Re: [asterisk-users] OT: headsets
- Bill Michaelson [EMAIL PROTECTED] wrote: The IP330 has a subminiature jack for headset/mic combos. Are there quality headsets anyone would recommend for in-office use for heavy users with these phones? Using any wiring path? I've tried a cell phone earphone/mic, and it sounds OK, but it's flimsy for this application. In our call center (250 agents), we use Panasonic KX-TCA60 headsets, which I think we get for 7 or 8 bucks apiece, and sell to the agents at cost. They have covered gooseneck tubes, decent padding on the earpiece, and are fairly sturdy. Turnover being what it is, we don't have to replace too many of them for breakage. They have 2.5mm plugs, and really good audio -- I've plugged mine into my Nextel/RIM BlackBerry 7100i, and called my best friend, who is almost as picky as I am... his opinion is that it not only sounds better than my Plantronics Voyager 510, it sounds better than the mic inside the phone. My opinion is the converse: receive audio is nice too. Recommended. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: headsets
- Bill Michaelson [EMAIL PROTECTED] wrote: Further to this, I'm in the client office today and dealing directly with the users who are reporters and editors for a periodical and conduct many telephone interviews. They want to use their old recording devices with the new phones, but are finding unpleasant audio experiences when they switch them over from the Nortel meridians to the Polycom IP330s. So I'm looking for kit to use here as well. Recommendations most welcome. Are you switching from Nortel kit to Asterisk? Why not set up a user function that starts a recording of the call inside Asterisk itself and save the results to a Samba share where the users can drag them to their desktops? Or not. And in the case of one user, she is adamant she not be required to use a different recording device. I don't know how to approach this except to try a different telephone or mess with Polycom gain settings that the manual advises not to touch. Anybody been down this road - have any wisdom? What is she using now? Some kind of analog recording adapter in the 4p4c handset cord? You may need to leave her for last, get a good solution going and prove it out with others, and then sell it to her boss and let *him* sell it to her. Asterisk will do a *much* better job of recording than anything on the analog side, I would expect. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE switch recommendations?
- Singer Wang [EMAIL PROTECTED] wrote: We've had some bad experiences with Linksys in general (prior to going VOIP) and avoided them. We're running now fully on the NetGear FS728TP switch (24 port 10/100 POE, 4 port 1000 uplink, and 2 slots for fiber modules). While I haven't worked with their PoE, let me say that every piece of NetGear kit I have ever touched is still working, solid as a rock, including the 5 port hub in my bag. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
- Tzafrir Cohen [EMAIL PROTECTED] wrote: There are some awkward methods for sending some text messages oversome channels (SMS in european POTS, SIMPLE and simpler texxt messages in SIP, XMMP for Jingle, and well, probably nothing in IAX. Bristuff ads even a few more bits there). But do we actually care routing those messages from one place to another? This is a major limitation of Asterisk for me. Text messages require much lower a bandwith and a text connection is much easier to setup. Hence it can work even when a voip connection is lousy. The specific thing here (that makes handling text messages within the framework of the more complicated protocol attractive) is *addressibility*. If you already have a path to someone, why should you be forced to *discover* another path to them for some other, simpler protocol? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seemingly easy question: NPA/NXX
- Ira [EMAIL PROTECTED] wrote: At 09:29 AM 9/22/2008, you wrote: ... except in some countries, the phone numbers vary in length in the same city. Say in Hamburg, Germany, your number can be as short as 5 digits or as long as 10. You really have no way of knowing. The unanswered part of that, is this? Can 5 digit number, say, 12345, be the beginning part of a 10 digit number, say, 1234567890? And the answer is: do not confuse E.164 addresses with dialling patterns, grasshopper. Cheers, -- jr '1-888-MITSU2008' a -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seemingly easy question: NPA/NXX
- Karl Fife [EMAIL PROTECTED] wrote: Theory 1 Is it all done with timeouts, but they're CONDITIONAL timeouts. i.e. give a LONG timeout if the number: -did not start with a 1 and is still shorter than 7 digits, -started with a 1 and is still shorter than 11 digits -started with a 011 and is shorter than the theoretical international minimum lenght Theory 2 As you know, a few years ago the 2nd digit of the NPA was always 1 or 0. Therefore the switch could easily determine(without the leading 1) if your first three digits were an NPA or just an NXX (exchange). They were nationally unambiguous. Now that's no longer true. STILL, it could be possible to consider all known valid NPA's and exchanges so they can determine via context what you're trying to do, and thereby optimize the dialing experience? Can anyone speak to this? I would very much appreciate any knowledgable input. Well, my input is knowledgeable, though not authoritative. Yes, each NANP switch actually does have a routing table loaded locally (they call them translations) that tells it where to route calls for each and every valid NPA-NXX in the NANP, and this could be used to authenticate the first 3/6 digits of 7/10/11 digit dialled numbers for intra-NANP calls, and in fact, I would bet that you're correct that that's how they accomplish it. I have never actually seen live switch code on this, but I think I could locate some people who have -- but yes, you'll play hell duplicating it exactly on something with as small a brain as an ATA. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Streaming MoH on 1.4
- Olivier [EMAIL PROTECTED] wrote: A somehow related question, is broadcasting streaming music as music on hold, submitted to any licencing fee ? I got here late. The only way you can legally use music as music on hold is if you either pay, or are not subject to pay, performance royalty money to *someone*. Who you might pay includes BMI, ASCAP, and SESAC, who have standardized annual blanket licenses for that sort of thing, which permit you to play any music to which they've been assigned the right to collect and disburse such monies. Or, if you have recordings directly from an act who have not sold their rights to, say, a music label, they could license you directly. Or you could play the music yourself. But note that if you do *that*, while you aren't liable for performance royalties, you as a performer will own the songwriter(s) money, usually in the form of compulsory mechanical royalties. How those are handled if you record your own arrangement of Hey Jude once and loop it on music on hold, I'm not clear on. No, IANAL. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] about application Jack and its runtime
On Thu, Sep 11, 2008 at 08:11:09PM -0500, Russell Bryant wrote: The Jack application acts as an endpoint for a call. A bit of nomenclature: is Jack the name of an Asterisk application? Or are you referring to JACK, the Jack Audio Connection Kit, whose name is all-caps, directly? And if not, of course, is Jack something that connects JACK to Asterisk? And why should I know all of this already? :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] about application Jack and its runtime
On Fri, Sep 12, 2008 at 09:04:57AM -0500, Russell Bryant wrote: Jay R. Ashworth wrote: A bit of nomenclature: is Jack the name of an Asterisk application? Or are you referring to JACK, the Jack Audio Connection Kit, whose name is all-caps, directly? And if not, of course, is Jack something that connects JACK to Asterisk? Sorry for the confusion. There is a JACK() application, and JACK_HOOK() function, which both connect Asterisk to JACK, the Jack Audio Connection Kit. Ok; that's rather what I thought. And why should I know all of this already? :-) You should be psychic. It's a new 1.6 thing, though, so I don't expect many people to already know all about it. Got it. As posted earlier, more info here ... http://www.russellbryant.net/blog/2008/01/13/jack-interfaces-for-asterisk/ I'll check it out. Any chance you or someone could chime in one more time on my TBCT thread? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
On Fri, Sep 12, 2008 at 10:56:40AM -0500, Matthew Fredrickson wrote: Will I actually need to do PRI debug on that span to tell? Or will seeing hangup messages while I'm still talking be the solution? Seeing hangup messages on the console while the audio path remains indicates success :-) Then, as I suspected, I'm failing. I need to confirm that it's actually provisioned with the carrier, and which switchtype I'm really on. Can *you* confirm, off hand, that 1.2 would do TBCT at *all*? Someone on IRC thinks it wouldn't. -- j -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
On Fri, Sep 12, 2008 at 12:12:56PM -0500, Matthew Fredrickson wrote: Can *you* confirm, off hand, that 1.2 would do TBCT at *all*? Someone on IRC thinks it wouldn't. It will only attempt it for DMS100 switchtype. You must have 1.4 libpri for any other switchtype. Will libpri 1.4 work with asterisk 1.2? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
On Mon, Sep 08, 2008 at 11:28:13AM -0500, Matthew Fredrickson wrote:
For DMS100's version of TBCT, called RLT, one leg *must* be inbound and
the other *must* be outbound. No other combination is going to work.
This is explicitly mentioned in the protocol in RLT.
Ok.
Just found this in my archive.
Matt: should I assume that this implies that if my switch is provisioned
for NI2, and my Asterisk is set to DMS, that things aren't going to work
well at all? :-) (Outbound calls, FWIW, seem to work fine like that...)
Probably not. You can obviously try this out, but don't be surprised if
this doesn't work. You usually want to have your switchtype (which
likewise sets the version of TBCT which is used) set to the same thing
that the other end is provisioned to be.
Ok. I've run a simple test:
exten = 727xxx,1,Dial(${TRUNKY}/727yyy,,r)
exten = 727xxx,2,Hangup
Where TRUNKY is a group that points to the same T-1 on which the calls
are coming in.
And what I get is:
-- Accepting call from '727zzz' to '727xxx' on channel 0/1, span 4
-- Executing Dial(Zap/73-1, Zap/g3/727yyy||r) in new stack
-- Requested transfer capability: 0x10 - 3K1AUDIO
-- Called g3/7276471274
-- Zap/74-1 is proceeding passing it to Zap/73-1
-- Zap/74-1 is ringing
-- Zap/74-1 answered Zap/73-1
-- Attempting native bridge of Zap/73-1 and Zap/74-1
-- Channel 0/1, span 4 got hangup request, cause 16
-- Hungup 'Zap/74-1'
== Spawn extension (default, 727xxx, 1) exited non-zero on 'Zap/73-1'
(I think I got all those numbers sanitized properly.)
