Re: [Asterisk-Users] screen safe_asterisk does'nt spawn asterisk
Tzafrir Cohen ha scritto: On Thu, Dec 15, 2005 at 01:45:24PM +0100, Simone Cittadini wrote: screen -d -m asterisk -vvvcng works well for me, but I'd prefer to run safe_asterisk in production Any reason you need to run asterisk in a console? asterisk -r allows you to view the current console. First of all, I think my complain about 'screen safe_asterisk' not working was a nonsense, even if I'd get it work it would detach the safe_asterisk script and not asterisk's process Anyway, screen seems the only way to see agi's output (old discussion in the list, and some lines in the wiki), for example : agy.py : [...] def Write(self,data): Write unbuffered line output to STDERR. Ensures data is flushed out. sys.stderr.write(str(data) + \n) sys.stderr.flush() [...] myhagi.py : import agy.py import hgsm.py agiDo = agi.AGI() hGsm = hgsm.HGSM() dst = sys.argv[1] gatDst = hGsm.getGatewayFromDst(dst) agiDo.Write(gatDst: +str(gatDst)) this last line will print on the CLI with 'asterisk -vvvc' nothing is printed with 'safe_asterisk' - 'asterisk -r' so I must 'screen -d -m asterisk -vvvcng' - 'screen -r' ___ --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] TDM01B answering issue
Hi: I saw a hardware in callshops that attached to analoge line and begin counting from the time call is answered to the time it hangup ,So is there ant hardware or a software added to asterisk to solve this answering issue? --- Steve Underwood [EMAIL PROTECTED] wrote: Andrew Kohlsmith wrote: On Saturday 17 December 2005 22:13, Eric ManxPower Wieling wrote: *sigh* Analog Zap FXO ports consider the call answered as soon as it's finished throwing the DTMF at the telco. This is because a Zap port CAN'T tell when an analog call has been answered. Bah, you're absolutely correct. I keep forgetting about POTS; I think PRI when I think Zap. Its not absolutely correct, but its relatively correct. :-) The above is true for most analogue lines around the world. However, there are some places which provide a positive answer indication on analogue lines. The form varies, but it is typically a reversal of line power, or a short timed break in line power. Similarly, while most of the world's analogue lines no longer provide a positive indication of hangup, some still do. Again, this is usually by reversal or a short timed break. Steve ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] PERL AGI DIALSTATUS
Hi all,
I wanted to execute one of mySQL query when the call is answered i
tried with the following code but it dones not seems to work.
$AGI-exec('Dial', $dialext);
my $dialstatus = $AGI-get_variable(DIALSTATUS);
if($dialstatus=ANSWER){$Accounting_update-execute($fdatetime,$Cuniq,$UserName,$CalledN);}
It is not updating my query when the call is successfull answered, and
i checked my query from outside using perl commond it is working well
without any issue.
--
Thank You,
Code Lover
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Re: [Asterisk-Users] Toll Free Providers
hi, how many mins a month do u have ? We can give you @ 4 cents a min if u want retail on virtualphoneline.com On 12/18/05, Tom Vile [EMAIL PROTECTED] wrote: Looking for a good toll free DID provider.Any suggestions?All ready tried Sellvoip and Gafachi and the experience was not desirable. Thanks,Tom Vile___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Rehan Ahmed AllahWala http://www.SuperTec.com - Tommrow's Technology, Today.http://www.didx.net - DID Number Exchange and Peering Service. ___ --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] Can't pickup call when dialing *8 extension
*8 is coded in res_features.so . What are the right extension to dial for pickup calls between sip=sip or zap=sip ... Harry --- Rich Adamson [EMAIL PROTECTED] a écrit : You might have to use *8#. At least I do with my 7960. I added callgroup=1 and pickupgroup=1 for sip channels however I can't pickup a call (see below ) between sip phones when i dial *8 . May I have to add app_pickup to solve this problem. I use asterisk-1.2 Regards Harry serveur1*CLI -- SIP read from 80.119.8.167:5060: ACK sip:[EMAIL PROTECTED]:5050 SIP/2.0 Via: SIP/2.0/UDP 80.119.8.167;branch=z9hG4bKe1bb.87855e92.0 From: alice sip:[EMAIL PROTECTED];tag=AF3B88E-55239161 Call-ID: [EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as543ba455 CSeq: 2 ACK User-Agent: Sip EXpress router(0.9.4 (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' -- Nobody picked up in 1 ms Reliably Transmitting (NAT) to 80.119.8.167:5060: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.119.8.167:5050;branch=z9hG4bK60e70916;rport From: alice sip:[EMAIL PROTECTED]:5050;tag=as7cefba23 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5050 Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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 ---End of Original Message- ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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] cdr mysql problem
