Re: [asterisk-users] CDR changes in 1.4.3?
On 4/27/07, Steve Murphy [EMAIL PROTECTED] wrote: I'm the guilty party. I've been trying to fix several CDR bugs, involving stuff like missing times, missing changes in state (like NO_ANSWER when the call was ANSWERED), etc. A-HA! Don't get me wrong, I am not opposed to progress as there have been a few CDR quirks that were annoyances to me as well. The result is that several more cases are more accurate, but also, that rather uninteresting CDR's can be generated. In contemplating what could be done to get rid of some of these, I sometimes have to ask, is this truly something we have to get rid of?... In the meantime, uninteresting CDR's with NO_ANSWER and billsec=0, should be easy to filter out, right? You're right, they can/will be easy to filter out, I'll update my script that pulls CDR data for me to do it. I will, in the coming days, look at some of the extraneous CDR's that are generated, and see what I can do to get rid of them. It's not always that simple. If we ring a phone, for instance, and no-one answers it, is that truly, really, something that no-one will ever be, could ever be, interested in? (just a fer-instance). I do think there is a potential desire for some people to have these records. In my experience, however, unanswered calls are logged as well even before 1.4.3. Perhaps it is related to my trunk configuration. What is wholly uninteresting is, as you mentioned above, billsec/duration = 0 calls terminating to s in each context associated with the call and I'm not really sure what they could be used for but I'm sure somebody could find something. Perhaps a flag could be set to request regular, verbose, or very verbose CDR? Regular could provide behavior similar to pre-1.4.3, and verbose/very verbose could add more and more detail. Would it be simple enough to identify which CDRs are trivial and only log them when the verbosity is set higher? I know, that's easy for me to say, I'm not the one who has to code it up. :) I welcome your input. Complain up a storm. I'll try my best to make you happy. Thanks for the positive attitude. We do really appreciate the work you guys are doing, even if it doesn't seem like it at times. :) As I mentioned above, my only suggestion would be to identify CDRs that are informational in nature and only log them when a flag is set. Thanks again. sl ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR changes in 1.4.3?
Hello all: I upgraded to 1.4.3 last night and use MySQL for CDR. I have noticed that 1.4.3 seems to log a lot of crap to CDR that 1.4.2 did not. I use a few macros in my dialplan to handle outgoing calls (lcr type stuff) and in addition to the proper CDR for the call itself I also have records to 's' in the same dest-context and entries to 's' in the default context. Up to 3 CDRs are generated for one outgoing call (SIP - Zap channel) with one being the legit CDR and two being the type described above. My dialplan executes a ResetCDR after calling the lcr macro so that the CDR is sane and accurate, however, it appears these spurious CDR entries are generated by the call the ResetCDR even though I do not call it with any options. Am I missing something obvious here? I have read the ChangeLog but I didn't see anything that addressed this particular issue. Thanks for the help. sl ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI DCHAN Errors
On 4/5/07, Rob Schall [EMAIL PROTECTED] wrote: Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82b8430', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82d9920', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82369f0', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x8242968', 10 retries! Apr 4 12:13:06 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:09 NOTICE[15466] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) Apr 4 12:13:11 NOTICE[6670] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 I have read that some people believe this is a driver problem, which others believe something could be wrong with the PRI itself. I did a zttool and there are and haven't been any red alarms. I also read some people believe this issue can be caused by another card in the box taking too long to respond, such as an IDE card. We have 2 sangoma PRI cards and 1 