[Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 111
Lawrence B Thaler
ltgreen1 at yahoo.com
Mon Dec 19 10:44:46 MST 2005
do i need any ports open inorder to use send mail from behind a router
asterisk-users-request at lists.digium.com wrote:
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> 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 (hgaillac-sip at yahoo.fr)
> 8. Re: ACD with polycom ip phones (Adam Goryachev)
> 9. Re: ACD with polycom ip phones (hgaillac-sip at yahoo.fr)
> 10. Re: ACD with polycom ip phones (Kevin P. Fleming)
> 11. Callware VoiceOne released: a new, easy web GUI
> (kleis-asterisk-dev at tiscali.it)
> 12. Re: ISDN/CAPI outgoing calls - weirdness with ringing
> (Jason Williams)
> 13. NVFaxDetect (hgaillac-sip at yahoo.fr)
> 14. Can't call out on ZAP channel - need help (Michael Sampson)
> 15. Re: ACD with polycom ip phones (hgaillac-sip at yahoo.fr)
> 16. Re: Asterisk <-> Skype anywhere/anyhow? (Paul Hewlett)
> 17. Re: DTMFMODE with grandstream (giti at dataproducts.ae)
> 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 <bweschke at gmail.com>
> Subject: Re: [Asterisk-Users] Shutting down Asterisk when not in RTP
> Stream
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <79cf6330512190449q6a5314cx975f5d56bd1966b7 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On 12/18/05, Douglas Garstang <dgarstang at oneeighty.com> 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" <thinking at 1am.com.au>
> Subject: RE: [Asterisk-Users] Asterisk Limitations
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <20051219130744.2FAD47E27 at mail.1am.com.au>
> 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: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] 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.
>
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