Hi All, Thank you all. As you all mentioned it wasnt so serious and was just a simple authentication problem. Its been solved. Regards. From: Diyanat Ali [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] cdr mysql problem Date: Fri, 16 Dec 2005 07:14:05 -0600 MIME-Version: 1.0 X-Originating-IP: [202.65.140.108] X-Originating-Email: [EMAIL PROTECTED] X-Sender: [EMAIL PROTECTED] Received: from lists.digium.com ([69.16.138.164]) by bay0-mc12-f6.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Fri, 16 Dec 2005 05:20:09 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 3F3FE4419;Fri, 16 Dec 2005 06:14:02 -0700 (MST) Received: from psmtp.com (exprod5mx26.postini.com [64.18.0.181])by lists.digium.com (Postfix) with SMTP id CA1AA4415for asterisk-users@lists.digium.com;Fri, 16 Dec 2005 06:13:57 -0700 (MST) Received: from source ([65.54.162.38]) by exprod5mx26.postini.com([64.18.4.10]) with SMTP; Fri, 16 Dec 2005 08:14:06 EST Received: from mail pickup service by hotmail.com with Microsoft SMTPSVC;Fri, 16 Dec 2005 05:14:06 -0800 Received: from 65.54.162.200 by by108fd.bay108.hotmail.msn.com with HTTP;Fri, 16 Dec 2005 13:14:05 GMT X-Message-Info: LGjzam7y+LuXeCVcGsTIL6sfvJRsVs3A23oEsKN3m/A= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com X-OriginalArrivalTime: 16 Dec 2005 13:14:06.0023 (UTC)FILETIME=[978B6570:01C60242] X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] i am using asterisk 1.2.1 with mysql 5 without any issues, please check your configuration again, make sure you have hostname=localhost too and the dbname, user, password are correct [global] hostname=localhost dbname=databasename user=user password=password port=3306 sock=/var/lib/mysql/mysql.sock Diyanat From: Mohammad Shokuie [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cdr mysql problem Return-Path: [EMAIL PROTECTED] Dear folks, I've just compiled asterisk-addon1.2.1 after installing MySQL and MySQl-devel packages. and adjust my cdr_mysql.conf to use the defined database using username and password. But as soon as starting asterisk i get error messages informing me of error, error message is as follows : cdr_addon_mysql.c : Failed to connect mysql database cdr on localhost and res_config_mysql.c : Failed to connect database server on . Im realy lost and dont know whats wrong. I've checked the connection to MySql in command line using the same user and host and its been connected without any problem. Anyone has any idea whats wrong here. Regards. --- M. Shokuie Nia. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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 ___ --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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] Can't pickup call when dialing *8 extension (resent)
*8 is coded in res_features.so . What are the right extension to dial for pickup calls between sip=sip or zap=sip ... Harry --- Rich Adamson radamson at routers.com a écrit : You might have to use *8#. At least I do with my 7960. I added callgroup=1 and pickupgroup=1 for sip channels however I can't pickup a call (see below ) between sip phones when i dial *8 . May I have to add app_pickup to solve this problem. I use asterisk-1.2 Regards Harry serveur1*CLI -- SIP read from 80.119.8.167:5060: ACK sip:*8 at nxs.yi.org:5050 SIP/2.0 Via: SIP/2.0/UDP 80.119.8.167;branch=z9hG4bKe1bb.87855e92.0 From: alice sip:85 at nxs.yi.org;tag=AF3B88E-55239161 Call-ID: b16b7b62-c85b30e0-5fdbcb3b at 192.168.0.20 To: sip:*8 at nxs.yi.org;tag=as543ba455 CSeq: 2 ACK User-Agent: Sip EXpress router(0.9.4 (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- Destroying call 'b16b7b62-c85b30e0-5fdbcb3b at 192.168.0.20' -- Nobody picked up in 1 ms Reliably Transmitting (NAT) to 80.119.8.167:5060: CANCEL sip:86 at 192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 80.119.8.167:5050;branch=z9hG4bK60e70916;rport From: alice sip:84 at 80.119.8.167:5050;tag=as7cefba23 To: sip:86 at 192.168.0.21 Contact: sip:84 at 80.119.8.167:5050 Call-ID: 50b2bf516e9f43a5415036b700b0e075 at 80.119.8.167 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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] asterisk 1.2.1 and mixmonitor problem
Hi there, Any one confronted a crash in asterisk when using mixmonitor app. When i'm using the mixmonitor app on a briged call as soon as the called party hangs up the call asterisk crashes and the process terminates with following error message : Segmentation fault. Ouch .. error while writing audion data :: broken pipe but when the calling party hangs up, everything is smooth. Anyone has any idea on this issue? TIA. M. Shokuie Nia _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] Too high volume on Music on Hold
Hi all. I have an asterisk box on gentoo , and when i try to play MOH, it get too much volume. At a point that it could damage my ear system :) If i normalize the music, decreasing the volume, it normalizes again and play at a volume that i could not use. What could it be wrong?. In other * box with gentoo too, it does not happen. Regards Alberto ___ --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] ztdummy problem !!!