sangoma FXO/FXS card. Any thoughts? FWIW I recently had some real troubles with something very similar, including the HDLC aborts, congestion, and terrible voice quality. Fortunately I caught the problem before the workweek so I could get it working. A little backstory: I had been running Linux 2.6.17-gentoo-r8 with Asterisk 1.2.14, Zaptel 1.2.12, libpri 1.2.4 and iaxmodem 0.2.0. As I was preparing to move all of my users over to Asterisk I wanted to upgrade before I moved them over so that I was on 1.4.x first. I also figured it wouldn't be a bad time to upgrade the kernel. (I know, one thing at a time) I upgraded to 2.6.19-gentoo-r5, Asterisk 1.4.2, Zaptel 1.4.1, libpri 1.4.0, and iaxmodem 0.2.1. After completing the upgrade the system would occasionally appear to hardware lock only to return to normal after 15-20 seconds. No process accounted for 100% cpu, however. It didn't do this all the time, maybe once every few hours. Of course this lockup would cause the PRIs on the system (quad port T1 card) to go a little nuts. VoIP calls would stutter terribly. I ended up in a rollback-reupgrade process that left me with kernel 2.6.17-gentoo-r8, the original kernel, but with the upgraded applications and libraries. I'm not convinced the problem was 2.6.19-gentoo-r5 as I also tried a 2.6.20 kernel and I have not seen anyone else asking about this problem but the evidence is pointing that way. During this process I thought for a moment I had lost my TE410P as it stopped talking to Asterisk but would talk to ztcfg/zttool/etc. Debug on the span showed Sending Set Asynchronous Balanced Mode Extended endlessly. I rolled back to my previous application setup and the server came up fine. So I rolled forward again with the apps without the kernel upgrade and it appears to work fine now. I'm running * on an IBM x306 server, 8836-1SU, specifically, with a TE410P. At some point I will attempt to upgrade the kernel again and see if that was the problem or I just had something screwed up. I don't know if that will help you but that was last weekend's hours of fun for me. :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)
On 3/22/06, Andrew D Kirch [EMAIL PROTECTED] wrote:
exten = s,2,GotoIf($[${CALLERIDNUM}300]?s,5);since 1xx is the pattern match for internal extensions anything lessthan 300 has to be internal so we already know that that is theextension they are wanting to forward
Downright hilarious. Great email.Food for thought, however. Is it possible to access this context externally? If so, someone with the ability set their own callerid (many people) could potentially redirect calls without authorization. Another possibility is that nefarious co-workers could use another's station to change call forwarding without authorization.
Perhaps adding an authorization routine wouldn't be a bad idea. Simply depends on how paranoid you and/or your users are.Sorry if these thoughts have been covered, just thinking about it from a different perspective.
sl
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Re: [Asterisk-Users] Please help with install *
I tihnk you may be risking the ire of the list. :) My suggestion would be for you to use something like [EMAIL PROTECTED], a pre-built install for Asterisk that requires minimal work on your part to get linux up and running. See http://asteriskathome.sourceforge.net/ for more information. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codec issues
On Tue, 21 Dec 2004 01:10:51 +0200, Shoval Tomer [EMAIL PROTECTED] wrote: Now the calls sounds fine, but the bandwidth is uses is near 20K instead of just 6K (both phones are near me, and the Asterisk server is at a remote location, and I can monitor bandwidth usage in my FW). I don't think you'll ever really see just 6 Kbps on the wire as IP, UDP, and RTP overhead for a 30 ms sample is around 10 Kbps all by itself. On an 8 Kbps codec like G.729 it wouldn't be unusual to see 18 Kbps of bandwidth used by the conversation. If it were using GSM you should see about 29 to 30 Kbps as it is a 13 Kbps codec plus 50 packets worth of overhead, 16 Kbps. (20 ms samples) I don't know why you're not seeing the same codecs indicated on both sides but it looks to me like you are getting G.729. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NPA NXX data