And yes, the call went through, and had the CNID of the originating
phone, as I want.
So, since I can't tell from the logs -- no timestamps -- I have to guess
from when the messages show up, but I can't tell if the attempted native
bridge is *succeeding*. How would I know that it had? We do
*successful* ones in other contexts, and I don't recall seeing a
'success' message on those.
Will I actually need to do PRI debug on that span to tell?
Or will seeing hangup messages while I'm still talking be the solution?
And how, again, can I tell what's actually at the other end of the span?
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
___
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Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
On Thu, Sep 11, 2008 at 12:41:12PM -0400, Jay R. Ashworth wrote: Will I actually need to do PRI debug on that span to tell? I did a pri debug to a file, I can see the call go, I see no indication that it actually tried to generate a TBCT/RLT request. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bell Canada (Nortel DMS100) PRI Outbound CNAM issue
On Wed, Sep 10, 2008 at 01:21:42PM -0400, Iain McBride wrote: I'm trying to send CallerID Name information out to the PSTN via a PRI with Bell Canada with no success. With inbound calls (originating from the PSTN) CNAM is received successfully, and we've not had any similar problems with other Telco PRIs, so I'm stumped. From searching the list archives, I believe that a DMS wants to receive the CNAM info in the Q.931 call setup message. I've tried all permutations of switchtype (dms100 national) and facilityenable that I can think of, but I still don't see CNAM coming out the other side. It was only from reading the other folo to your query that I found out that *any carrier* *ever* gets CNAM to send it to an FXS by any means *other* than a database dip, myself. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Network Monitoring
On Tue, Sep 09, 2008 at 09:21:50AM -0500, Darrick Hartman (lists) wrote: I'm using Zabbix to monitor network interfaces, storage, cpu load and a few other things on several asterisk boxes. I'm just looking at adding Asterisk specific monitoring. Simple things like sip registration is pretty easy. Getting the actual status of zap-daddy hardware might be a little trickier. When I get something together I can pass it along. $ head -1q /proc/zaptel/* Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Operator switch billing
On Tue, Sep 09, 2008 at 10:55:14PM +0530, Sriram wrote: I think i wasnt clear here - It'll be either a premium rate line/toll free line but the customer should be charged Rs.6/- per minute only when he hears a prompt(where it'll ask him to press 1 to continue) once he presses 1 to accept the terms . Till the time he hears only the prompt the asterisk box should not send the reversal to the billing switch.. only after pressing 1 should the charging begin...I hope am clear now I assume you're using digital trunking, and such systems do not provide an end to end audio path until supervision. So, without active assistance from the carrier delivering the calls -- who presumably is doing the billing themselves, if I read you correctly -- no, you cannot accomplish this. If you're doing the billing yourself, how to do it should be obvious. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime MySQL clients from same IP problem
On Mon, Sep 08, 2008 at 04:28:52PM +0200, Philipp Kempgen wrote: Apart from that I'd appreciate if you could get a better email client which does not insert so many useless blank lines. :-) SCNR. Likely, for you, like me, it's not that his email client is indersting blank lines... it's that whatever you're using to render his HTML email into text is doing it -- for me, it's lynx under Mutt. He probably just needs to find the send HTML email knob and turn it off. *Break* it off, by preference, but what can you do. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX over T1 Question
On Mon, Sep 08, 2008 at 10:54:12AM -0500, Eric ManxPower Wieling wrote: On a PRI calls come in on ANY B-channel. Therefore you cannot just disable EC on the Fax channels, because there are no dedicated channels for fax. On a Channelized T-1 you can dedicate channels for fax or any other thing. You can't do that on PRI. Strictly speaking, I don't think that's actually true. I believe some carriers can put multiple trunk groups on the same PRI. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
On Fri, Aug 15, 2008 at 03:03:23PM -0500, Matthew Fredrickson wrote: Let me clarify some of this. Under no circumstances can Asterisk receive a TBCT request. We just ignore them. We can initiate them however. There are different TBCT implementations, dependent on which switch type is used, with different restrictions associated with each switch type selected. For true TBCT (on switchtypes of NI2 and 5ESS, AFAIK), you can have any combination of inbound and/or outbound channels (one inbound/one outbound, two inbound, two outbound) and transfer them to the upstream switch. The protocol doesn't care. For DMS100's version of TBCT, called RLT, one leg *must* be inbound and the other *must* be outbound. No other combination is going to work. This is explicitly mentioned in the protocol in RLT. Ok. Just found this in my archive. Matt: should I assume that this implies that if my switch is provisioned for NI2, and my Asterisk is set to DMS, that things aren't going to work well at all? :-) (Outbound calls, FWIW, seem to work fine like that...) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lists.digium.com monthly reminders
On Mon, Sep 01, 2008 at 10:30:18PM +0200, Benny Amorsen wrote: If you normally read this list via gmane and only keep a subscription to be able to post, it's an infinite increase. Yes, I could write a rule to junk those mails, but they aren't really consistent enough between mailing lists. Sure they are: the only MLM that *sends* them is Mailman, and it uses the same body copy always. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM recordings
On Thu, Aug 28, 2008 at 01:56:43PM -0500, Javier Prieto Gomez wrote: I think that you can use quicktime.. Correct; Quicktime as a plugin can play GSM. One small note, though: if you're unlucky enough to be using FireFox on Win2k still, you're screwed. The newest QT that will run on Win2k is 7.1.6, and that's blocked by newer versions of FireFox due to a bug. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura
On Wed, Aug 27, 2008 at 06:38:48PM -0700, Trevor Peirce wrote: I'm pretty confident the Linksys device does not support this functionality. Asterisk can't really do much with it anyway as it can't answer the call waiting call as long as the original call is still engaged. To the OP: is your issue merely that you're trying to get Asterisk to pass the CWCID through to a phone that already knows how to do it? Not that you're trying to get Asterisk itself to deal with it? If so, how does it behave now? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Atlanta Asterisk User's Group Conference Tonight Tuesday, August 26th at 7PM EDT
On Tue, Aug 26, 2008 at 08:36:35PM -0400, SIP wrote: Jay R. Ashworth wrote: On Tue, Aug 26, 2008 at 05:10:35PM -0400, Asterisk wrote: The shared desktop is available using a Java enabled browser at ???http://callin.xelatec.com/vnc??? with a password of ???aretta???. Of course you must first have Zoiper installed and then add a new Zoiper IAX account with Account name ???AtlaugConf???, Server Hostname ???pbx.aretta.net???, Username ???guest???, and no password or other information. Select Show Advanced Options for that account and uncheck ???Register on startup???. Apply the new account and click OK. Then from the main user interface, select the new account, go off hook and dial ???2284???. That should connect you to the conference. Finally, for those really brave souls, you can also connect using the ITAD number of ???2284*455???. ITAD details are available at ???http://www.freenum.org/cookbook/???. Please don't post to mailing lists in non-7bit-ASCII unless you really have no other choice? That's how Mutt rendered your message here. I *think* those ???'s represent smart-quotes, but I really can't tell... and I'm probably not alone. Now, Jay... it's the global telecom world! Not everyone speaks ASCII-only. I retract the tone; the headers clearly say UTF-8; it's my copy of Mutt's fault; apparently. That's a little bit like standing in the United Nations and complaining that not everyone speaks 'murican. ;) That said, you're correct. They're smart-quotes. I'm guessing it was copy-pasted from a Word doc or some such. Yup. It is, however, rarely safe to assume that Any Given Mailing-list will tolerate anything other than non-HTML ASCII-7, without advance notice... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Atlanta Asterisk User's Group Conference Tonight Tuesday, August 26th at 7PM EDT
On Tue, Aug 26, 2008 at 05:10:35PM -0400, Asterisk wrote: The shared desktop is available using a Java enabled browser at ???http://callin.xelatec.com/vnc??? with a password of ???aretta???. Of course you must first have Zoiper installed and then add a new Zoiper IAX account with Account name ???AtlaugConf???, Server Hostname ???pbx.aretta.net???, Username ???guest???, and no password or other information. Select Show Advanced Options for that account and uncheck ???Register on startup???. Apply the new account and click OK. Then from the main user interface, select the new account, go off hook and dial ???2284???. That should connect you to the conference. Finally, for those really brave souls, you can also connect using the ITAD number of ???2284*455???. ITAD details are available at ???http://www.freenum.org/cookbook/???. Please don't post to mailing lists in non-7bit-ASCII unless you really have no other choice? That's how Mutt rendered your message here. I *think* those ???'s represent smart-quotes, but I really can't tell... and I'm probably not alone. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to block incoming calls on PRI
On Thu, Aug 21, 2008 at 08:36:44PM -0500, Dwayne Hubbard wrote: I also want to reiterate that the libpri and Asterisk branches above are development branches, so be careful in a production environment. This functionality will be available in Asterisk 1.6.2. To disable a channel via the CLI type 'pri service disable channel chan' and to enable the channel type 'pri service enable channel chan'. That sounds cool. Two questions: 1) can you do it gracefully (both that and immediate are sometimes useful)? 2) can you take down either an entire span, or a channel range on the command line? (Oh yeah: can I backport it to 1.2? :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FTC Bans Prerecorded Telemarketing Drivel
On Thu, Aug 21, 2008 at 02:15:58AM -0400, Alex Balashov wrote: I would be curious to know where, in this classification, fall various telemarketing schemes that are technically not cold-calls, but are generated from leads that come from customer-provided information, but where the customer does not know explicitly that they are signing up to receive calls. For instance, this is common in a number of industries such as financial services. You do a search to get a quote on something, and provide your phone number in the process, although the phone number bears no relation to the submission and is just an ancillary required item. Several places' telemarketing organisations call you back in response. For example, lendingtree.com. Is this a solicited call? In order to classify that as a solicited call, I believe, you have to have language *on the form the customer fills out* that says they're authorizing you to call, and you have to be able to produce ink-on-paper if the FTC ever calls you on it. IANAL. YMMV. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DSS1 vs SS7
On Thu, Aug 21, 2008 at 02:38:19AM -0400, Alex Balashov wrote: Use DSS1. It's European ISDN and would give you the equivalent of a North American PRI. You don't want SS7. I would assume that means SS7 protocol over a link not routed directly to the SS7 backbone. At least I hope it means that. shudder Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FTC Bans Prerecorded Telemarketing Drivel
On Thu, Aug 21, 2008 at 09:44:50AM -0600, Anthony Francis wrote: Jay R. Ashworth wrote: On Thu, Aug 21, 2008 at 02:15:58AM -0400, Alex Balashov wrote: I would be curious to know where, in this classification, fall various telemarketing schemes that are technically not cold-calls, but are generated from leads that come from customer-provided information, but where the customer does not know explicitly that they are signing up to receive calls. For instance, this is common in a number of industries such as financial services. You do a search to get a quote on something, and provide your phone number in the process, although the phone number bears no relation to the submission and is just an ancillary required item. Several places' telemarketing organisations call you back in response. For example, lendingtree.com. Is this a solicited call? In order to classify that as a solicited call, I believe, you have to have language *on the form the customer fills out* that says they're authorizing you to call, and you have to be able to produce ink-on-paper if the FTC ever calls you on it. IANAL. YMMV. Actually in the US all you have to do is provide some proof of a business relationship with them. Companes get away with calling you if you have ever bought even one item from them. Which doesn't actually speak to the situation about which Alex asked, and I posited. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DSS1 vs SS7
On Thu, Aug 21, 2008 at 12:21:36PM -0400, Alex Balashov wrote: I would assume that means SS7 protocol over a link not routed directly to the SS7 backbone. At least I hope it means that. shudder Indeed, it is certainly private SS7. :-) But that does not mean it is any more desirable for the user. Thank ghod. :-) -- j -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question: Soft phone for ACD agents?