I have the same problem here. It happend after I upgraded my server with Mandriva 2006. What kernel are you using ? Hey, I´m trying to *modprobe ztdummy, *but when i make modprobe, return one error. I use kernel 2.4 and have UHCI USB Controller allowed in my kernel. This problem can be, because i dont have any pci card (fxo) at the computer ? Thanks. ___ --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] ztdummy problem !!!
On Sun, Dec 18, 2005 at 02:42:21PM +0100, Insider KT wrote: I have the same problem here. It happend after I upgraded my server with Mandriva 2006. which has kernel 2.6 . ztdummy there does not depend on USB. What kernel are you using ? Hey, I´m trying to *modprobe ztdummy, *but when i make modprobe, return one error. I use kernel 2.4 and have UHCI USB Controller allowed in my kernel. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Is this possible in Asterisk?
Hi, Let's say an office has 20 people with 20 extensions and they want to enter a code on their phone when they leave for lunch and a voice will tel lthe caller like: The person you are calling is out of the office and will return at 1 pm. Is this something that is possible? Many thanks, Christian ___ --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] ztdummy problem !!!
Gabriel Sartor wrote: Hey, I´m trying to *modprobe ztdummy, *but when i make modprobe, return one error. I use kernel 2.4 and have UHCI USB Controller allowed in my kernel. http://bugs.digium.com/view.php?id=5236 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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is this possible in Asterisk?
Christian wrote: Hi, Let's say an office has 20 people with 20 extensions and they want to enter a code on their phone when they leave for lunch and a voice will tel lthe caller like: The person you are calling is out of the office and will return at 1 pm. Is this something that is possible? Yes -- 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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is this possible in Asterisk?
Hi, Great, do you know where I can find info about this? Many thanks! - Original Message - From: Doug Lytle [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, December 18, 2005 4:02 PM Subject: Re: [Asterisk-Users] Is this possible in Asterisk? Christian wrote: Hi, Let's say an office has 20 people with 20 extensions and they want to enter a code on their phone when they leave for lunch and a voice will tel lthe caller like: The person you are calling is out of the office and will return at 1 pm. Is this something that is possible? Yes -- 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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] ISDN/CAPI outgoing calls - weirdness with ringing
On Fri, 16 Dec 2005, Michael J. Tubby B.Sc (Hons) G8TIC wrote:
All,
I have the following set up:
Fedora Core 4 box (yum updated to current)
Asterisk 1.2.1 + Chan_Capi-cm-0.6.1
AVM C4 card
2 x ISDN2e lines bonded with switchboard number, fax number and 10 x DDI
numbers from British Telecom
14 x Cisco 7960 phones with SIP 7.5
The ISDN lines work in P2P mode and calls are presented with the last 4 digits
only - I land them in a context and branch out from there - everything to do
with incoming calls works just fine!
I have a problem with outgoing calls that are routed over the BT network and
the way in which 'ringing' is presented... depending on the called party
number (hence phone provider) I get different results. For example:
a) if I dial another BT number I get a fraction of a second's ring followed by
silence until the called party answers. The Cisco phone displays:
Proceeding (in 100)
very briefly and is almost immediately over-written by:
Session Progress (in 183)
until the called party answers - at no point is Ringing Destination (in 180)
displayed
b) if I dial an Orange or O2 mobile number I get a second or two's worrth of
silence [while the Orange network locates the mobile] then the mobile rings in
the normal way and the Cisco phone plays out US style ringing. When the number
is dialled the phone displays:
Proceeding (in 100)
when the mobile starts to ring the Cisco phone displays:
Ringng Destination (in 180)
c) if I dial a Bulldog phone number then I get three messages:
Proceeding (in 100) - for a second or so
Session Progress (in 183) - for a couple of seconds
Ringng Destination (in 180) - while the called party's phone rings
d) and the really weird one - if I dial *some* international numbers I get
both UK (BT) ringing tone overlaid with Asterisk/VoIP (US) ringing tone
I have two ways of dialling out:
1. with an explicit 9 for an outside line -- get dialtone from BT and then
dial rest of the digits - like a legacy PBX
2. dialing just based on the fact that the extension starts with a zero so its
an outside call via BT
I have tried all combinations of early B3 connect 'always', 'on success' and
'never' and it doesn't appear to change things... the relevant part of
extensions.conf is below for completness.
