do i need any ports open inorder to use send mail from behind a router
[EMAIL PROTECTED] wrote:
Send Asterisk-Users mailing list submissions to
[email protected]
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
[EMAIL PROTECTED]
You can reach the person managing the list at
[EMAIL PROTECTED]
When replying, please edit your Subject line so it is more specific
than "Re: Contents of Asterisk-Users digest..."
Today's Topics:
1. Re: Shutting down Asterisk when not in RTP Stream (BJ Weschke)
2. RE: Asterisk Limitations (James Sturges)
3. Re: ACD with polycom ip phones (Matthew)
4. Re: Re: Codecs. (Rich Adamson)
5. Re: DTMFMODE with grandstream (Rich Adamson)
6. Re: ACD with polycom ip phones (Kevin P. Fleming)
7. Re: ACD with polycom ip phones ([EMAIL PROTECTED])
8. Re: ACD with polycom ip phones (Adam Goryachev)
9. Re: ACD with polycom ip phones ([EMAIL PROTECTED])
10. Re: ACD with polycom ip phones (Kevin P. Fleming)
11. Callware VoiceOne released: a new, easy web GUI
([EMAIL PROTECTED])
12. Re: ISDN/CAPI outgoing calls - weirdness with ringing
(Jason Williams)
13. NVFaxDetect ([EMAIL PROTECTED])
14. Can't call out on ZAP channel - need help (Michael Sampson)
15. Re: ACD with polycom ip phones ([EMAIL PROTECTED])
16. Re: Asterisk <-> Skype anywhere/anyhow? (Paul Hewlett)
17. Re: DTMFMODE with grandstream ([EMAIL PROTECTED])
18. Problem using Queue and Sip Soft (Julien SIRBU)
19. RE: Can't call out on ZAP channel - need help (O'Connor, Jonathan)
20. Re: What is the best Dell Machine for Asterisk? (Walt Reed)
21. Re: Can't call out on ZAP channel - need help (Michael Sampson)
22. RE: Re: ztdummy / timer problem with kernel 2.6.14
(Fredrik Emil Jensen)
23. unsubscribe please (Jason Brashear)
----------------------------------------------------------------------
Message: 1
Date: Mon, 19 Dec 2005 07:49:41 -0500
From: BJ Weschke <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Shutting down Asterisk when not in RTP
Stream
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1
On 12/18/05, Douglas Garstang <[EMAIL PROTECTED]> wrote:
Hi Tyler.
We're registering users with OpenSER, which also routes the calls to a series
of Asterisk systems. The really tricky part is allowing different phones
entering through different Asterisk systems to reach other. Currently, the
solution is to, upon registration from phones, issue a forward() command in
OpenSER to forward the registration to every Asterisk system. In this way,
every Asterisk box knows about every phone and it doesn't matter which Asterisk
system takes the call.
It's not a perfect solution though. When OpenSER sends the forward() request to
Asterisk, it also sends back the 'Trying' and 'Ok' messages to the phones
(We're using Polycom's). The phones don't seem to have a problem with these
extraneous messages.... so far. A better solution would have been to use
t_replicate() in OpenSER, which absorbs these messages, but you can only call
t_replicate once.
We may still end up sending all calls BACK through OpenSER again to terminate
the call, as it knows the location of all the phones as well. This is easy from
a simple dial plan perspective, but I'm not sure yet how some of the more
advanced Asterisk features such as hints and ACD Queues will work when
specifying @proxy for their location. I'd prefer to leave OpenSER out of the
equation though. Just trying to get it to do failure_route() etc to Asterisk
is a huge pain considering the docs on it are soooo bad. Oh yeah.... check out
the use of failure_route with t_relay() when sending calls to Asterisk in a
redundant fashion. It seems to be working well so far. Failover is very fast. I
also saw a post on the OpenSER list last night saying that the dispatcher
(which we had looked at before) now supports failure_route too. We liked it
initially because it can load balance on call-id and give you a roughly even
call distribution.
Don't try using realtime either.... it's hard to believe but you can't use it
for sharing a common contact database between Asterisk systems. Digium have
admitted to this.
Asterisk is not a SIP proxy. That's why you see that it still knows
about the calls even though the media has been reinvited away.
Asterisk always knows about the state of its SIP calls given that it's
a B2BUA instead of a SIP proxy.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
------------------------------
Message: 2
Date: Mon, 19 Dec 2005 23:06:19 +1000
From: "James Sturges" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Asterisk Limitations
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
Hey,
That is not what I meant!!!!
I L O V E ASTERISK, every other PBX I have had to deal with, always had
some limitation, I am only using 1.0.7 and really have found nothing
limiting.
We run ISDN 30 line, Reception get 440+ calls per day, dial out 23,000 calls
per month all fully integrated into an old PABX.
My reference was spouse to come across "Go back to old style PBX and you
will be disappointed!"
My public apologies for anyone on the list.
Thanks
James
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter aka
Bret McDanel
Sent: Monday, 12 December 2005 9:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Limitations
On Mon, 2005-12-12 at 09:00 +1000, James Sturges wrote:
Or.....
You can go back to a Traditional PBX and really experience the meaning of
the phrase { significant "limitations". }
That is a really bad excuse for limitations however, and actually does
more harm than good. While it may be true that asterisk has fewer
limitations than another product to say that your option is to use
asterisk or something else more limiting doesnt get any of the problems
fixed. At least the person you replied to gave constructive answers to
remove some of the limitations, such as looking at CVS/bugtracker for
patches or paying money to get someone motivitated to fix them.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