On Thu, Aug 21, 2008 at 09:40:04AM -0700, Michael Collins wrote: To those running call centers I have a question: what kinds of soft phones, if any, do you use? Iâm wondering what is out there that has some hooks for custom applications or host system integration, etc. OTOH, do you prefer a desk phone for any reason? If so, why? The experience in the center I manage the network for was that softphones didn't work out that well, and that regular phones on ATAs weren't much of a win either: ATAs apparently weren't built for 8+ hr/day service; they'd melt down. What we ended up with was Panasonic hardphones with headset jacks (the KX-TS105) plugged into Zhone zPlex 10 FXS channel banks, then T-1 into quad cards on our VICIdial diallers. That gives you 3 spans per room, which works out pretty well even for automatic outbound, though we only do manual these days. I'm in the market for a better station: I don't need the dial, or the handset, or even the ringer (a neon bulb would be fine), but I *do* want something more rugged than those Panasonic's. I'm not sure why no one seems to build a ruggedized agent phone. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FTC Bans Prerecorded Telemarketing Drivel
On Wed, Aug 20, 2008 at 09:38:48AM -0400, Steve Totaro wrote: The worst one is a company that sells extended vehicle warranties. I get them on my cell phone and all of my toll frees. If I wait to get someone on the phone to ask to be removed, they immediately hangup, not saying a word. With any luck that will stop. I told them: if you can continue the manufacturer's warranty on my 21 year old BMW, you bring that right on. I also told them that I'd already asked to be put on their DNC list, both manually and automatically, and had gotten a 4th call. I haven't gotten a fifth. Their CNID was bogus, too. Gives us legitimate telemarketers a bad damn name. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2BCT from the Asterisk side
Ok, now that, thanks to Kevin, I understand what I'm talking about with
TBCT (:-), I think a couple people alluded to the current state of
affairs for the other situation that I was *not* looking for
information about, originally, which was:
What is the current state of affairs WRT an Asterisk instance
*requesting* TBCT from an FXS wrt 2 timeslots on its PRIs?
By version, if you know; I'm still on 1.2.2{47}
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
___
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[asterisk-users] 2BCT from the Asterisk side
[ Unthreaded, cause I know better than to thread-jack; sorry ]
Ok, now that, thanks to Kevin, I understand what I'm talking about with
TBCT (:-), I think a couple people alluded to the current state of
affairs for the other situation that I was *not* looking for
information about, originally, which was:
What is the current state of affairs WRT an Asterisk instance
*requesting* TBCT from an FXS wrt 2 timeslots on its PRIs?
By version, if you know; I'm still on 1.2.2{47}
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
___
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Register Now: http://www.astricon.net
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Re: [asterisk-users] FTC Bans Prerecorded Telemarketing Drivel
On Wed, Aug 20, 2008 at 03:32:02PM -0700, Michael Collins wrote: Gives us legitimate telemarketers a bad damn name. :-) Isn't legitimate telemarketers an oxymoron? I don't think so, no. We do B-to-B only, and the FTC apparently thinks that's sufficiently less intrusive that the TSR's exempt us. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
On Tue, Aug 19, 2008 at 07:35:03AM -0500, Kevin P. Fleming wrote: Jay R. Ashworth wrote: I'll assume you've watched it on a PRI, so I'll defer, but I wouldn't expect that myself; I would expect that when you tell the switch to transfer it, you go immediately from one B channel to 0. You should expect that; in fact, that's what the 'TB' in 'TBCT' stands for... for a time, there are two B-channels involved. TBCT is a method of taking two existing already connected B-channels and linking them together into the network, it is not a 'transfer' facility where you provide a target DN and an existing call is 'transferred' to that destination. That feature is ELT (Explicit Line Transfer) and may also be known by other names, or possibly Call Deflection (CD) depending on whether you do it before the call is answered or after. In the scenario you outlined, the original caller (party A) calls this mediator (who answers as party B1). They then place a call (party B2) to you (party C), which you answer. Once that call is established, they can TBCT party A and party C, thus dropping the party B1/B2 legs. You will never see party A's identifying information on the call to you unless party B decides to provide it to you in some fashion; the network signaling would never know to provide it to you, since this is not a call transfer in the RDNIS sense of 'call transfer'. *Aha*. Got it. So whether I can, as a recipient, get the CNID of the call originated by the second party depends on whether they can send it to me. That is, *that* part is a capability of the second party, and the only capability the second party's IXC has to provide is bare TBCT. I understand better now, Kevin; thank you very much. I'm in that conference call in about 20 minutes. ;-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
On Sat, Aug 16, 2008 at 09:35:10PM -0400, Ron Joffe wrote: On Saturday 16 August 2008 14:37, Jay R. Ashworth wrote: TBCT is a feature of LEC/IXC edge switches; there isn't much use for it in any other context. I don't care if you're using Asterisk to be an edge switch, but it's a *carrier* feature, by and large. Certainly in the specific instance I'm discussing, it is. Why would TBCT not be applicable in a scenario where * is being utilized as a slave to a main PBX. * might receive a call from the PBX, and then want to transfer it to another extension on the PBX itself. Hmmm. Perhaps. But if the Asterisk is upstream of the PBX, then the *PBX* would need to know how to deal with it. I see your point... but it's still orthogonal to what I need to know. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ANSI terminal colors
On Fri, Aug 15, 2008 at 11:58:48PM -0500, Tilghman Lesher wrote: On Friday 15 August 2008 22:31:18 Jay R. Ashworth wrote: On Fri, Aug 15, 2008 at 07:08:33PM +0200, Philipp Kempgen wrote: True. But lines of varying length with a black background in a white terminal window don't make it any better. Just causes the ragged margins to stand out. Run it into a file with screen(1l) and use less -r to watch it. That's what I do, anyway... Or one could simply go try the patch which has been posted for the past 24 hours. You'll forgive me if I'm not inclined to patch my 14-node production 1.2.24 cluster, right? :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
On Fri, Aug 15, 2008 at 11:56:30PM -0500, Tilghman Lesher wrote: There's no hairpin involved: the point of TBCT is that you tie up *0* timeslots instead of 2, to forward a call. There is a hairpin involved. The call (for several milliseconds at least) is using two channels on the PRI before the 2BCT succeeds, and then the call no longer takes up any channels. It is only when the PRI detects the hairpin, through the native bridge code that it is able to detect that the call is eligible for 2BCT. I'll assume you've watched it on a PRI, so I'll defer, but I wouldn't expect that myself; I would expect that when you tell the switch to transfer it, you go immediately from one B channel to 0. Why would an Asterisk instance call itself on the same span? Very simple. Call your main number, and if you don't have special logic in your internal dialplan context to handle that, the call will go out to the telco and dutifully come right back in on the same circuit. Sure, but that's not a target for TBCT anyway. Similarly, Asterisk cannot complete a 2BCT request, if Asterisk is on the NET side of the PRI circuit. That might could be added in the future, but it is not supported now. So in summary, Asterisk can request 2BCT, but it cannot perform a 2BCT if requested from the other side. Nothing can perform a TBCT unless it's a PRI server, not a client; it's function of 5ESS's and DMSen; you have to be an SS7 speaker to do it in the first case. I don't think that's the case. Matt would know more, and his reply suggests that it certainly would be possible for Asterisk to do this. SS7 is not at all required here. Allow me to phrase it differently: TBCT is a feature of LEC/IXC edge switches; there isn't much use for it in any other context. I don't care if you're using Asterisk to be an edge switch, but it's a *carrier* feature, by and large. Certainly in the specific instance I'm discussing, it is. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ANSI terminal colors
On Thu, Aug 14, 2008 at 03:23:57PM -0500, Tilghman Lesher wrote: On Thursday 14 August 2008 13:59:37 Philipp Kempgen wrote: Jared Smith schrieb: On Thu, 2008-08-14 at 20:35 +0200, Philipp Kempgen wrote: Whenever something spits out lines with a different background color (of a varying runlength!) my eyes start to hurt. You can turn of the ANSI color support completely by adding nocolor=yes to the [options] section of asterisk.conf and then restarting Asterisk. Sure. But I want colored output on my default background color. :-) ls with dircolors works perfectly. So I was curious if there is a reason for Asterisk to behave differently (forcing the background color to black). Because nobody else ever asked, perhaps? No, I would suspect that it's because there's probably a subconscious perception that colorcoded text works better on black than white, and indeed, it does: you often have to fiddle the colors chosen to get it to be readable against black: the default red has too little luminance to stand out from black, for instance. It's the same problem as backlit keyboards with silver coatings, or the display on a Palm IIIe: in certain lighting conditions, you simply can't read it at all. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstDB/Berkely DB - Hash function? Balanced-Tree? b-Tree? Linked List?
On Fri, Aug 15, 2008 at 12:56:49AM -0500, Karl Fife wrote: The key-space is ideal. It's just npa/nxx lookups so it's UNIQUE and EVENLY DISTRIBUTED Based on my knowledge of the NPA/NXX space, I wouldn't expect that either a) A given batch of random DNs would have either or both NPA/NXX components evenly distributed over all the valid NPA/NXXs in the NANPA, or b) that the assigned NPA/NXXs in the NANPA are themselves evenly distributed over all the valid NPA/NXXs. Could you clarify the background that brings you to that assumption? Do you have empirical data? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] noise cancelling headset vs handset
On Fri, Aug 15, 2008 at 04:59:26AM -0700, Vikas wrote: Observation: When the agents talk using the polycom 650 handset the voice quality on the other end is much better compared to if they talk using the plantronics noise cancelling headset. If you would like a recorded phone call when a agent is taking using the headset vs when a agent is talking using the polycom 650 handset let me know and I can post a URL to the file under this thread. Is this happening because they are using a noise cancelling headset. It would seem reasonable to expect that the noise cancelling headset is cancelling out some of the tones in the voice. Correct: a noise cancelling headset is going to have substantially altered audio reproduction characteristics vs. a non-noise cancelling handset, partially because it's a headset, partially because it's noise cancelling, and partially because the Polycom desksets have substantially the best audio I have ever come across. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI TBCT - Practical Experience, Anybody?
I may have to do some work with TBCT, and probably cross-carrier TBCT, here shortly, and I haven't ever worked with it. If anyone on the list ever has, I'd be interested to know: 1) Only the carrier first involved with the call has to actually be provisioned for it, correct? 2) Both incoming and outgoing calls can be TBCT'd? 3) If a placed call is transferred to me via TBCT, can I get the DN of the original target call sent to me as CNID? 4) Does it, in fact, matter if the call placer, and the TBCT target, are on the same IXC? I want someone to place calls for me, talk to the people for a while, and then do an unsupervised transfer to me wherein I can capture the other party's number off the call itself and feed the calls into my VICIdial/Asterisk instance. All my incoming lines are Zap/PRI. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
On Fri, Aug 15, 2008 at 02:37:46PM -0400, Matt Florell wrote: Most carrier sales people don't know what TBCT is unfortunately, and even if a carrier is capable of doing it, it is a possiblity that not all of their equipment is capable of doing it. One client of mine tried to get TBCT working across all 16 of their PRIs(all on the same carrier) and it only worked on 4 of them, supposedly because not all of the telco equipment was capable of the feature. I expect to fight this battle, yes. :-) This actually depends on the kind of PRI service you have. For instance with DMS100 circuits you can only do TBCT with calls that come in to your circuit, not with outgoing calls. As for connecting two incoming calls, since that is not possible in Asterisk(to natively bridge two incoming calls together) I can't see how you would get that to work even if it is possible in TBCT. To be more clear, what I'm after is to have *someone else besides me* place calls out their PRI, and then TBCT those placed calls to my DN. By the time the calls get to me, they should just be standard phone calls. So I expect the call-placing-party to need TBCT, but not me. I believe that only DMS100, NI2 and 5ESS PRI signalling protocals are capable of TBCT with the current zaptel code-base. Also, the two B channels involved in the TBCT have to use the same D channel. And I'm probably not concerned with whether Asterisk can deal with TBCT, because Asterisk probably won't be involved at that stage; just once the call's transferred to me. But before I inquire of said second party whether they *can* do that, I wanted to confirm it was possible. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
On Fri, Aug 15, 2008 at 02:49:17PM -0500, Tilghman Lesher wrote: To be more clear, what I'm after is to have *someone else besides me* place calls out their PRI, and then TBCT those placed calls to my DN. By the time the calls get to me, they should just be standard phone calls. So I expect the call-placing-party to need TBCT, but not me. I believe that only DMS100, NI2 and 5ESS PRI signalling protocals are capable of TBCT with the current zaptel code-base. Also, the two B channels involved in the TBCT have to use the same D channel. And I'm probably not concerned with whether Asterisk can deal with TBCT, because Asterisk probably won't be involved at that stage; just once the call's transferred to me. But before I inquire of said second party whether they *can* do that, I wanted to confirm it was possible. 2BCT works when the telco originates the call and Asterisk is hairpinning the call back out the same PRI circuit. However, Asterisk does not support the opposite direction. That is, a call originated from Asterisk that comes back in via the same PRI circuit cannot be 2BCT. I'm not certain whether this is a limitation of Asterisk alone or of the protocol, but it cannot be done. I'm not sure we're not talking at cross purposes, here, Tilghman... but TBCT is an instruction to an end-office that sent you a call to yank it back off your timeslot and forward it along to someone else. There's no hairpin involved: the point of TBCT is that you tie up *0* timeslots instead of 2, to forward a call. Why would an Asterisk instance call itself on the same span? Similarly, Asterisk cannot complete a 2BCT request, if Asterisk is on the NET side of the PRI circuit. That might could be added in the future, but it is not supported now. So in summary, Asterisk can request 2BCT, but it cannot perform a 2BCT if requested from the other side. Nothing can perform a TBCT unless it's a PRI server, not a client; it's function of 5ESS's and DMSen; you have to be an SS7 speaker to do it in the first case. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
On Fri, Aug 15, 2008 at 03:03:23PM -0500, Matthew Fredrickson wrote: Under no circumstances can Asterisk receive a TBCT request. We just ignore them. We can initiate them however. There are different TBCT implementations, dependent on which switch type is used, with different restrictions associated with each switch type selected. For true TBCT (on switchtypes of NI2 and 5ESS, AFAIK), you can have any combination of inbound and/or outbound channels (one inbound/one outbound, two inbound, two outbound) and transfer them to the upstream switch. The protocol doesn't care. For DMS100's version of TBCT, called RLT, one leg *must* be inbound and the other *must* be outbound. No other combination is going to work. This is explicitly mentioned in the protocol in RLT. Oddly, I learned about TBCT *from the feature planning guide concerning the DMS100*, to which I had a subscription 10 years or so ago; I don't recall it having a different name or limitations. I can lay my hands on that issue; I will. But, again, I wasn't concerned with whether Asterisk could do anything specific with TBCT, except catch calls sent to me by someone else performing one. Which my original message was pretty clear on. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ANSI terminal colors
On Fri, Aug 15, 2008 at 07:08:33PM +0200, Philipp Kempgen wrote: True. But lines of varying length with a black background in a white terminal window don't make it any better. Just causes the ragged margins to stand out. Run it into a file with screen(1l) and use less -r to watch it. That's what I do, anyway... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] seeking hardware recommendation PCI versus PCI Express E1 card (te407p vs te420bf)
On Wed, Aug 13, 2008 at 03:06:58PM -0500, Kevin P. Fleming wrote: Keep in mind that even if you use 4 E1 circuits with the card, the total bandwidth consumption of card is approximately 1 megabyte per second (4 times 2 megabits per second), which is drastically below the PCI bus bandwidth of 132 megabytes per second (33 MHz bus with 32-bit transfers). No quad-T1/E1 card will ever be able to saturate a PCI bus, especially not PCI-X or PCI-E. Though, depending on the design of the card and drivers, interrupts/sec may be an issue -- sometimes a deciding issue. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New GUI for Realtime Asterisk - RAGUI
On Wed, Aug 13, 2008 at 07:32:09PM -0400, Mike Clark wrote: It manages the MySQL database tables directly via RoR. It is specifically built for using Realtime Asterisk. It uses #exec statements in the extensions.conf file to set up the context declarations. I do hope that when you designed it, in lieu of a formal API for modifying those tables, you created one of your own, and used it for all your higher level code -- so that when they rev the tables in a version upgrade, you only have to track it in one place... hers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?
On Wed, Aug 13, 2008 at 11:54:23PM -0400, Steve Totaro wrote: NIC card is redundant ;-) And you can take that to the ATM machine. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922
On Wed, Aug 13, 2008 at 11:01:46PM -0500, Darrick Hartman wrote: You can get an adapter for the Plantronics that will plug into the 2.5mm jack on the phone. I need the opposite adapter: to plug a 2.5 headset into an RJ-9 Polycom. Anyone know where I can find that? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?
On Thu, Aug 14, 2008 at 11:53:57AM -0400, Steve Totaro wrote: On Thu, Aug 14, 2008 at 11:50 AM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: What would on-board NIC be? Er, it would be one integrated with the MoBo, on the board if you will... I believe the thrust of Eric's riff is: how can it be a card if it's on the mobo. Cheers, -- jr 'the Joke Explainer' a -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Set Asynchronous Balanced Mode Extended
On Thu, Aug 14, 2008 at 02:18:33PM -0400, Steve Totaro wrote: Oh, I didn't realize that your job was riding on this. In that case, you can PayPal me a weeks pay as my consulting fee. ;-) Ironically, Asking Questions the Smart Way recommends that you not go into how urgent your need for answers to your question is... though probably not for this exact reason. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922
On Thu, Aug 14, 2008 at 11:10:10PM -0500, Darrick Hartman wrote: Jay R. Ashworth wrote: On Wed, Aug 13, 2008 at 11:01:46PM -0500, Darrick Hartman wrote: You can get an adapter for the Plantronics that will plug into the 2.5mm jack on the phone. I need the opposite adapter: to plug a 2.5 headset into an RJ-9 Polycom. Anyone know where I can find that? If you use the Plantronics headsets, many of them have a quick disconnect plug part way down the cord. That was what I was referring to, not an adapter which converts RJ-9 to 2.5mm. I'm clear on that, yes. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR accuracy
On Wed, Aug 13, 2008 at 02:37:58PM +0200, Klaus Darilion wrote: Thanks for the detailed information. What about the following scenario: ANSWER and HANGUP happens in the same second. Thus, the call duration will be 0 seconds. How are such use cases usually handled in the billing system? Are you billing the user (e.g. 1 second or the minimum fee) if the call is ANSWERED even if Asterisk reports 0 seconds? Seems to me that's a policy issue; ie: why are you asking *us*? :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HP server and Meetme applications
On Mon, Aug 11, 2008 at 02:45:18PM -0400, aymen warfalli wrote: Hi list Hi. Please don't thread-jack. I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2.83 with 4 gig RAM I install Centos 5.2 64 bit and it is rumming pretty well and I need to use it as voice conferencing application (Meetme) server for high number of users fit to 8 E1 links (240 users ) with echo cancellation using same coding use g711 I would be surprised. I have a Core2Quad 2.4G machine with one Sangoma quad-T card (admittedly, with*out* HWEC), that's meetme-ing 72 Zap channels from Channelbanks into an equivalent number of IAX channels over 100BaseT; each conference has at least one extra Record() app running full time, as well as one playback. This is a production machine, but it takes careful tuning to keep it production-stable at that load (where my definition of production-stable is if the load average breaks 2.0 for more than 5 consecutive half-minutes, I get nervous.). You *might* get that to work, with HWEC, but I have no experience with whether MeetMe is going to behave if you want it all to be One Big Conference -- it will *certainly* depend on which Asterisk release you're running, and my intuition says if you have to ask, you're not the guy to set it up; pay someone. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI and Call Center to do CRM integration
On Thu, Aug 07, 2008 at 04:12:26PM -0700, bilal ghayyad wrote: CRM: Customer Record Module which is any kind of application. Well, no; CRM means Customer Relationship Management... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vars in Macros called by DIAL with option M
On Tue, Aug 05, 2008 at 02:13:15PM -0500, Tilghman Lesher wrote: above the original post is very confusing. Please stop doing this. The format of this post is in reverse, to demonstrate why posting a reply option is only in trunk. So no, it would not help him out. Yes, it works the same way, by using the U() option to Dial. However, this Question 2: prior to the origination of the slave channel. channel. No inheritance is possible, because the master channel originated channel, so any values set in the slave channel will not affect the master Apple. The variable is only set in the slave channel, not in the master Question 1: And the award for Best Illustration of a Point goes to... Tilghman Lesher! Mr Lesher has been nominated for this award 4 times; this is his first win. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2000+ user Asterisk PBX
On Sun, Aug 03, 2008 at 07:11:41PM +0100, Femi wrote: Thanks for the design tips Based on all the information I have been able to gather this is the config that I believe will work best: 1. SER on 2 servers in HA (failover) config 2. Asterisk cluster of 4 (or more) servers 3. MySQL and SMTP on 2 servers with HA config 4. Cisco or FoneBridge E1 gateways for telco access 5. Other servers for ITSP access 6. All systems separated on two racks at different ends of the building 7. HA switches, routers and firewalls Now, there's one small problem. You are likely to play pluperfect hell getting carriers to do physically diverse entry into your building, especially if it's not really large. And if you *don't* get it, then the rest of your investment in HA is pretty much for naught. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push to talk over cellular with asterisk (was: Autoanswer in Nokia SIP clients?)
On Mon, Aug 04, 2008 at 03:45:32PM +0100, Gordon Henderson wrote: So I'll throw the cat amongst the pigeons by saying: Who cares about PTT? Me. Seriously - Push to talk - Half duplex Communications. How ancient is that! It's really reminiscent of ancient US style truckers - Smokey and the Bandit and all that. That's just so last century. Lets put all that behind us and get with the 21st century! We all have hands-free, full duplex communications now, so lets just forget all that old rubbish and get on with the programe. Clearly, you don't work in an environment where PTT is a feature, not a bug. Lucky you. :-) I think a provider tried it over here for a while - Orange IIRC, but they charged an arm and a leg for it, so no-one used it. Everyone I talked to about it just laughed. If I want to talk to someone with my mobile while working, I use a bluetooth headset. I touch my borg implant, say HOME into it and it dials home... I'm not going to hold an E90 to my ear, push a button, say Hello? Can you hear me? Kchchchct let the button go then wait for a reply. Amazingly, Nextel has made a pretty good living out of it over here for almost 20 years, notwithstanding the number of subs they've lost because they didn't budget the network expansion. PTT communications have quite a number of advantages, actually, some of which I itemized here: http://bestpractices.wikia.com/wiki/Nextel though not all of them. Clearly, they're popular enough that Sprint went through the 7 layers of hell necessary to port QChat to EV-DO and make it interoperate with the old iDen Direct Connect system, in observance of Metcalfe's Law. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push to talk over cellular with asterisk
On Mon, Aug 04, 2008 at 08:04:31PM +0300, Stefan Gofferje wrote: If I really want PTT, then I'll go out buy a pair of Motorola handsets. Which would reach how far? 500m? Surely not from Helsinki to Oulu or even internationally... And, as a folo, Nextel is slowly transitioning their Direct Connect service from the custom iDen backbone originally built for it... to Qualcomm's QChat, which is, effectively, a voice-capable Instant Messaging and Presence client that runs over their EVDO backbone. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how many quad T1 cards
On Fri, Aug 01, 2008 at 03:24:51PM -0400, C F wrote: 2 or 3 cheaper gateway machines that have just the T1 cards in it, will do way better than one monstrous machine. Indeed. We run VICIdial here for about 255 agents, and we're doing that on roughly 11 dialler boxes with 2-3 spans and 15-22 fronters per box, and one closer box with 77 seats on it; 4 channelbanks but no T-spans. Our DBMS is a separate Quad-Opteron with 16GB and we have a smattering of other boxes in the datacenter; our current server count is 41. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
On Thu, Jul 31, 2008 at 05:36:14PM +1000, Lee, John (Sydney) wrote: Yes, I tried all sorts of cables and ended up getting the local contact to complain to NETCOM. An engineer came and swapped the Fast Ethernet to E1 converter. Hmmm. Whose side is Fast Ethernet, and whose side is E1? Are you trying to take the E1 that they've *converted into 100BT* for you and plug it into an E1 port? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] comparing pots solutions for asterisk
On Thu, Jul 31, 2008 at 01:20:00PM -0700, Eric Fort wrote: I've been looking at various solutions for getting FXS and FXO lines in and out of asterisk. one solution is using TDM-400 cards. Another solution is using the grandstream GXW400x and GXW410x gateways. Cost per port seems lower on the gateways and no pci slot is required. Why would one choose to use the TDM-400 cards? what would be the advantages and disadvantages of each approach? How many lines do you need to move? We tend to use, for hooking up our analog agent phones, the discontinued Zhone zPlex-10 T-1 channel bank... though I suppose that's only cost effective, even at the $250 used we pay for them, if you already need to have a T-1 card in the machine. Going from a single to a dual or a dual to a quad T card is cost effective. Putting in a T-card from scratch, maybe not so much. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
On Tue, Jul 29, 2008 at 01:56:23PM +0300, Tzafrir Cohen wrote: An output of 'head -n 1 /proc/zaptel/*' after the fact might help us help you. Did you re-run ztcfg after editing zaptel.conf ? On a sidebar, let me suggest head -1q; it's neater. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 06:33:35PM -0400, Eric ManxPower Wieling wrote: The something is generated by Asterisk at the time the call is created. You should never add it, since you don't control that call instance info. In fact, you should almost never care about the call instance string. The -1 means first instance of a call on this channel, a -2 would be seen in you answer a 2nd call for call waiting. Ah. Got it. Thanks. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
the argument to Dial, I get CHANUNAVAIL.
Zap/01-1 ??? How come?
Zap/01 is valid and equivalent to Zap/1 .
And yet, feeding it to Dial didn't work, and stripping the 0 off did.
I'm on 1.2 if that makes a diff.
So I guess I need finally to end up with
exten = _88XX1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _88XX1NXXNXX,2,Dial(Zap/${EXTEN:2:2}-1/${EXTEN:4},30,o)
Err.. that's not mine. It seems like a dial-by-span syntax.
Just remove the '-1' .
Well, it worked, but ok, I'll take it off.
Now to figure out how to do it across IAX channels from one server to
another.
Which I have, but I haven't tested it yet.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
___
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Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 01:09:16AM +0300, Tzafrir Cohen wrote: On Fri, Jul 25, 2008 at 05:52:20PM -0400, Al Baker wrote: Quote seems like a dial-by-span syntax. What is Dial-by-span ? Zap/span-num-channel-in-span Hmmm. Zap/2 here means the second Zap timeslot on the machine, as does Zap/2-1, using all PRI's on Digium and Sangoma cards. I would have *expected* that it might behave the way you suggest, but it appears not to. Unless it has something to do with the way my zaptel presents the spans to Asterisk... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 09:34:29PM +0300, Tzafrir Cohen wrote:
On Sat, Jul 26, 2008 at 01:12:42PM -0400, Jay R. Ashworth wrote:
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
the argument to Dial, I get CHANUNAVAIL.
Zap/01-1 ??? How come?
Zap/01 is valid and equivalent to Zap/1 .
And yet, feeding it to Dial didn't work, and stripping the 0 off did.
I'm on 1.2 if that makes a diff.
I've used this extensively since 1.0, FWIW.
Looking at the code: the paarsing is done by sscanf. Maybe it does not
consider a number with a leading 0 as a number?
What error/warning do you get when trying to use Zap/01 ?
Chanunavail/Congestion.
Here, let me go get the exact message...
==88
-- Executing AGI(SIP/101cathy-b7619990, call_log.agi|880116142154432)
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
+ CALL LOG START : |1216995262.36|SIP/101cathy-b7619990|880116142154432|SIP|
7274514974|2008-07-25 10:14:22
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial(SIP/101cathy-b7619990, Zap/01-1/16142154432|30|o) in
new stack
Jul 25 10:14:22 NOTICE[25497]: app_dial.c:1076 dial_exec_full: Unable to create
channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing NoOp(SIP/101cathy-b7619990, CHANUNAVAIL) in new stack
-- Executing NoOp(SIP/101cathy-b7619990, Hangup Cause: 0) in new stack
-- Executing Hangup(SIP/101cathy-b7619990, ) in new stack
== Spawn extension (default, 880116142154432, 5) exited non-zero on 'SIP/101
cathy-b7619990'
==88
Copied and pasted. I later extended the rules, as you saw, to have a
special rule for 880X, and it worked just fine.
Not sure what to tell you, but it seems to be that.
Note that I have not *yet* taken the -1 off the end, so it cannot be
that.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
___
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Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 09:32:03PM +0300, Tzafrir Cohen wrote: Zap/2 here means the second Zap timeslot on the machine, as does Zap/2-1, using all PRI's on Digium and Sangoma cards. I would have *expected* that it might behave the way you suggest, but it appears not to. Unless it has something to do with the way my zaptel presents the spans to Asterisk... Right. This is not supported. And you get there a warning: zt_request: Unknown option '-' As the '-' is parsed as a channel option (like 'r' or 'c'). Time to fix voip-info. Except that that is what Asterisk is giving *us*: -- Local/[EMAIL PROTECTED],1 answered Zap/73-1 -- IAX2/VICIast26-19 answered Zap/73-1 -- Zap/11-1 is ringing -- Zap/11-1 answered SIP/101cathy-0824cda0 As nearly as I can discern, those are messages where the Zap channel ide is being generated by Asterisk, based on no particular configuration we gave it (there are lots of others, but they could just be repeating an argument they were passed; mostly Application messages). We do in fact, see that zt_request message, but it's not like we made *up* the whole 73-1 thing... :-) Cheers, - jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 10:38:07PM +0300, Tzafrir Cohen wrote: [ quoting me ] Chanunavail/Congestion. Here, let me go get the exact message... ==88 -- Executing AGI(SIP/101cathy-b7619990, call_log.agi|880116142154432) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi + CALL LOG START : |1216995262.36|SIP/101cathy-b7619990|880116142154432|SIP| 7274514974|2008-07-25 10:14:22 -- AGI Script call_log.agi completed, returning 0 -- Executing Dial(SIP/101cathy-b7619990, Zap/01-1/16142154432|30|o) in new stack Why do you keep adding that -1? Because, as I noted in my other message, *ASTERISK KEEPS ADDING IT*. :-) Try Zap/01 Though I tried originating a call to Zap/04 and Zap/04-1 and both worked well here (1.4). With the -1 I got the warning I mentioned above about the unknown option. Sure. But did *the call go out*? Jul 25 10:14:22 NOTICE[25497]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing NoOp(SIP/101cathy-b7619990, CHANUNAVAIL) in new stack -- Executing NoOp(SIP/101cathy-b7619990, Hangup Cause: 0) in new stack -- Executing Hangup(SIP/101cathy-b7619990, ) in new stack == Spawn extension (default, 880116142154432, 5) exited non-zero on 'SIP/101 cathy-b7619990' ==88 Copied and pasted. I later extended the rules, as you saw, to have a special rule for 880X, and it worked just fine. Not sure what to tell you, but it seems to be that. Note that I have not *yet* taken the -1 off the end, so it cannot be that. See? I *knew* I mentioned it. Note that Mike Cargile at VICIdial looked over that dialplan, and he didn't seem to have a problem with the -1; I'm pretty sure it's in the VICIdial standard dialplans. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 10:48:54PM +0300, Tzafrir Cohen wrote: Except that that is what Asterisk is giving *us*: -- Local/[EMAIL PROTECTED],1 answered Zap/73-1 -- IAX2/VICIast26-19 answered Zap/73-1 -- Zap/11-1 is ringing -- Zap/11-1 answered SIP/101cathy-0824cda0 Trying to dial to Zap/11-1 instead of to Zap/11 is like trying to SIP/101cathy-0824cda0 installed of dialing to SIP/101cathy So, clearly, I'm not smart enough; precisely what are the semantics of the 'Something' in Technology/Channel-Something? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing an Asterisk Server behind a MeridianNorstar
On Wed, Jul 23, 2008 at 07:12:40PM -0600, Joseph L. Casale wrote: Not odd at all as far as I'm concerned - I know a number of places that segregate LAN traffic from VoIP traffic using multiple VLANs over the one physical link. VLANs would be the best solution (short of running multiples cables for PC and phone) to achieve this. I would have about 30 phones I think over 6-12 lines. Vlans would be a must as I would surely be using the same network infrastructure. I will keep hunting... A small switch in each office might not be a big deal. The question you haven't answered yet, Joseph, is how does your Meridian connect to the PSTN? Is it a T-1 now, or analog? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
the argument to Dial, I get CHANUNAVAIL.
So I guess I need finally to end up with
exten = _88XX1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _88XX1NXXNXX,2,Dial(Zap/${EXTEN:2:2}-1/${EXTEN:4},30,o)
exten = _88XX1NXXNXX,3,NoOP(${DIALSTATUS})
exten = _88XX1NXXNXX,4,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _88XX1NXXNXX,5,Hangup
exten = _880X1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _880X1NXXNXX,2,Dial(Zap/${EXTEN:3:1}-1/${EXTEN:4},30,o)
exten = _880X1NXXNXX,3,NoOP(${DIALSTATUS})
exten = _880X1NXXNXX,4,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _880X1NXXNXX,5,Hangup
Which I just retested and it works.
Now to figure out how to do it across IAX channels from one server to
another.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
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Those who count the vote decide everything.
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Re: [asterisk-users] T1/PRI dialing
On Thu, Jul 24, 2008 at 09:39:49AM -0400, Jerry Geis wrote: When dialing using a T1/PRI with a outgoing call files Like Channel: Zap/1/95551212 is there ever a need to delay or pause in there? I have gotten feedback from a customer that instead of dialing the 95551212 it seems to have dialed 55512 which just happened to be an internal extension. Well, if you send 55512 out a PRI timeslot, would you expect that call to suddenly show up back inside the building? Unless your PRIs go to a Centrex, I wouldn't think so. :-) This sounds like a problem on the station set. What type is it? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Thu, Jul 24, 2008 at 09:39:42AM -0400, Doug Lytle wrote:
Jay R. Ashworth wrote:
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls out over each channel for testing?
I use:
exten = _71NXXNXX,1,Read(ZAPLINE|conf-getchannel)
exten = _71NXXNXX,n,Set(CALLERID(number)=734XXX)
exten = _71NXXNXX,n,Dial(ZAP/${ZAPLINE}/${EXTEN:1})
exten = _71NXXNXX,n,NoOP(${DIALSTATUS})
exten = _71NXXNXX,n,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _71NXXNXX,n,Hangup()
Nice. I assume the Noop's capture the text in the log, then? (See?
Told you I was fresh caught :-)
Hold it: how do I specify the channel? Ah, no, I see what you're
doing. I wanted to actually dial the channel number.
I came up with this:
; dial a long-distance call; allow the user to select a Zap channel
manually
exten = _88XX1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _88XX1NXXNXX,2,Dial(Zap/${EXTEN:3:2}-1/${EXTEN:4},30,o)
exten = _88XX1NXXNXX,3,Hangup
But I'll add the noops.
Course I have to fix the dialplan in my Poly600, too.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
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[asterisk-users] Acceptance testing of a new PRI
So I have these 4 new PRIs turning up tomorrow. Anyone have any suggestions on some dialplan that I could use to allow me to manually dial calls out over each channel for testing? I assume I'd have to make a separate group for each channel in the /etc/asterisk/zapata.conf? Or could I just specify the channel number directly in the dialplan and make 24 trunkgroups there with a dialpattern for each one? (I know enough to be dangerous, but not quite enough to implement without a little help. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly my Asterisk Box Hanged up all calls
On Wed, Jul 23, 2008 at 06:19:58PM +0200, Philipp Kempgen wrote: While it may sound rude that's absolutely correct. As a software developer in many cases you are more or less sure that an issue has already been solved so you expect the user to upgrade to the latest version or at least to the latest minor version. Having to hunt down problems in old versions is annoying especially for issues that have probably already been addressed. Not arguing. But please note the recently added section at the end of How To Ask Questions The Smart Way that says, in effect, Please Don't Bite The Newbies. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly my Asterisk Box Hanged up all calls
On Tue, Jul 22, 2008 at 06:39:28PM -0500, Tilghman Lesher wrote: On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote: My * version: 1.4.17 Please upgrade to 1.4.21.2. Just a suggestion, Tilghman: it might have been nice to add because it fixes your specific problem, so that we wouldn't assume because we don't want to talk to you if you rev is too old. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly my Asterisk Box Hanged up all calls
On Wed, Jul 23, 2008 at 10:44:12AM -0500, Tilghman Lesher wrote: On Wednesday 23 July 2008 10:15:18 Jay R. Ashworth wrote: On Tue, Jul 22, 2008 at 06:39:28PM -0500, Tilghman Lesher wrote: On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote: My * version: 1.4.17 Please upgrade to 1.4.21.2. Just a suggestion, Tilghman: it might have been nice to add because it fixes your specific problem, so that we wouldn't assume because we don't want to talk to you if you rev is too old. :-) It probably fixes his specific problem, AND because I don't like diagnosing an issue that we've already solved and that he would have figured out, if he had bothered to try the latest release. 1.4.21.1 should have been fixed, as well, but at that point, I had just spent 3 hours working frantically to get two security advisories out the door, so that the community wouldn't be vulnerable to two critical issues, and suggesting that he try a version that was vulnerable would have been bad. Oh, sure. I'm just sayin... It's pretty clear to me that while you're playing in the NFL, he may not be. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Recording tools
On Mon, Jul 21, 2008 at 01:36:10PM -0400, Alex Balashov wrote: OrecX comes with a GUI. Now, I won't refrain from allegations of braindeath related to its design; it is some gargantuan JSP/servlet-driven monstrosity that could have been reproduced in probably 50 lines of PHP or Perl. I've never seen anything else that looks quite so much like a Java web development fanboy's work on a rooftop, in a snowstorm, to which - along with good software development practice - he was oblivious because he was loaded up on meth. But it does work, you might say. Don't hold back, Alex. Tell us how you /really/ feel. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT Astricon/Digium Beach Ball Mailing
On Sat, Jul 19, 2008 at 03:40:46AM -0400, Alex Balashov wrote: Steve Totaro wrote: I post this not to put down Digium, the thought was nice, I wish I could play with my Digium beach ball, but Digium should know about it if it was common. Postage alone was costly. I mean this without a hint of sarcasm or derision toward you or Digium, but: Award for ... most bewildering asterisk-users list post ever! :-) Makes perfect sense to me. Matt F had one of them at our local users group meetup last week. They forgot to put the city on them, though at least they had the full date. They did seem kind of cheaply made... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Experience with Vicidial
On Wed, Jul 16, 2008 at 08:33:44PM -0400, Ein Bielaczyc wrote: I have a small customer looking to update their aged telemarketing system. I ran across astGUIclient and Vicidial (http://astguiclient.sourceforge.net/) during a Google search and was wondering if anyone had any experiences to share, positive or negative. Well, I'm not quite as biased as Matt... :-) (For those not playing the home game, I've been the network admin at VICI Marketing, where VD was written, for about 4 months now...) It's not bad. It's very big and powerful, and therefore it's a touch complex, which is unavoidable, but by and large, it Just Works: most of the problems I see here are peripherals issues, rather than the core. We run 240ish seats, which we're about to expand to 260ish, on 12 fronter servers and one Core2Quad closer server (which handles over 70 seats by itself); all closers record all calls (tmpfs). Is it perfect? No. But a lot of the problems I have with it come more from the fact that I'm new to it, I think, than that it's bad. Let's just say that managing that for production and 30 other servers and 40 other Windows machines leaves me plenty of time to keep up on Slashdot. :-) If there are any specific questions you have, lay 'em on me. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Magnetic door locks
On Thu, Jul 17, 2008 at 07:43:44AM -0500, c james wrote: I have an opportunity to interface asterisk with a security system to open their magnetic door locks. The security system needs a dry contact close upon activation to signal the door. Has anyone done this before? Worst case, Ciarcia's Circuit Cellar magazine has an ad this month for a TCP to RS-232 box for something like 69 bucks; you could probably use the RTS line to drive a relay. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
On Thu, Jul 17, 2008 at 11:34:30AM -0400, Kristian Kielhofner wrote: Could someone please explain to me why business desk phones are so expensive? I'm not knocking my friends over at Polycom, Snom, or any other manufacturer but in some cases you can buy a cheap but usable laptop for less than you can buy a phone. What gives? Production quantity S-curve Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] show channels concise parsing script?
I'm about to go off and try to write a script that parses the output of show channels concise so that I can get something readable (since show channels doesn't help me much either)... unless someone else has already done somethign similar and wants to share? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme replacement with native 729 support
On Tue, Jul 15, 2008 at 04:43:44PM -0400, John covici wrote: http://en.wikipedia.org/wiki/?-law_algorithm OK, thanks -- I was a bit puzzled because if I want to play audio over asterisk, it has to be 16-bit 8khz signed, so that is what I thought ulaw was -- thanks for the clarification. Nope. mu-law, and it's European counterpart, a-law, are standard PCM lin-log compression algorithms that date back to the beginning of T-1 service, if not even further... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
On Fri, Jul 11, 2008 at 03:59:22PM -0500, Joe Greco wrote:
On Fri, Jul 11, 2008 at 12:58:59PM -0700, Daniel Hazelbaker wrote:
Really? You have an RJ-21X block that contains both analog AND T1
wires? That's really uncommon. I hope they at least put the red
special service caps on the T1 wires.
Yup. I thought that pretty funny myself. 10 year old analog wires
running a digital T1. :) And they do have some caps on them, I think
it was red but not 100% sure.
No, that's not the unusual part. The unusual part is just that both
analog and digital services are on the same block. Maybe it's a
regional think...
That's really not unusual. It's not /preferred/, but that's an entirely
different can of worms.
I'll bet. :-)
In general, if copper is available into a building, the telco is going to
look very seriously at the possibility of using that. If the building is
already wired and the copper tests clean, the telco will want to use that.
In most existing situations, that will already be terminated in a can with
lightning suppression and will have been crossed over to RJ21X's that are
going to whatever suites are in the building.
So we don't pay a lot of attention to Tx and Rx in separate jackets,
or shielded anymore? Or is so much T-1 delivery over 1-pair HDSL that
no one cares anymore?
Since the telco will have /no/ /problem/ running the T1 over their outside
plant and up to the can on what is approximately Category 3 wire, and the
T1 signal is going to have been running alongside those same analog wires
for probably a few miles, what happens next should be obvious.
Cat 3 is optimistic, IME. Cat 2 is good enough for T-1, though; I
looked once.
Suite 214 wants a T1. There's already a 25-pair going up there from the
RJ21X. It's second story, so do you go and spend an {hour, afternoon,
etc} figuring out how to run fresh wire, or do you notice that only 6 pair
are in use on the RJ21X, and decide to feed up on the existing cable?
Now, if you're nasty and you don't separate it (typically I see the bottom
used for data) and you don't put redcaps on, yeah, then that is just
looking for eventual trouble. And who knows, the wire may be cruddy, so
maybe you still end up doing the separate run. But it probably works.
I've seen this often enough. Would I prefer to see new cable run? Sure.
But we've all done our copper sins. I've seen a lot of things that are
uglier than that. Here's one of them:
http://www.sol.net/hallofshame/
Slithering jesus. :-)
(I've always meant to expand that page, but it seems that I never get the
good photos of bad stuff)
I was going to ask...
Lack of space, lack of need, lack of having another RJ21X in the truck are
just a few other obvious reasons that this might be done.
True.
