found Department. I do have allowguest set to “yes” already, but of course I also have type=peer and the other stuff for a sip trunk. The Avaya engineer is telling me that I need to change my “From” header, and I don’t know how to do that. I tried various things in sip.conf and now I get> sip show peer sessionmgr1... Callerid : "10.90.0.103 at mcts.org" <> ... My sip.conf file now has: fromdomain=mcts.org realm=mcts.org callerid="10.90.0.103 at mcts.org" I have to figure this out in the next few days, or I’m in deep doo-doo. -T Thomas M. Peters | Sr. Systems Administrator | mailto:tpeters at mcts.org From: asterisk-users <mailto:asterisk-users-bounces at lists.digium.com> On Behalf Of John Kiniston Sent: Tuesday, February 26, 2019 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <mailto:asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM Thomas, Does the Asterisk box need to do anything other than handle calls for this one specific IVR? IE does it ever originate calls? If it's only recieving calls then I'd turn on guest access and not even bother with a peer. Just set [general] context=transit-ivr allowguest=yes On Tue, Feb 26, 2019 at 3:13 PM Thomas Peters <mailto:TPeters at mcts.org> wrote: Hello all, I hope someone can help me with this old Asterisk version. I have to run this version because of a custom IVR written on it. Porting it would take much too long and we’d have to hire a consultant because of all the hooks it has into Oracle databases and real-time information. We have a brand-new Avaya phone system in place and we will be cutting over to it in late March 2019. Presently: • We have an Asterisk 13.3.2 server with no phones registered to it, acting as a PSTN gateway. Calls come into it and get distributed to other Asterisk boxes with phones. • If a call comes in from the provider marked as having been dialed as xxx-xxx-6711 (those are digits, not a pattern) it gets routed to the IVR box • The IVR box runs Asterisk 1.8.7.0 and a custom IVR. Where we have to get to: • The new Avaya Session Manager has to have a working SIP trunk to the IVR so it can pass calls that come into xxx-xxx-6711 to it. What the problem is: • I don’t fully understand what’s going on here, neither how it works now, nor what I need to do to make Avaya’s SM happy. • When I do sip show peers on my IVR box, I see the Avaya session manager: jerec*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status sessionmgr1 10.10.0.17 5060 OK (1 ms) • The Avaya engineer says he is seeing “SIP/2.0 400 Bad FROM header” in his trace screen, and his SM status screen shows “500 NOT REACHABLE” as the status for our IVR. o He says we are sending “asterisk” sip:asterisk@(null):0;tag=as682f2c53 as the “From” in the SIP header. • He wants us to send mailto:10.10.0.103 at mcts.org or more likely <sip:mailto:10.10.0.103 at mcts.org> instead. • Pings from either end to the other work just fine. • nmap doesn’t show port 5060 open. It shows only port 22/tcp open. But then again, my main asterisk PBX doesn’t show that port open either. So I don’t think that means anything. The IVR machine (Asterisk 1.8.7.0) sip.conf file has an old section for the old PSTN gateway, and a new section I just added for the session manager. Old section for existing connections to the IVR: [general] ;context=transit-ivr context=incoming disallow=all allow=ulaw canreinvite=no [sipivr] host=dynamic secret=1NA6oZjTg1rjhZN8lArDgzLI7z8V2fxV type=peer ;context=transit-ivr context=incoming dtmfmode=inband The new section, with many failed experiments commented out, is after the [sipivr] section: [sessionmgr1] type=peer ;type=friend port=5060 host=10.90.0.17 dtmfmode=inband allowguest=yes qualify=yes realm=http://mcts.org promiscredir=yes ;Some have suggested using canreinvite=no with Avaya- didn't try that yet ;canreinvite=no canreinvite=yes transport=tcp ;context=incoming context=from-internal ;username=10.90.0.103 fromdomain=http://mcts.org disallow=all allow=ulaw allow=alaw tcpenable=yes tcpbindaddr=http://0.0.0.0:5060 Nothing I tried seems to make it stop sending asterisk@(null) in the header. This is supposed to be a sip trunk, not an extension, so I think I should NOT be user a username or secret. I’m not even sure what promiscredir does, or if it’s helping or harming me. There’s virtually nothing in the logs about this connection, other than this: [Feb 26 16:05:42] NOTICE[32142] chan_sip.c: Peer 'sessionmgr1' is now Reachable. (1ms / 2000ms) Can anyone help? Thomas M. Peters | Sr. Systems Administrator | mailto:tpeters at mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or mailto:helpdesk at mcts.org http://www.ridemcts.com/ 1942 N 17th Street | Milwaukee, WI 53205 Check us out on https://www.facebook.com/mcts & https://twitter.com/RideMCTS -- _____________________________________________________________________ -- 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 -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein