Re: [asterisk-users] Web page to show online extensions?
This is probably coming into this thread a bit late and in the wrong place, however: http://www.drogon.net/dsx/onlineStatus.html is a page of PHP code I use in my systems. You end up with an array, indexed by extension number with values SIP, IAX, or SI (I create both SIP and IAX accounts in my system so they're interchangeable) It's maybe not the most elegant bit of code, but it works for me. Turning the contents of the array into a web page is left as an exercise to the user ;-) Gordon ___ -- 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] Asterisk with lumenvox
I had posted earlier asking about folks real world experiences with with Lumenvox, and the thread 'strangely' disappeared after some bloke from down under justed sodded himself over my straight simple questions. Hm- makes you wonder. Josué Conti wrote: Hello everyone. I wish I could continue with the approval of the engine Lumenvox, for voice recognition, but not a development of acceptable engine, Please could help me in achieving test? As I said earlier we have a project that will involve a very large number of licenses for Voice recognition, but I would count on help from Lumenvox, for this case. Could you help me? Best Regards Josué 2008/3/19 Josué Conti [EMAIL PROTECTED]: Hello everyone, Rodrigo and Philipp Hello, I would like to know how to properly configure the engine Lumenvox no asterisk, I am trying to dial by vox actually like that the user should dial for receipt of my business, is attended by an IVR system with voice recognition that allows the user to say who would like to talk and the asterisk foward the call. Set up the asterisk below, but the system recognizes the voice, but does not guide the call, running immediately after a hangup, what is wrong with my settings? I can not very material support on the issue, could help me? I am not really achieving great results in my tests with engine Lumenvox: I am trying to test a simple scheduling dialing by voice, where the system identify the user by name and system called in your phone number, but I am not able, could help me? If I did not say any word, the system is static, but if I say any Word, even different words grammar.gram (ura.gram) of the system Performs the following priorities file extensions.conf, please, can You help me? Best Regards Josué Our programming files are configured this way: Ipbx: / etc / asterisk # vim lumenvox.conf ; LumenVox configuration file [General] Servers = 127.0.0.1; Speech Engine Servers to use. Save_sound_files = no; Set to yes to save sound files for use with Speech Tuner [Grammars] ura = / etc / asterisk / grammars / ura.gram [Default] Vad_snr_sensitivity = 50 Vad_volume_sensitivity = 50 Vad_eos_delay = 1250 Vad_wind_back = 750 End_of_speech_timeout = 15000 Use_oov_filter = no ;; ;; Ipbx: / etc / asterisk # vim extensions.conf [General] [Globals] DYNAMIC_FEATURES = # pickupexten hangup atxfer # # blidxfer [Default] Length = 2000.1, Playback (Ura / instit / instit_casa) Length = 1515.1, Playback (Ura / parabens) ;; ;; ; Pilot URA Length = 6969.1, GotoIfTime (07:50-18:05 | mon-fri |*|*? ura, s, 1) Length = 6969.2, GotoIfTime (18:06-23:59 | mon-fri |*|*? ura, s, 1) Length = 6969.3, GotoIfTime (00:00-07:49 | mon-fri |*|*? ura, s, 1) Length = 6969.4, GotoIfTime (* | sat-sun |*|*? ura, s, 1) ; IVR URA ; [URA] ; Length = s, 1, Answer () Length = s, n, Wait (3) Length = s, n, NoOp (entry Ura) Length = s, n, Set (TRIES = 0) ; Length = s, n, ResponseTimeout (10) Length = s, n, BackGround (Ura /abertura) Length = s, n, Playback (beep) ; Length = s, n, BackGround (Ura / abertura1) Length = s, n, Goto (lumenvox-test, s, 1) [Lumenvox-test] Length = s, 1, Answer Length = s, n, Wait (1) Length = s, n, SpeechCreate () Length = s, n, SpeechActivateGrammar (Ura) Length = s, n, SpeechStart () Length = s, n, SpeechBackground (liggol / abertura) Length = s, n, SpeechDeactivateGrammar (Ura) Length = s, n, Goto (institutional, s, 1 - $ SPEECH_TEXT (0) ()) [Institutional] Length = s, 1, Playback (Ura / instit / instit) Length = s, 2, congestion (3) Length = s, 3, hangup ipbx: / etc / asterisk / grammars # vim ura.gram # - Grammar: ura.gram # ABNF 1.0; Language es-CO; Voice mode; Tag-format lumenvox/1.0; Root $ URA; $ Continent = ((Josue | Conti) []): 2000; $ Palms = () 1515 ; $ Ura = ($ conti | $ palms) = $ $ $ (); 2008/3/19, Rodrigo Gonzalez [EMAIL PROTECTED]: Josué Conti escribió: Hello all, how are you? I would like to know from someone uses or has used the engines of LumenVox for integration with the asterisk for voice recognition. Best Regards Josué ___ -- 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 I've configured for a customer. What do you need to know? ___ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] CentPBX mirror?