On Fri, 17 Dec 2004 20:35:49 -0500, Jon Bebeau [EMAIL PROTECTED] wrote: HI all - I know, slightly off list, but.. I'm looking for a NPA NXX database with City and State. Actually it's for an Asterisk routing app I'm working on. I see several vendors that want a few bucks to those that want an arm and leg. I expect this is published somewhere by some government agency, but Google hasn't got me to it yet. Search by my name with the list. I posted a list of the zip files plus a mysql table definiton for such a database a few months ago. sl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SpanDSP 0.0.2pre6 undef symbol on gentoo-ppc
Hello all: In a quest to resolve the last apparent timing problems I am having with SpanDSP I am trying out some PPC hardware, specifically a BW G3. I am running gentoo-ppc on this system and have compiled spandsp 0.0.2pre6 along with a release version of asterisk. Zaptel device (T100P) is configured and appears to be ready to go. When loading asterisk it crashes out with the following error: [app_rxfax.so]Dec 2 13:47:57 WARNING[10027]: loader.c:248 ast_load_resource: /usr/local/lib/libspandsp.so.0: undefined symbol: top_bit Google reveals nothing of substance related to this error. An Intel system running recent gentoo with the same tiff and audiofile packages works just fine for me. As I am not a programmer by any means I am at a loss of where to go from here. Perhaps I am missing something rather simple but I just don't see what it is. :( Any help would be appreciated. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Top posting
On Fri, 12 Nov 2004 18:57:05 -, Kevin Walsh [EMAIL PROTECTED] wrote: Perhaps you should adopt a more flexible attitude and learn to follow up properly. There is no excuse at all for lazily top-posting. This from the resident signature size champion. Lay off of it. I've seen this goofy flame several times from you, every time accompanied by your childish and egotistical signature, but do you see us flaming you for it every time you post? People will do things that annoy you, that's life. Perhaps you should adopt a more flexible attitude in accepting other people's idiosyncrasies instead of flaming everyone who may cause you distress by top posting. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpanDSP + Lexmark 6170 = Cut off faxes?
On Tue, 09 Nov 2004 07:41:36 +0800, Steve Underwood [EMAIL PROTECTED] wrote: I have looked at many similar reports, and all but one turned out to be due to data slips. However, if you are sure this is only happening with one particular FAX machine your problem might be different. In t30.c uncomment the first line. Rebuild spandsp and try FAXing. You should get audio log files in /tmp. Send them to me, and I will investigate. Steve: It appears I shouldn't have spoken so soon, I apologize; that is, without a larger sample of faxes to observe. I was concerned about this particular machine because it fails on every page but in reviewing a sample of about 45 faxes this morning it appears that I am having trouble with other machines as well but not to the same extent I have with the Lexmark. The Lexmark fails on *every* page while some others fail only on one or two pages out of a fax, but still most others work fine. For what its worth it also looks like the failures occur more frequently on faxes from longer distances away, I'm in Pennsylvania and the Lexmark is in Florida. It also appears that a machine in California is having an occasional problem with me too. Before wasting any more of your time tracking down a problem that likely doesn't exist in the software I will approach it as a timing problem first. On that tack, what should I look for as a potential source of a timing problem? As I indicated previously I see this problem even when I am connected directly to the PSTN and (believe) that I am using the PSTN as a clock source on the PRI. Assuming that the clock from the telco is good should I start looking at the motherboard/system itself as a source of a timing problem? I have verified that the T1 card is on its own interrupt. Perhaps the motherboard just isn't cut out for handling this volume of interrupts? Thank you for the help. sl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SpanDSP + Lexmark 6170 = Cut off faxes?
Hello all. I'm running SpanDSP 0.0.2pre4 with Asterisk v1.0 (10/23 from CVS) and am having trouble with receiving cut off faxes from a Lexmark 6170 on the distant end. I'm running this on a Tyan S2420 board that is about four years old with a P3-800 and 256 MB on it. I have tested with a PRI connected to both my Nortel Opt 11C and directly to Verizon's (LEC) network. I have the same problem with either configuration, which I am hoping rules out timing at least in relation to the PRI and PSTN. I receive between 300 and 400 rows of pixels of each page and then it cuts off and goes to the next page. There are a few rows of identical data then an abrupt end of the page. Then the next page starts out looking great only to suffer the same fate. The proper number of pages appears to be received but only ~350 rows of pixels worth. This Lexmark 6170 is the only machine I have had trouble with so far. All others appear to be able to send to me properly and completely. For what it is worth I am including the tiffinfo on the received fax file. Any help or pointers would be appreciated. sl -- tiffinfo TIFF Directory at offset 0x153e Image Width: 1728 Image Length: 403 Resolution: 77, 38.5 pixels/cm Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Date Time: 2004/11/08 17:27:40 Host Computer: cs Image Description: 2125551212 Orientation: row 0 top, col 0 lhs Samples/Pixel: 1 Rows/Strip: (infinite) Planar Configuration: single image plane Page Number: 0-3 Software: spandsp Group 3 Options: 2-d encoding+EOL padding (5 = 0x5) Fax Data: receiver regenerated (1 = 0x1) Bad Fax Lines: 10 Consecutive Bad Fax Lines: 7 Fax Receive Time: 1115882420 secs TIFF Directory at offset 0x2b00 Image Width: 1728 Image Length: 335 Resolution: 77, 38.5 pixels/cm Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Date Time: 2004/11/08 17:27:50 Host Computer: cs Image Description: 2125551212 Orientation: row 0 top, col 0 lhs Samples/Pixel: 1 Rows/Strip: (infinite) Planar Configuration: single image plane Page Number: 1-3 Software: spandsp Group 3 Options: 2-d encoding+EOL padding (5 = 0x5) Fax Data: receiver regenerated (1 = 0x1) Bad Fax Lines: 18 Consecutive Bad Fax Lines: 16 Fax Receive Time: 1115882388 secs TIFF Directory at offset 0x418c Image Width: 1728 Image Length: 337 Resolution: 77, 38.5 pixels/cm Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Date Time: 2004/11/08 17:28:08 Host Computer: cs Image Description: 2125551212 Orientation: row 0 top, col 0 lhs Samples/Pixel: 1 Rows/Strip: (infinite) Planar Configuration: single image plane Page Number: 2-3 Software: spandsp Group 3 Options: 2-d encoding+EOL padding (5 = 0x5) Fax Data: receiver regenerated (1 = 0x1) Bad Fax Lines: 18 Consecutive Bad Fax Lines: 9 Fax Receive Time: 1115882388 secs ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phones -India
On Wed, 20 Oct 2004 17:27:38 -0500, Henry Devito [EMAIL PROTECTED] wrote: HI I am in the US and have a customer using * in the US they just acquired a call center in India. Does anyone know if I can legally sell/ship Grandstream IP phones and IAXy's to India? I am sure you can legally ship them to India but you'll find a difficult time getting them through customs unless you plan ahead. Make sure the Indian call center is set up properly under Indian tax law so that they are not liable for any duties, have someone from the call center in contact with customs before you ship and have them go to the port of entry the day they arrive and try to get them walked through. In my experience it is best to use air freight or to fly them in as hold baggage yourself. If you just ship them over you might find they get caught in customs or disappear. I've also seen equipment destroyed by Indian customs too by leaving it exposed, so make sure the equipment is sealed to prevent moisture getting in. I don't mean to discourage you, it certainly can be done, but you *must* do your research first and follow through properly in India. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Embedded Asterisk System (was QoS Router/Software Suggestions)
On Wed, 13 Oct 2004 11:36:09 -0700, Geoff Nordli [EMAIL PROTECTED] wrote: When Bering boots it loads everything into a RAM based filesystem. Anyone see any drawbacks in running an * system using a RAM based file system and booting from CF? If you needed additional capacity for VM you could add an IDE drive. The benefits to not using an IDE drive would mean one less mechanical device prone to failure. You'll want some way to keep CDR records if you need them. I wouldn't replace the IDE drive for VM with a CF as you might run into a CF failure to due excessive writes per sector. It seems CFs are good for something on the order of 10k writes per sector, a number that might be deceptively easy to achieve. Now, booting from CF and writing CDR back once (or a few times) per day might be a worthwhile solution while using a remote * or disk for VM. Of course an IBM 1 GB Microdrive is on the order of $175 or so and would provide the ability to run the OS, * and VM all from disk without worry or kludge. sl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing Fun - anybody know where to get aNPA/NXX db?
On Fri, 24 Sep 2004 15:02:53 +0200, Patrick [EMAIL PROTECTED] wrote:
Would you mind sharing the perl script and the database schema?
Perl script and database layout are below. Its not pretty since I
never intended it for external consumption but it does get the job
done.
If you unzip the files you get from the previous post I made you need
to run this perl script against each file individually, ie script.pl
esutld.txt
I'm using npa-nxx as the key in the database as it is much much faster
to query based on the key than to do a npa=212 and nxx=555 type
query with some other field as a key. On the P3-800 I'm hosting this
on right now the difference was 20 fold, from 0.2s to 0.01s.
Also, as far as I know, NANPA doesn't list individual carriers in NXXs
that are split up, for example NXXs where several different carriers
are assigned blocks of 1000 numbers. This database set up certainly
doesn't take that into account. I would assume, however, that rate
center information would be the same for all numbers in a particular
NXX.
Longer term, access to a LNP database may be needed to accurately
determine rate centers. For example, in my town, State College, PA,
the A and B cellular carriers do not have exchanges in the State
College rate center but rather in nearby rate centers that are local
calls. The A carrier is in Boalsburg and the B carrier is in
Bellefonte, each a distinct rate center, but it is possible to port
numbers between them. This means that someone in an outlying rate
center here, for example, Port Matilda, could believe they are making
a local call to a Bellefonte mobile number but since that number has
been ported to the other mobile carrier it becomes a regional toll
call to Boalsburg.
I don't know how the big telcos are dealing with this but I can't
imagine they're giving up the opportunity for revenue.
sl
-- perl script
#!/usr/bin/perl
use DBI;
my $dbh = DBI-connect(DBI:mysql:pbx:localhost, user, pass) or
die $DBI::errstr;
my $sth;
open (IN,$ARGV[0]);
foreach (IN) {
s/\x22//g;
@npanxx = split(\t);
if ($npanxx[0] eq State) {
next;
}
# print $npanxx[1] / $npanxx[4], $npanxx[0] / $npanxx[3]
($npanxx[2]) $npanxx[5] / $npanxx[7]\n;
$sth = $dbh-prepare(INSERT INTO npanxx VALUES (\$npanxx[0]\,
\$npanxx[1]\, \$npanxx[2]\, \$npanxx[3]\, \$npanxx[4]\,
\$npanxx[5]\, \$npanxx[7]\ , NULL));
$rv = $sth-execute();
}
-- MySQL Table
CREATE TABLE npanxx (
state CHAR(2),
npanxx CHAR(7) NOT NULL PRIMARY KEY,
ocn CHAR(4),
company CHAR(52),
ratecenter CHAR(11),
switch CHAR(11),
usetype CHAR(2),
timestamp TIMESTAMP);
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Re: [Asterisk-Users] Local Outbound Calls on PRI
On Fri, 24 Sep 2004 10:06:06 -0500, Paul Oster [EMAIL PROTECTED] wrote: I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. Any chance you're sending more or fewer digits than the CLEC expects to see from you? I would expect that the tech would have picked up on it when he was watching your call attempts but just in case... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing Fun - anybody know where to get aNPA/NXX db?