Your netmon link is 404, BTW.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
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Re: [asterisk-users] US T1 Hangup Detection
On Tue, Jul 08, 2008 at 11:13:02AM -0700, Daniel Hazelbaker wrote: D-Marc that terminates the 25-pair analog line coming in (this does not just contain our lines as I can tap into other peoples lines and hear there conversations, love security). The T-1's aren't on that, though, right? Next to that is a box with 4 slots for T1 cards, we used to have a T1 internet connection and its card is still in there. Slot 2 has the flex-grow T1 card in it. That's the smartjack, in the box. One of the pairs from the D-Marc goes into this T1 card and it Really? You have an RJ-21X block that contains both analog AND T1 wires? That's really uncommon. I hope they at least put the red special service caps on the T1 wires. provides a RJ-45 connection for the T1 line that runs either to the Adtran or to our Digium T1 card. Probably an RJ-48, actually, but who's counting. :-) I hope that answers the question, as I am not entirely sure what a shelf or smartjack are. Though I will feel really stupid if you say a shelf is something you store stuff on. See above. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
On Fri, Jul 11, 2008 at 12:58:59PM -0700, Daniel Hazelbaker wrote: Really? You have an RJ-21X block that contains both analog AND T1 wires? That's really uncommon. I hope they at least put the red special service caps on the T1 wires. Yup. I thought that pretty funny myself. 10 year old analog wires running a digital T1. :) And they do have some caps on them, I think it was red but not 100% sure. No, that's not the unusual part. The unusual part is just that both analog and digital services are on the same block. Maybe it's a regional think... I may have figured out the problem this morning, but I won't be able to test for a few days (again, aggravating that the only T1 line I have to test with is the live one). I noticed this morning while telneted into the Adtran that when I hangup on our normal incoming lines the Receive A bit toggles. I then noticed that two of the lines do NOT toggle the RA bit during hangup. These happen to the be last two lines in the rotary so I would not normally get incoming calls and complaints on them. They also happen to be the lines I was using to do my testing with. Grrr. I called Verizon and opened a ticket for why those 2 lines are behaving differently and that sounds like the problem, but I won't know for sure until I can test and try calling on one of the lines that does toggle the RA bit. As soon as I get that tested I will report that, though I expect that should fix the hangup issue. Aha! Good luck with that. Cheers, - jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
On Tue, Jul 08, 2008 at 10:00:08AM -0700, Daniel Hazelbaker wrote: Again, I am pretty sure T1. It is a Verizon Flex-Grow package, The Flex-grows I've seen were indeed T1, ESF as I recall the lights on the Adit 600 they terminated them into. Daniel: did Verizontal supply you with a shelf? Or just the smartjack? -- j -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Telco MWI with Asterisk 1.6-beta9
On Mon, Jun 23, 2008 at 09:44:47AM -0500, Jerry Jones wrote: Actually CLASS MWI, ie FSK, is a standard feature of Telco COs. It may be used with stutter, they are not exclusive, check with Verizon. I designed and manufactured these 20 years ago and Kevin is correct it uses same technology as CallerID known as CLASS. The analog styles, which I also designed and manufactured, are only found on PBX systems. There are several flavors that all signal based on some voltage mechanism. Well, not to pick a nit or anything... CLASS stands for Custom Local-Area Signalling Services, a name applied by (I think) Bellcore to a collection of New Added Spiffiness that started appearing on phones in the early 80s when the switches started to go digital. It's used (IME) as an umbrella term to describe a bunch of stuff: R.E.M's favorite, '*69', is called a CLASS code. CNID is a CLASS service, as was MWI. Distinctive ringing is a CLASS service. These services were define by various publications in Bellcore (now Telcordia)'s LATA Switching Systems Generic Requirements, FR-64, a spec document that RBOCs and CLECs bought and used to write RFPs for class-5 switches to switch makers who also bought them and used them to spec and build said switches. Last I looked it came in about 18 binders, and cost something like $15k to buy (er, excuse me: license). No, no educational discounts, (damnit :-). If you're the sort of geek who enjoys this stuff, there's a list here: http://telecom-info.telcordia.com/site-cgi/ido/index.html Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Controlling cell phone VM / Fax waiting notification icon for asterisk VM
On Mon, Jun 23, 2008 at 08:24:42AM -0400, Matt Watson wrote: Now the tough part...does anyone want to create an app to send notification to a cell phone to set/clear these bits? could you provide a link to where you got the info from? I'd be interested in seeing if i can get this to do anything useful. Good luck. The strong possibility exists, though, that you'll find the customer-exposed API doesn't have a path to set the bits you need. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendations for Motel Instalation.
On Sat, Jun 21, 2008 at 01:47:47PM -0500, Lyle Giese wrote: I agree with using used chan banks off of Ebay, but I would not touch a Zhone. I had one and sold it as soon as I could. They are a real PITA to program and don't pass caller id. Yeah, I gather they're a bit touchy to program. We need ours all the same way, and have a script that Matt wrote to put them there. As for CNID, I wouldn't know; I dont need it, and therefore haven't tried. IME, though, analog hotel phones don't have displays anyway. I have purchased several Adtran chan banks and have been extremely happy with them and tech support at Adtran. Support from Adtran has been nothing short of excellent even though they knew I calling about used chan banks from purchased on Ebay. One note is if the admin/craft interface has a password on it, you have to call Adtran to reset it. There is a way to bring up a numeric challenge code and support will tell you the response to it and you are in. Yeah, Adtran builds good stuff. I hear decent things about the Adit's, as well; they're certainly all pretty and modular... Challenge/response is a nice idea for that; I hear the Adit's are a bear to break into. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendations for Motel Instalation.
On Fri, Jun 20, 2008 at 03:42:28PM -0600, Arturo Ochoa wrote: I have a customer who owns a little Motel, and he wants to upgrade to a Asterisk PBX. There is one analog phone per room (aprox 80), and the cable is CAT 3. You might want to consider snagging an FXS channelbank off of eBay (we use the Zhones, which work pretty well for us), and using a multi-port T-1 card. If this is not a business motel, you'll likely get by with 24 trunks, so a quad-T card would support both your incoming lines and 3 channel banks (we seem to pay about $180-240 for them, making this cost effective), assuming approximately 80 isn't more than 72. :-) If that's not enough ports, then yeah, you'll probably be best served going to a Ethernet gateway; I personally have never liked the idea of stuffing that much FXS inside a PC chassis. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting Directory Behaviour (not)
On Wed, Jun 18, 2008 at 05:27:04PM -0500, Tilghman Lesher wrote: Here are the details: If caller enters only three digits/letters: Jane Smith, Extension 123, If this is the person you are looking for... If the caller types in more than three letters, the person's name is not spoken, and the caller hears: Extension 123, If this is the person you are looking for... Callers, not hearing the person's name, have no idea if extension 123 is the correct extension and so are reluctant to confirm without hearing the person's name. What's with this? From the customer: Annoying that people aren't following the directions and only entering 3 digits, but we've had some high level meetings here with a string of clients coming through in an unusually compressed frequency. And I've had 5 complaints over 2 days that callers couldn't find Jane Smith. The issue is that the 4th digit is actually interrupting the playback of the name, which is why they're not hearing it. Simple training issue. Or alternatively, you could play the name with DTMF-cut-through disabled, assuming that's not down inside C code... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting Directory Behaviour (not)
On Thu, Jun 19, 2008 at 03:49:01PM -0500, Tilghman Lesher wrote: On Thursday 19 June 2008 13:38:05 Jay R. Ashworth wrote: On Wed, Jun 18, 2008 at 05:27:04PM -0500, Tilghman Lesher wrote: Annoying that people aren't following the directions and only entering 3 digits, but we've had some high level meetings here with a string of clients coming through in an unusually compressed frequency. And I've had 5 complaints over 2 days that callers couldn't find Jane Smith. The issue is that the 4th digit is actually interrupting the playback of the name, which is why they're not hearing it. Simple training issue. Or alternatively, you could play the name with DTMF-cut-through disabled, assuming that's not down inside C code... That's a non-starter. Power users like to be able to interrupt prompts and press '1' immediately when they are sure that they've got the right person. Changing it now would be considered a regression to many. Or, alternatively, you could play the first $USERCONF milliseconds of the name with DTMF-cut-through disabled. This is akin to modal dialogs that pop up with the default button disabled for $LONGER-THAN-THE-AVERAGE-HUMAN-REACTION-TIME so that you don't keep on typing from whatever window you were previously in, and accidentally hit the Yes, delete all my files and kill my wife [OK] button by typing ENTER before you figure out what happened. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
On Tue, Jun 17, 2008 at 12:00:18PM -0400, C F wrote: On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote: Happens in the commercial world all the time; it's a common way to get cash out of the corporation -- a business's building is owned by the corporation's owners, and rented to the corporation. This is actually illegal in some states and considered a breach of Fiduciary everywhere. May be, but I know at least 3 owners of private corporations who are doing it, and their auditors seem fine with it. I think that it matters whether the corporation is public or not... LLCs? No, my assertion was that I believe that 'Steve's assertion that it is illegal and a breach of duty for a corporation's officers to own its real estate and lease it back to the company' may be dependent on whether the company is publicly owned or not. I suspect that there is no breach in the case of a private company, because different fiduciary duties pertain. I'll ask my client who's the ex-president of one of the companies I was talking about. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
On Tue, Jun 17, 2008 at 01:05:59PM -0400, Steve Totaro wrote: On Tue, Jun 17, 2008 at 1:02 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 12:00:18PM -0400, C F wrote: On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote: Happens in the commercial world all the time; it's a common way to get cash out of the corporation -- a business's building is owned by the corporation's owners, and rented to the corporation. This is actually illegal in some states and considered a breach of Fiduciary everywhere. May be, but I know at least 3 owners of private corporations who are doing it, and their auditors seem fine with it. I think that it matters whether the corporation is public or not... LLCs? No, my assertion was that I believe that 'Steve's assertion that it is illegal and a breach of duty for a corporation's officers to own its real estate and lease it back to the company' may be dependent on whether the company is publicly owned or not. I suspect that there is no breach in the case of a private company, because different fiduciary duties pertain. I'll ask my client who's the ex-president of one of the companies I was talking about. Please don't attribute quotes to me that I did not make or even assert. It was CF and maybe Alex I'm entirely sorry, Steve; you're right. I misread it. It was CF who made the assertion I was commenting on. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
On Sun, Jun 15, 2008 at 01:25:18PM -0400, Alex Balashov wrote: Is there a contradiction between them? Naw; Steve's just showin' his ass again. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