pbxinaflash.com (source based) I've used this before on other machines (on which it works perfectly), but the version on the website definitely doesn't include drivers for the SAS controller on the Dell R200. Elastix.com (rpm based) Not familiar with that one. Will investigate. trixbox.org (rpm based) Bit too heavy for what I'm after. Thanks! Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- 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] iaxmodem + hylafax w/ DID routing
Edwin Lam wrote:
in extensions.conf:
exten = _,1,Dial(IAX2/iaxmodem0/${EXTEN}|20|r)
exten = _,n,Dial(IAX2/iaxmodem1/${EXTEN}|20|r)
exten = _,n,Busy
exten = _,n,Hangup
according to some documentations i've found $CALLID4
Correct. Here is my extensions.conf:
exten = _[4-8]XXX,1,Macro(faxreceive,${EXTEN})
exten = _[4-8]XXX,n,Hangup()
[macro-faxreceive]
exten = s,1,Dial(IAX2/iaxmodem.com01/${ARG1})
exten = s,n,Dial(IAX2/iaxmodem.com02/${ARG1})
exten = s,n,Dial(IAX2/iaxmodem.com03/${ARG1})
exten = s,n,Dial(IAX2/iaxmodem.com04/${ARG1})
exten = s,n,Dial(IAX2/iaxmodem.com05/${ARG1})
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] iaxmodem + hylafax w/ DID routing
Edwin Lam wrote: in FaxDispatch: FILETYPE=pdf case $CALLID4 in 1000) [EMAIL PROTECTED] 1001) [EMAIL PROTECTED] *) [EMAIL PROTECTED] esac This is also incomplete, One of my entries with archiving of the PDF and TIF: case $CALLID4 in '5051') # ## Bankers Life/Conseco (Rose Parker) (Previously Louise Taylor)# # FILETYPE=pdf; [EMAIL PROTECTED]; /usr/local/bin/tiff2pdf $FULLPATH -p letter -o faxdata/$CALLID4/pdf/$FILENAME.pdf cp $FULLPATH /var/spool/hylafax/faxdata/$CALLID4/tif/ ;; Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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
[asterisk-users] IAX IP Phone
Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? Regards Bilal You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com ___ -- 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] CentPBX mirror?
On Sat, 5 Apr 2008, Chris Bagnall wrote: pbxinaflash.com (source based) I've used this before on other machines (on which it works perfectly), but the version on the website definitely doesn't include drivers for the SAS controller on the Dell R200. Is it worthwhile rolling your own? I've installed Debian on lots of Dell kit in the past with good results, including Dells SAS controllers. Gordon ___ -- 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
[asterisk-users] TDM400P UK CID ISSUE
Hi,
I know this issue has raised its head so many times before, and I have
been over so many threads, bug reports, mantis and other resources and
still unable to resolve.
I was using Asterisk 1.4.13 and have upgraded to 1.4.19 and was using
Zaptel 1.4.5.1 and now using 1.4.9.2
I found a bug tracking issue where someone has posted a UK CID fix
which appeared to work under 1.4.5.1 which was:
Index: wctdm.c
===
--- wctdm.c (revision 2300)
+++ wctdm.c (working copy)
@@ -315,6 +315,7 @@
#else
int wasringing;
#endif
+ int lastrdtx;
int ringdebounce;
int offhook;
int battdebounce;
@@ -859,30 +860,29 @@
return;
#ifndef AUDIO_RINGCHECK
if (!wc-mod[card].fxo.offhook) {
- res = wc-reg0shadow[card];
- if ((res 0x60) wc-mod[card].fxo.battery) {
- wc-mod[card].fxo.ringdebounce +=
(ZT_CHUNKSIZE * 16);
- if (wc-mod[card].fxo.ringdebounce =
ZT_CHUNKSIZE * 64) {
+ res = wc-reg0shadow[card] 0x60;
+ if (wc-mod[card].fxo.ringdebounce) {
+ wc-mod[card].fxo.ringdebounce--;
+ if (res res != wc-mod[card].fxo.lastrdtx
wc-mod[card].fxo.battery) {
if (!wc-mod[card].fxo.wasringing) {
wc-mod[card].fxo.wasringing
= 1;
- zt_hooksig(wc-chans[card],
ZT_RXSIG_RING);
if (debug)
printk(RING on %d/%d!
\n, wc-span.spanno, card + 1);
+ zt_hooksig(wc-chans[card],
ZT_RXSIG_RING);
}
- wc-mod[card].fxo.ringdebounce =
ZT_CHUNKSIZE * 64;
- }
- } else {
- wc-mod[card].fxo.ringdebounce -= ZT_CHUNKSIZE
* 4;
- if (wc-mod[card].fxo.ringdebounce = 0) {
- if (wc-mod[card].fxo.wasringing) {
+ wc-mod[card].fxo.lastrdtx = res;
+ wc-mod[card].fxo.ringdebounce = 10;
+ } else if (!res) {
+ if (wc-mod[card].fxo.ringdebounce ==
0 wc-mod[card].fxo.wasringing) {
wc-mod[card].fxo.wasringing
= 0;
- zt_hooksig(wc-chans[card],
ZT_RXSIG_OFFHOOK);
if (debug)
printk(NO RING on %d/
%d!\n, wc-span.spanno, card + 1);
+ zt_hooksig(wc-chans[card],
ZT_RXSIG_OFFHOOK);
}
- wc-mod[card].fxo.ringdebounce = 0;
}
-
+ } else if (res wc-mod[card].fxo.battery) {
+ wc-mod[card].fxo.lastrdtx = res;
+ wc-mod[card].fxo.ringdebounce = 10;
}
}
#endif
@@ -1462,6 +1462,10 @@
reg16 |= (fxo_modes[_opermode].rz 1);
reg16 |= (fxo_modes[_opermode].rt);
wctdm_setreg(wc, card, 16, reg16);
+
+ /* Enable ring detector full-wave rectifier mode */
+ wctdm_setreg(wc, card, 18, 2);
+ wctdm_setreg(wc, card, 24, 0);
/* Set DC Termination:
Tip/Ring voltage adjust, minimum operational current,
current limitation */
With this patch I was able to get fairly reliable CID from my TDM400P
card (Wildcard TDM400P REV I (4 modules)) , however this now fails to
patch against the latest Zaptel 1.4.9.2 and I am unable to get CID
working reliably - some calls do show the CID correctly .
This is what appears in the output more often than not:
[Apr 5 16:21:13] NOTICE[12685]: chan_zap.c:6191 ss_thread: Got event
2 (Ring/Answered)...
[Apr 5 16:21:15] WARNING[12685]: chan_zap.c:6254 ss_thread: CID timed
out waiting for ring. Exiting simple switch
So for some calls we get it, other times the CID is empty. When
plugging the DECT unit into the BT line I get CID perfectly every
time, so I am sure this is a driver/card issue.
So has anyone found a reliable way in the UK using one of these cards
on BT to show UK CID ?
I think I have all the right settings in the zapata.conf i.e
usecallerid = yes
cidsignalling = v23
cidstart = polarity
immediate = no
So where am I going wrong ? Sorry if this has been covered somewhere
else or a fix .. I am just unable to find it - and I am slowly loosing
hair !
Regards
Matt Brown
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Re: [asterisk-users] SellVOIP
At 09:42 PM 4/4/2008, you wrote: Common practice is to check every bill. Withing the last month, I have found two several hundred dollar mistakes on Credit Card and Checking account. I am nos sure if companies are charging extra to make up for the economy slow down or they are genuine mistakes, but I have never had these issues in the past besides a mistake here and there over the course of a year.. Good advice, but that wasn't what my message was about. They're a VOIP provider that I thought went out of business months ago or maybe a year ago with $13 of credit on my account. Today they re-appeared and my $13 is still there. Likely that's true for others too. Ira ___ -- 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
[asterisk-users] TDM410 Callerid UK
Hi all, Has anyone got any experience with getting a TDM410 to work with callerid in the UK, I've spent some time fiddling with the options but haven't made any headway. I've also contacted digium support who haven't been able to help either. Many thanks, James ___ -- 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] Advice on best operator phone (with attendant console)
We have been marketing ipPBX systems based on asterisk for 3+ years. For the last year we've been placing Aastra 57iCT with 560M sidecars. Our attendants like the idea of a cordless handset so the attendant can go to the copy room, etc. The LCD based sidecar means you can keep it up to date without marking up paper strips. We deploy Thirdlane PBX Manager which allows us to setup the BLF (busy lamp field) via a web interface. Aastra 57iCT: http://neobits.com/aastra_-_a1758-0131-10-05_-_57i_ct_p11471.html Aastra 560m: http://neobits.com/aastra_-_a1760--10-55_-_560m_p11472.html Thirdlane PBX Manager: http://www.thirdlane.com/products/pbxmanager Feel free to contact me off list if I can be of any assistance. Regards, Jim ph: 408-701-9929 Faraz R. Khan wrote: One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets stuck 4. Voice 'gaps' (blackouts) for 4-5 seconds 5. The phone also completely locks up regularly 6. ping response goes from 8ms to 3000ms (after which the phone locks up) Wondering which operator phone would work best. I have the following choices: 1. Linksys SPA 932/962 with attendant console 2. Polycom 601/650 with attendant console I cant confirm online whether the BLF functionality will work with Asterisk 1.2.26. Is somebody using either of these phones in a high volume environment successfully? Thank you. ___ -- 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] Ring back when free?
Hi, Am Freitag, den 04.04.2008, 13:03 + schrieb Tony Mountifield: In article [EMAIL PROTECTED], Faraz R. Khan [EMAIL PROTECTED] wrote: Thinking out loud: write a asterisk call file (when the calling user presses 5) which keeps on trying to connect the two. I thought about that, but the trouble is, it's not event-driven. It just keeps on trying until it runs out of retries. We realized someting like that with a call file. If a caller presses 5 store this as an open callback in a database. Place a script in the h-extension and call it with the DeadAGI-Application. This script looks up any pending callbacks for both parties of the closed connection and generates the call files. Regards, Karsten ___ -- 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] IAX IP Phone
On 04/05/08 05:16, bilal ghayyad wrote: Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? Regards Bilal I've not seen IAX phone so your best option will be IAXy adapter from digum. It works OK; but it is not free of bugs. -- #Joseph ___ -- 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
[asterisk-users] realtime errors
Hi All, I just started playing around with asterisk realtime, added some extensions and started making test call, sometimes i can call the extension sometimes i can't. below are errors i see on the CLI, has anyone encountered this before? [settings] sippeers = mysql,sipdb,sip_customer sipusers = mysql,sipdb,sip_customer extensions = mysql,sipdb,extensions_customer voicemail = mysql,sipdb,voicemail_customer [Apr 6 01:04:53] WARNING[18959]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. [Apr 6 01:05:04] WARNING[18959]: app_voicemail.c:2262 inboxcount: Failed to obtain database object for 'asterisk'! regards, nhadie You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com ___ -- 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] IAX IP Phone
Joseph wrote: On 04/05/08 05:16, bilal ghayyad wrote: Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? Regards Bilal I've not seen IAX phone so your best option will be IAXy adapter from digum. It works OK; but it is not free of bugs. Did you mean IAX2, this one is not bad; http://www.atcom.cn/En_products_At530.html -- Powered by Gentoo GNU/LINUX http://www.linuxcrazy.com ___ -- 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] IAX IP Phone
Atcom supports IAX: http://www.voip-info.org/wiki/view/AT-530 On Sat, Apr 5, 2008 at 11:17 AM, Joseph [EMAIL PROTECTED] wrote: On 04/05/08 05:16, bilal ghayyad wrote: Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? Regards Bilal I've not seen IAX phone so your best option will be IAXy adapter from digum. It works OK; but it is not free of bugs. -- #Joseph ___ -- 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 ___ -- 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] IAX IP Phone
This one on our website works fine too (at least from the feedback we have gotten): http://www.voipperiod.com/product_info.php?cPath=22_28products_id=268 //Peter david wrote: Joseph wrote: On 04/05/08 05:16, bilal ghayyad wrote: Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? Regards Bilal I've not seen IAX phone so your best option will be IAXy adapter from digum. It works OK; but it is not free of bugs. Did you mean IAX2, this one is not bad; http://www.atcom.cn/En_products_At530.html ___ -- 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] TDM400P UK CID ISSUE
On Sat, 5 Apr 2008, Matt Brown wrote: So has anyone found a reliable way in the UK using one of these cards on BT to show UK CID ? Upgrade to asterisk 1.2 ;-) I think I have all the right settings in the zapata.conf i.e usecallerid = yes cidsignalling = v23 cidstart = polarity immediate = no Thats good. So where am I going wrong ? Sorry if this has been covered somewhere else or a fix .. I am just unable to find it - and I am slowly loosing hair ! I posted about this some time back and got the same patch you posted here, but am only using 1.2 as yet. I got that patch over a year ago, so I find it odd that wctdm hasn't had it properly integrated by now... Gordon ___ -- 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] IAX IP Phone
bilal ghayyad a écrit : Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? We are selling IP0023 and IP0027 phones (IAX and SIP). Please contact off line if you're interested. -- Daniel ___ -- 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
[asterisk-users] Paging for analoge devices
Hi; Anyone knows (tried) to use Page for analoge phone(zaptel channel - fxs)? If yes, how? Regards Bilal You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com ___ -- 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] SellVOIP
Well, my $21 is still there and all of my calls are being declined. Over a year ago, I requested a refund and regardless of all promises that I would receive one, Jed never followed through. I'd use up the credit if the calls would only complete. On Sat, Apr 5, 2008 at 1:03 AM, Ira [EMAIL PROTECTED] wrote: At 09:42 PM 4/4/2008, you wrote: Common practice is to check every bill. Withing the last month, I have found two several hundred dollar mistakes on Credit Card and Checking account. I am nos sure if companies are charging extra to make up for the economy slow down or they are genuine mistakes, but I have never had these issues in the past besides a mistake here and there over the course of a year.. Good advice, but that wasn't what my message was about. They're a VOIP provider that I thought went out of business months ago or maybe a year ago with $13 of credit on my account. Today they re-appeared and my $13 is still there. Likely that's true for others too. Ira ___ -- 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 ___ -- 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] Paging for analoge devices
Bogen Rulez On 4/5/08, bilal ghayyad [EMAIL PROTECTED] wrote: Hi; Anyone knows (tried) to use Page for analoge phone(zaptel channel - fxs)? If yes, how? Regards Bilal You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com ___ -- 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 ___ -- 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] SellVOIP
Reminds me of my NuFone experience. On 4/5/08, Tom Lynn [EMAIL PROTECTED] wrote: Well, my $21 is still there and all of my calls are being declined. Over a year ago, I requested a refund and regardless of all promises that I would receive one, Jed never followed through. I'd use up the credit if the calls would only complete. On Sat, Apr 5, 2008 at 1:03 AM, Ira [EMAIL PROTECTED] wrote: At 09:42 PM 4/4/2008, you wrote: Common practice is to check every bill. Withing the last month, I have found two several hundred dollar mistakes on Credit Card and Checking account. I am nos sure if companies are charging extra to make up for the economy slow down or they are genuine mistakes, but I have never had these issues in the past besides a mistake here and there over the course of a year.. Good advice, but that wasn't what my message was about. They're a VOIP provider that I thought went out of business months ago or maybe a year ago with $13 of credit on my account. Today they re-appeared and my $13 is still there. Likely that's true for others too. Ira ___ -- 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 ___ -- 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] Paging for analoge devices
Steve Totaro wrote: Bogen Rulez That it does! -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] rxfax crashes Asterisk (segmentation fault)
Hello, Rxfax from agx-ags-addons always crashes for us also. You can download apps we use from: http://193.138.191.205/packets/fax_apps_asterisk14.tgz Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced VoIP Billing From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mark morreny Sent: Friday, April 04, 2008 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] rxfax crashes Asterisk (segmentation fault) Hi, I am using spandsp-0.0.4, tiff-3.8.2, and agx-ags-addon with Asterisk 1.4.18. Everytime rxfax executes, Asterisk crashes: -- Executing [EMAIL PROTECTED]:1] Set(Zap/2-1, FAXFILE=/var/spool/asterisk-fax/1207322398.0.tif) in new stack -- Executing [EMAIL PROTECTED]:2] RxFAX(Zap/2-1, /var/spool/asterisk-fax/1207322398.0.tif) in new st ack [Apr 4 23:20:35] NOTICE[23925]: chan_iax2.c:6025 update_registry: Restricting registration for peer ' iaxmodem' to 60 seconds (requested 50) [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: = = [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Pages transferred: - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Image size: - 1209075756 x -1221451281 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Image resolution- 1209075756 x -1221451281 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Transfer Rate: - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Bad rows- 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Longest bad row run - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Compression typea st_speech_unregister [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Image size (bytes) - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: = = Segmentation fault Is rxfax supposed to be working? What could have caused this problem? Thanks, Mark ___ -- 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
[asterisk-users] Question about Cisco IP phone + Asterisk + channels
Hi all, I'm planning on picking up a Cisco IP phone or 2 and have a question about the multiple lines feature of them, and Asterisk channels in general. Lets say I have 2 Cisco IP phones and a call comes in, each one rings line 1, and I pick up. Is there any way to have notification on the other phone that I'm currently on that channel? If so, then what about if a 2nd call comes in, will it automatically start ringing the 2nd line on the phone? I've never played around with these phones except at my wife's college dorm, which wasn't much. Basically right now I have some ATAs with cordless phones hooked up to them and each ATA has it's own line sort to speak (I'm sure you guys know what I mean), where as one call comes in and whoever answers first wins the channel. If anyone is confused by this and needs clarification, let me know. Thanks in advance! :) ___ -- 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
[asterisk-users] Zaptel data mode not supported?
Hello: Have a TE110P laying around and decided to see if I could build a router around it. I've tried compiling several versions of zaptel .1.4.x with the same results. I checked the zaptel changelog and can't find anything about it no longer being supported (or that it ever was for that matter). ztcfg: Zaptel Configuration SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Network HDLC (Default) (Slaves: 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31) 31 channels configured. Changing signalling on channel 1 from Unused to Network HDLC ZT_CHANCONFIG failed on channel 1: Function not implemented (38) dmesg: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.9.2 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :02:06.0[A] - GSI 16 (level, low) - IRQ 209 FALC version: TE110P: Setting up global serial parameters for E1 FALC V1.2 TE110P: Successfully initialized serial bus for card Found a Wildcard: Digium Wildcard TE110P T1/E1 Zaptel networking not supported by this build. make data: make[1]: Entering directory `/usr/src/zaptel-1.4.9.2/menuselect' make[2]: Entering directory `/usr/src/zaptel-1.4.9.2/menuselect' make[2]: `menuselect' is up to date. make[2]: Leaving directory `/usr/src/zaptel-1.4.9.2/menuselect' make[1]: Leaving directory `/usr/src/zaptel-1.4.9.2/menuselect' make -C datamods datamods make: *** datamods: No such file or directory. Stop. make: *** [data] Error 2 Adding datamods to SUBDIR_MODULES in top level Makefile make: CC [M] /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.o /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `fr_lmi_send': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:450: error: `LMI_CISCO' undeclared (first use in this function) /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:450: error: (Each undeclared identifier is reported only once /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:450: error: for each function it appears in.) /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `fr_set_link_state': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:575: error: structure has no member named `bandwidth' /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `fr_lmi_recv': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:646: error: `LMI_CISCO' undeclared (first use in this function) /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:825: error: structure has no member named `bandwidth' /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:829: error: structure has no member named `bandwidth' /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:845: error: structure has no member named `bandwidth' /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `fr_rx': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:878: error: `LMI_CISCO' undeclared (first use in this function) /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `hdlc_fr_ioctl': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:1209: error: `LMI_CISCO' undeclared (first use in this function) make[4]: *** [/usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.o] Error 1 make[3]: *** [/usr/src/zaptel-1.4.9.2/kernel/datamods] Error 2 make[2]: *** [_module_/usr/src/zaptel-1.4.9.2/kernel] Error 2 Am I missing something, or does zaptel.conf.sample need some updating? Alex ___ -- 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] Advice on best operator phone (with attendant console)
For a receptionist, you generally want to go with a quality phone since they're going to be the heaviest user of the phone system in the building. (Inbound/outbound call agents may take/make more calls, but their requirements are far more simple than the complex call juggling a receptionist can do) Go with the Polycom 601/650 with attendant consoles - the fact that the display is LED and is automatically updated whenever the directory changes is going to be a big plus here, and the sound quality of the Polycoms outperform anything else on the market. The only drawback is that IIRC the directory will have to be configured through an XML file - the web interface will allow you to add entries, however I have a sneaking suspicion that you /can't/ configure BLF through the web interface. Faraz R. Khan wrote: One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets stuck 4. Voice 'gaps' (blackouts) for 4-5 seconds 5. The phone also completely locks up regularly 6. ping response goes from 8ms to 3000ms (after which the phone locks up) Wondering which operator phone would work best. I have the following choices: 1. Linksys SPA 932/962 with attendant console 2. Polycom 601/650 with attendant console I cant confirm online whether the BLF functionality will work with Asterisk 1.2.26. Is somebody using either of these phones in a high volume environment successfully? Thank you. ___ -- 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] Question about Cisco IP phone + Asterisk + channels
The only way someone else will know that another using is using a phone (or zap channel) is thru BLF, which IIRC the Ciscos do NOT support. Setting up the Ciscos with multiple call appearances is quite easy, just give as many lines as you want the same Sip username and password. and it will juggle it automagicly for you. On Sat, Apr 5, 2008 at 7:07 PM, Jon Miron [EMAIL PROTECTED] wrote: Hi all, I'm planning on picking up a Cisco IP phone or 2 and have a question about the multiple lines feature of them, and Asterisk channels in general. Lets say I have 2 Cisco IP phones and a call comes in, each one rings line 1, and I pick up. Is there any way to have notification on the other phone that I'm currently on that channel? If so, then what about if a 2nd call comes in, will it automatically start ringing the 2nd line on the phone? I've never played around with these phones except at my wife's college dorm, which wasn't much. Basically right now I have some ATAs with cordless phones hooked up to them and each ATA has it's own line sort to speak (I'm sure you guys know what I mean), where as one call comes in and whoever answers first wins the channel. If anyone is confused by this and needs clarification, let me know. Thanks in advance! :) ___ -- 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 ___ -- 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] Zaptel data mode not supported?
You need to have the kernel compiled specially for it to work. Thanks, Steve Totaro On 4/5/08, Alex Kauffmann [EMAIL PROTECTED] wrote: Hello: Have a TE110P laying around and decided to see if I could build a router around it. I've tried compiling several versions of zaptel .1.4.x with the same results. I checked the zaptel changelog and can't find anything about it no longer being supported (or that it ever was for that matter). ztcfg: Zaptel Configuration SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Network HDLC (Default) (Slaves: 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31) 31 channels configured. Changing signalling on channel 1 from Unused to Network HDLC ZT_CHANCONFIG failed on channel 1: Function not implemented (38) dmesg: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.9.2 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :02:06.0[A] - GSI 16 (level, low) - IRQ 209 FALC version: TE110P: Setting up global serial parameters for E1 FALC V1.2 TE110P: Successfully initialized serial bus for card Found a Wildcard: Digium Wildcard TE110P T1/E1 Zaptel networking not supported by this build. make data: make[1]: Entering directory `/usr/src/zaptel-1.4.9.2/menuselect' make[2]: Entering directory `/usr/src/zaptel-1.4.9.2/menuselect' make[2]: `menuselect' is up to date. make[2]: Leaving directory `/usr/src/zaptel-1.4.9.2/menuselect' make[1]: Leaving directory `/usr/src/zaptel-1.4.9.2/menuselect' make -C datamods datamods make: *** datamods: No such file or directory. Stop. make: *** [data] Error 2 Adding datamods to SUBDIR_MODULES in top level Makefile make: CC [M] /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.o /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `fr_lmi_send': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:450: error: `LMI_CISCO' undeclared (first use in this function) /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:450: error: (Each undeclared identifier is reported only once /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:450: error: for each function it appears in.) /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `fr_set_link_state': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:575: error: structure has no member named `bandwidth' /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `fr_lmi_recv': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:646: error: `LMI_CISCO' undeclared (first use in this function) /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:825: error: structure has no member named `bandwidth' /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:829: error: structure has no member named `bandwidth' /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:845: error: structure has no member named `bandwidth' /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `fr_rx': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:878: error: `LMI_CISCO' undeclared (first use in this function) /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `hdlc_fr_ioctl': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:1209: error: `LMI_CISCO' undeclared (first use in this function) make[4]: *** [/usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.o] Error 1 make[3]: *** [/usr/src/zaptel-1.4.9.2/kernel/datamods] Error 2 make[2]: *** [_module_/usr/src/zaptel-1.4.9.2/kernel] Error 2 Am I missing something, or does zaptel.conf.sample need some updating? Alex ___ -- 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 ___ -- 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] Advice on best operator phone (with attendant console)
We are using 57i + 560M combination as well... though we are not using the 57i ct... but the idea of giving them a cordless is a good idea. The only downside to the Aastra 57i + 560M is that it can only subscribe to 50 extensions for BLF... i haven;t run into this cap yet myself, but I have heard others talk about it... I think it was a cap introduced in one of the newer versions of firmware... not sure though, and not sure why. I'm running the latest 2.2 firmware on it... the addition of one-touch transfers in the last firmware was very nice so operator can transfer very fast, instead of having to do xfer-BLF key-xfer (for attended transfer), now they can just hit the BLF key for a blind transfer. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Sigma Networks [EMAIL PROTECTED] Sent: Saturday, April 05, 2008 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Advice on best operator phone (with attendant console) We have been marketing ipPBX systems based on asterisk for 3+ years. For the last year we've been placing Aastra 57iCT with 560M sidecars. Our attendants like the idea of a cordless handset so the attendant can go to the copy room, etc. The LCD based sidecar means you can keep it up to date without marking up paper strips. We deploy Thirdlane PBX Manager which allows us to setup the BLF (busy lamp field) via a web interface. Aastra 57iCT: http://neobits.com/aastra_-_a1758-0131-10-05_-_57i_ct_p11471.html Aastra 560m: http://neobits.com/aastra_-_a1760--10-55_-_560m_p11472.html Thirdlane PBX Manager: http://www.thirdlane.com/products/pbxmanager Feel free to contact me off list if I can be of any assistance. Regards, Jim ph: 408-701-9929 Faraz R. Khan wrote: One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets stuck 4. Voice 'gaps' (blackouts) for 4-5 seconds 5. The phone also completely locks up regularly 6. ping response goes from 8ms to 3000ms (after which the phone locks up) Wondering which operator phone would work best. I have the following choices: 1. Linksys SPA 932/962 with attendant console 2. Polycom 601/650 with attendant console I cant confirm online whether the BLF functionality will work with Asterisk 1.2.26. Is somebody using either of these phones in a high volume environment successfully? Thank you. ___ -- 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 ___ -- 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] realtime errors
On Saturday 05 April 2008 12:27:03 ronald ramos wrote: [Apr 6 01:05:04] WARNING[18959]: app_voicemail.c:2262 inboxcount: Failed to obtain database object for 'asterisk'! This error typically means that you failed to configure res_mysql.conf, or that the parameters that you provided are not sufficient to connect. The two most common reasons are: either the socket is incorrect (MySQL's default is /tmp/mysql.sock, but most distributions place the socket at /var/lib/mysql/mysql.sock) or that you've specified a TCP socket, yet MySQL is not listening on TCP (also the default on most distributions). Check your settings in /etc/my.cnf or /etc/mysql/my.cnf or /usr/local/etc/my.cnf and compare with the settings in /etc/asterisk/res_mysql.conf. -- Tilghman ___ -- 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] Advice on best operator phone (with attendant console)
I'd find that very strange considering that the 57i itself has facility for at least 20 BLF buttons and /each/ attendant console has facility for another 60! Matt Watson wrote: We are using 57i + 560M combination as well... though we are not using the 57i ct... but the idea of giving them a cordless is a good idea. The only downside to the Aastra 57i + 560M is that it can only subscribe to 50 extensions for BLF... i haven;t run into this cap yet myself, but I have heard others talk about it... I think it was a cap introduced in one of the newer versions of firmware... not sure though, and not sure why. I'm running the latest 2.2 firmware on it... the addition of one-touch transfers in the last firmware was very nice so operator can transfer very fast, instead of having to do xfer-BLF key-xfer (for attended transfer), now they can just hit the BLF key for a blind transfer. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Sigma Networks [EMAIL PROTECTED] Sent: Saturday, April 05, 2008 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Advice on best operator phone (with attendant console) We have been marketing ipPBX systems based on asterisk for 3+ years. For the last year we've been placing Aastra 57iCT with 560M sidecars. Our attendants like the idea of a cordless handset so the attendant can go to the copy room, etc. The LCD based sidecar means you can keep it up to date without marking up paper strips. We deploy Thirdlane PBX Manager which allows us to setup the BLF (busy lamp field) via a web interface. Aastra 57iCT: http://neobits.com/aastra_-_a1758-0131-10-05_-_57i_ct_p11471.html Aastra 560m: http://neobits.com/aastra_-_a1760--10-55_-_560m_p11472.html Thirdlane PBX Manager: http://www.thirdlane.com/products/pbxmanager Feel free to contact me off list if I can be of any assistance. Regards, Jim ph: 408-701-9929 Faraz R. Khan wrote: One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets stuck 4. Voice 'gaps' (blackouts) for 4-5 seconds 5. The phone also completely locks up regularly 6. ping response goes from 8ms to 3000ms (after which the phone locks up) Wondering which operator phone would work best. I have the following choices: 1. Linksys SPA 932/962 with attendant console 2. Polycom 601/650 with attendant console I cant confirm online whether the BLF functionality will work with Asterisk 1.2.26. Is somebody using either of these phones in a high volume environment successfully? Thank you. ___ -- 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 ___ -- 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 ___ -- 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] Zaptel data mode not supported?
On Sat, Apr 05, 2008 at 10:38:52PM -0400, Steve Totaro wrote: You need to have the kernel compiled specially for it to work. Are you sure? What exactly is needed? I think you need to rebuild the kernel on Centos, but on Debian this happens to be supported in the default kernel. Didn't get to test that support yet, though. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Advice on best operator phone (with attendant console)
Guys thanks a lot. I should be going with a Polycom 650 for all such jobs. If grandstream receives such bad reviews- how are they selling anything? Phones hanging or voice cut-outs are simply unacceptable!! On Sun, 2008-04-06 at 14:12 +1000, Rob Hillis wrote: I'd find that very strange considering that the 57i itself has facility for at least 20 BLF buttons and each attendant console has facility for another 60! Matt Watson wrote: We are using 57i + 560M combination as well... though we are not using the 57i ct... but the idea of giving them a cordless is a good idea. The only downside to the Aastra 57i + 560M is that it can only subscribe to 50 extensions for BLF... i haven;t run into this cap yet myself, but I have heard others talk about it... I think it was a cap introduced in one of the newer versions of firmware... not sure though, and not sure why. I'm running the latest 2.2 firmware on it... the addition of one-touch transfers in the last firmware was very nice so operator can transfer very fast, instead of having to do xfer-BLF key-xfer (for attended transfer), now they can just hit the BLF key for a blind transfer. -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Sigma Networks [EMAIL PROTECTED] Sent: Saturday, April 05, 2008 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Advice on best operator phone (with attendant console) We have been marketing ipPBX systems based on asterisk for 3+ years. For the last year we've been placing Aastra 57iCT with 560M sidecars. Our attendants like the idea of a cordless handset so the attendant can go to the copy room, etc. The LCD based sidecar means you can keep it up to date without marking up paper strips. We deploy Thirdlane PBX Manager which allows us to setup the BLF (busy lamp field) via a web interface. Aastra 57iCT: http://neobits.com/aastra_-_a1758-0131-10-05_-_57i_ct_p11471.html Aastra 560m: http://neobits.com/aastra_-_a1760--10-55_-_560m_p11472.html Thirdlane PBX Manager: http://www.thirdlane.com/products/pbxmanager Feel free to contact me off list if I can be of any assistance. Regards, Jim ph: 408-701-9929 Faraz R. Khan wrote: One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets stuck 4. Voice 'gaps' (blackouts) for 4-5 seconds 5. The phone also completely locks up regularly 6. ping response goes from 8ms to 3000ms (after which the phone locks up) Wondering which operator phone would work best. I have the following choices: 1. Linksys SPA 932/962 with attendant console 2. Polycom 601/650 with attendant console I cant confirm online whether the BLF functionality will work with Asterisk 1.2.26. Is somebody using either of these phones in a high volume environment successfully? Thank you. ___ -- 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 ___ -- 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 ___ -- 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 -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- 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
[asterisk-users] Half-duplex call on TDM2400p with VPMADT032
Hi, I've used all kinds of digium cards without troubles. My last installation is using a TDM2400p with VPMADT032 echo cancel module and after a week of use we noticed that any incoming audio stream gets clipped / dropped when you speak or when ambient noise is high. The call basically feels as in a half-duplex channel, but only to the person behind our asterisk. I found a quick way to recreate by placing a call using zapata channel, someplace that has an audio stream (ie. music on hold from another pbx). When one talks into the phone, one can notice the incoming audio getting muted until you stop talking. First I thought it had to do with polycom configuration although we use the same setup for all installations (VAD, etc), but the same happens with other sip phones and after more tests I can only recreate this using the TDM2400p's FXO trunks. I have an older TDM2400p with no VPMADT032 in production (without this problem), this leads me to believe there maybe something wrong with VPMADT032 module or with my card in particular. Today I rebuilt everything from scratch using latest asterisk 1.2 release, rechecked with the TDM2400p manual zapata configs just to make sure I wasn't missing something. As the manual suggests, I am just using echocancel=yes and this should set 128 default value for the card. In the general zapata options there we have echocancelwhenbridged=yes. I have played with all yes/no combinations without luck. Interrupts and timing stuff are OK, we have good incoming and outgoing audio quality (as long as its not at the same time). Anyone else using this card showing the same problems? Any zaptel/asterisk gurus wanna take a shot at this? Thanks in advance for your feedback/comments. Lex ___ -- 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] Asterisk with lumenvox
Hi Al, how are you? You use Lumenvox? What we think of the performance of the engine? Thank you for your attention Regards! Josué 2008/4/5, Al Baker [EMAIL PROTECTED]: I had posted earlier asking about folks real world experiences with with Lumenvox, and the thread 'strangely' disappeared after some bloke from down under justed sodded himself over my straight simple questions. Hm- makes you wonder. Josué Conti wrote: Hello everyone. I wish I could continue with the approval of the engine Lumenvox, for voice recognition, but not a development of acceptable engine, Please could help me in achieving test? As I said earlier we have a project that will involve a very large number of licenses for Voice recognition, but I would count on help from Lumenvox, for this case. Could you help me? Best Regards Josué 2008/3/19 Josué Conti [EMAIL PROTECTED]: Hello everyone, Rodrigo and Philipp Hello, I would like to know how to properly configure the engine Lumenvox no