The NANPA maintains this database and the full database is available in 8 files. I have imported this into a MySQL table and can query based on rate center and npa-nxx. I use wget -a using a file containing the following: http://www.nanpa.com/nanp1/cnutlzd.zip http://www.nanpa.com/nanp1/csutlzd.zip http://www.nanpa.com/nanp1/eautlzd.zip http://www.nanpa.com/nanp1/enutlzd.zip http://www.nanpa.com/nanp1/esutlzd.zip http://www.nanpa.com/nanp1/wnutlzd.zip http://www.nanpa.com/nanp1/wputlzd.zip http://www.nanpa.com/nanp1/wsutlzd.zip Then I have a perl script that reads each file in and puts them into a MySQL table. I hope this helps with that you are looking for. sl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer and Release of a call out to PSTN
On Thu, 16 Sep 2004 12:41:47 -0400, Christopher Jacob [EMAIL PROTECTED] wrote: When using Asterisk with a PRI to the CO is it possible to transfer a call back out and release. In other words, once the call is connected (caller and external 3rd party) Asterisk is removed from the equation thereby freeing the PRI channels. I ask because my scenario is going to require frequent external transfers and I would like to control the PRI costs. You're looking for a feature called Take Back and Transfer, TBT for short. It works by the telco always monitoring the trunks for DTMF from your end, for example, the TBT code might be *8. You would send *8,12125551212 down the line and the telco will pull the call back and send it to the number specified, in this example direct service for Manhattan. The last time I looked into it it was a specialized service and had a higher per minute rate than conventional termination. sl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Astersk as AVAYA IVR
On Mon, 13 Sep 2004 10:07:30 -0600, Jason Kawakami will require a path back to the Index. not sure what you mean by 'tromboning' but it may be in reference to using the same b-channel on the On tromboning and anti-tromboning, this is a feature that is part of the Nortel ITG system we are currently using on my voice network. (Which I am hoping to replace with * soon) I believe ITG has had this feature since its start, it certainly did when we first deployed it four years ago. What it basically is: It is a feature to detect if a call is being sent back to the system it originated from and to, in effect, transfer the call to its destination on the originating system and tear down the voice path to the remote system rather than opening a new voice path and having the call use two voice paths on a particular link when no one at the distant end is involved in the call anymore. The feature is very nice for use in a multi site environment where calls are transferred around or where latency is large. It saves on bandwidth and simultaneous path planning and helps a lot when sending calls back to a main office from a remote facility located on the other side of the world. To the original poster, you might want to see if you can configure * and the Avaya to allow you to transfer calls back to the Avaya system as though the * system was nothing but a group of extensions off the Avaya. You might not be able to do this with PRI but if the Avaya has the capability to do Off-Premise Extensions (OPX) via T1 you might be able to use that. Something like determine the route for the call and then transfer it to that queue. This should mean that you only need enough capacity between the Avaya and * systems to handle the maximum number of simultaneous IVR calls you expect at one time. This keeps * out of the middle of the call and means it is only used to determine a queue to enter. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - Experience using Gmail for AsteriskMailingList
On Tue, 7 Sep 2004 18:14:44 +0100, Kevin Walsh [EMAIL PROTECTED] wrote: You should probably read RFC 1855 (Netiquette Guidelines), which clearly defines a four-line limit for signatures. Perhaps you'd like to count the number of lines in my signature, or get someone to count the lines for you. IMHO it is rather egotistical to expect a free email list to propagate many multiple thousands of copies of your 231 byte signature. This egotism is multiplied by the fact that your signature is useless, it contains no information that is not otherwise available from the headers of your posting. I do not know how many people subscribe to this list but I would be unsurprised if it numbered in the 5,000 range. At such a level that means your signature accounts for 1.15 MB of data for every posting you make. Sometimes it is not about what you are allowed to do rather what is considerate of others, just like your request that others not top-post. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxy vs sipura
On Tue, 7 Sep 2004 15:40:00 -0500, Lyle Giese [EMAIL PROTECTED] wrote: Hmmm, I thought 56k modems were 56k outbound only and maxs at 33k inbound or did the standard change again when I wasn't looking? And besides when did you get better than 28.8 through a hotel PBX? 33k with a 56k modem in the real world is not that common. When I first read the above posts I thought as you did but found through some Googling that v.92 is 56k (53k) downstream and up to 48k upstream. Also, a note that the dialup connection will add in the neighborhood of 140-170 ms of latency. If this is on top of an already high delay connection you might find the conversation to be more of a walkie-talkie type rather than natural flow, assuming you are using a codec that will function over dialup to begin with. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