asterisk, I am trying to dial by vox actually like that the user should dial for receipt of my business, is attended by an IVR system with voice recognition that allows the user to say who would like to talk and the asterisk foward the call. Set up the asterisk below, but the system recognizes the voice, but does not guide the call, running immediately after a hangup, what is wrong with my settings? I can not very material support on the issue, could help me? I am not really achieving great results in my tests with engine Lumenvox: I am trying to test a simple scheduling dialing by voice, where the system identify the user by name and system called in your phone number, but I am not able, could help me? If I did not say any word, the system is static, but if I say any Word, even different words grammar.gram (ura.gram) of the system Performs the following priorities file extensions.conf, please, can You help me? Best Regards Josué Our programming files are configured this way: Ipbx: / etc / asterisk # vim lumenvox.conf ; LumenVox configuration file [General] Servers = 127.0.0.1; Speech Engine Servers to use. Save_sound_files = no; Set to yes to save sound files for use with Speech Tuner [Grammars] ura = / etc / asterisk / grammars / ura.gram [Default] Vad_snr_sensitivity = 50 Vad_volume_sensitivity = 50 Vad_eos_delay = 1250 Vad_wind_back = 750 End_of_speech_timeout = 15000 Use_oov_filter = no ;; ;; Ipbx: / etc / asterisk # vim extensions.conf [General] [Globals] DYNAMIC_FEATURES = # pickupexten hangup atxfer # # blidxfer [Default] Length = 2000.1, Playback (Ura / instit / instit_casa) Length = 1515.1, Playback (Ura / parabens) ;; ;; ; Pilot URA Length = 6969.1, GotoIfTime (07:50-18:05 | mon-fri |*|*? ura, s, 1) Length = 6969.2, GotoIfTime (18:06-23:59 | mon-fri |*|*? ura, s, 1) Length = 6969.3, GotoIfTime (00:00-07:49 | mon-fri |*|*? ura, s, 1) Length = 6969.4, GotoIfTime (* | sat-sun |*|*? ura, s, 1) ; IVR URA ; [URA] ; Length = s, 1, Answer () Length = s, n, Wait (3) Length = s, n, NoOp (entry Ura) Length = s, n, Set (TRIES = 0) ; Length = s, n, ResponseTimeout (10) Length = s, n, BackGround (Ura /abertura) Length = s, n, Playback (beep) ; Length = s, n, BackGround (Ura / abertura1) Length = s, n, Goto (lumenvox-test, s, 1) [Lumenvox-test] Length = s, 1, Answer Length = s, n, Wait (1) Length = s, n, SpeechCreate () Length = s, n, SpeechActivateGrammar (Ura) Length = s, n, SpeechStart () Length = s, n, SpeechBackground (liggol / abertura) Length = s, n, SpeechDeactivateGrammar (Ura) Length = s, n, Goto (institutional, s, 1 - $ SPEECH_TEXT (0) ()) [Institutional] Length = s, 1, Playback (Ura / instit / instit) Length = s, 2, congestion (3) Length = s, 3, hangup ipbx: / etc / asterisk / grammars # vim ura.gram # - Grammar: ura.gram # ABNF 1.0; Language es-CO; Voice mode; Tag-format lumenvox/1.0; Root $ URA; $ Continent = ((Josue | Conti) []): 2000; $ Palms = () 1515 ; $ Ura = ($ conti | $ palms) = $ $ $ (); 2008/3/19, Rodrigo Gonzalez [EMAIL PROTECTED]: Josué Conti escribió: Hello all, how are you? I would like to know from someone uses or has used the engines of LumenVox for integration with the asterisk for voice recognition. Best Regards Josué ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users
Re: [asterisk-users] Ring back when free?
Has anyone here implemented Ring back when free in Asterisk?
Here is what I do; the dialplan enclosed is in AEL2 format, but you can get the
idea.
When a call is originated I save the called and callee numbers in a database.
If the user gets busy he/she hangs up and dial *41. I then retreive the last
number they dialled and place a flag in the database that someone is camping on
it. The H extension checks this flag and if found generates a .call file.
First, I have a macro to save the for each extension who is the last they
called and the last who called them:
// Save the calling and called numbers in To and From and in the database so
// they can be used by *41 and *42. This way the h extension can acecss this
// database for all destinations.
macro Save_From_to ( ) {
// To and From are used in the dialplan, since we might change ${EXTEN}
Set(_To=${MACRO_EXTEN});
Set(_From=${CALLERID(num)});
NoOp(= ${From} - ${To});
// Save them in database for later use.
// Save the caller number at the called extension for *42 usage.
Set(DB(${To}/LastCaller)=${From});
// Where we called for *41
Set(DB(${From}/LastCalled)=${To});
};
This macro is called at the beginning of the normal dialplan.
Now, the *41 which registers the camp-on using the data saved above:
// *41: Camp on the last extension dialled
*41 = {
Set(tmp=${DB(${CALLERID(num)}/LastCalled)});
// Save it so when the other side hangs it will see it and dial us.
Set(DB(${tmp}/CallBack)=${CALLERID(num)});
// Say the number to caller so he can verify...
SayDigits(${tmp});
Hangup();
};
And now the H extension for handling it:
// The Hangup extension which is called when the call is hanged. See whether
// we have some waiting callback waiting on this extension.
h = {
ResetCDR(w);// To make the CDR correct.
NoOp(${From});
// We have to check the two sides of the call: Those who camp on the calling
and
// those who camp on the called.
Set(tmp=${DB(${From}/CallBack)}); // The calling.
if(${tmp} != ) {// Something is there.
DBdel(${From}/CallBack); // And delete it...
// Create the callfile and then move it to the spool directory to make the
call.
System(echo Channel: SIP/${tmp}
/tmp/test.tmp${From})
;
System(echo WaitTime: 20 /tmp/test.tmp${From});
System(echo Extension: ${From}
/tmp/test.tmp${From});
System(echo CallerID: Callback \\\${tmp}\\\
/tmp/te
st.tmp${From});
System(mv /tmp/test.tmp${From}
/var/spool/asterisk/outgo
ing/);
};
Set(tmp=${DB(${To}/CallBack)}); // The called
if(${tmp} != ) {// Something is there
DBdel(${To}/CallBack); // And delete it...
// Create the callfile and then move it to the spool directory to make the
call.
System(echo Channel: SIP/${tmp} /tmp/test.tmp${To});
System(echo WaitTime: 20 /tmp/test.tmp${To});
System(echo Extension: ${To} /tmp/test.tmp${To});
System(echo CallerID: Callback \\\${tmp}\\\
/tmp/te
st.tmp${To});
System(mv /tmp/test.tmp${To}
/var/spool/asterisk/outgoin
g/);
};
};
Good luck! __Yehavi:
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