Before I dive in to the next level down:
- is this a known issue?
- is there a solutiuon/workaround/patch/fix
- do I need to get down and dirty with CAPI and SIP debug?
Have you tried CAPI-Dial option 'o' ? Together with 'b' it should give
you progress in any case.
Armin
Mike
;
; external-routes: this is where we get to dial out
;
[external-routes]
;
; outgoing via main ISDN line using explicit 9 for an outside line
; and ISDN eqarly B3 connect (overlap sending) to drop us to the
; BT provided dialtone and work like a normal/legacy phone system -
; we force the caller ID to our exchange number so that DDI's dont
; leak out
;
exten = 9,1,NoOp(ISDN: Pickup outside line (early B3 connect) for:
${CALLERIDNUM})
exten = 9,2,SetCallerId(${THORCOM_MAIN})
exten = 9,3,Dial(CAPI/g1//b)
exten = 9,4,Hangup
;
; implicit trunked call - here we could/should do an ENUM look
; up to see if we can place the call via IP and fall back to BT
; if not... just for now this isn't implemented and we always call
; out via BT!!
;
exten = _0.,1,Dial(CAPI/g1/${EXTEN}/b); early B3 connect
always
;exten = _0.,1,Dial(CAPI/g1/${EXTEN}/B) ; early B3 connect
on success
;exten = _0.,1,Dial(CAPI/g1/${EXTEN}) ; no special
options
exten = _0.,2,Hangup
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[Asterisk-Users] Asterisk - Avaya system
Just the other day I tried connecting an Avaya IP403 Office IP PBX to my asterisk. The IP403 is currently used for all the phones at our office and it is connected via it's own PRI to the PSTN. Now I have a Asterisk machine with three PRIs used for our SIP services. To be able to utilize our capacity better I would like to let the Avaya connect to the Asterisk and share the three PRIs that it has. So, I connected the Avaya to my Asterisk, configured the Sangoma card to act as the CO side. PRI came up and I'm all happy. I try: dial [EMAIL PROTECTED] and voila it dials the correct extension on the Avaya. I'm even happier! :) Now I try to dial out, after punching the four first digits the Avaya dials out. The asterisk in turn dials out to the PSTN. No matter which number I try it just dials out after the fourth digit. If I punch something shorter, like a three digit number it waits for a while and then dials. Is this some feature to let the CO know of which area code the calls is going ahead of time? Anyone with Avaya knowledge know how to turn this off? Is there some way to circumvent this using hacks on the asterisk side? Thanks Regards Kristian ___ --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] astcc issue
You should be able to edit prices from within the routes page. However, you can't set different prices on different brands more accurately than by using markup. That is one of the reasons that I've branched / mostly rewritten the product. ASTPP, www.astpp.org, does provide support for doing this with prices but the calling card stuff is only in cvs yet. Darren Wiebe [EMAIL PROTECTED] jonny hashem wrote: Hi list: I need to create a routes list to specific card number wih different prices than the initial routes list ,because markup donot achieve my purpose and markup use for changing prices for all routes,and i need to change prices for specific routes. So is there any possible way to do that? Regards; jonny __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] Is this possible in Asterisk?
Let's say an office has 20 people with 20 extensions and they want to enter a code on their phone when they leave for lunch and a voice will tel lthe caller like: The person you are calling is out of the office and will return at 1 pm. Is this something that is possible? I'm tot shure if there is any documentation regarding this specific topic. For Realisation I would suggest three parts: - Define an Pseudo-Number to be dialed on going to / coming back from lunch - The dialplan for this numbers should be modifiyng the state and playing an appropriate message. - The general dialplan has to read the current stat for the dialled target and act corresponding to this. - To Store the state there are DB-like functions in asterisk - or you can write an AGI. Elmar ___ --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] Shutting down Asterisk when not in RTP Stream
Hi Tyler. We're registering users with OpenSER, which also routes the calls to a series of Asterisk systems. The really tricky part is allowing different phones entering through different Asterisk systems to reach other. Currently, the solution is to, upon registration from phones, issue a forward() command in OpenSER to forward the registration to every Asterisk system. In this way, every Asterisk box knows about every phone and it doesn't matter which Asterisk system takes the call. It's not a perfect solution though. When OpenSER sends the forward() request to Asterisk, it also sends back the 'Trying' and 'Ok' messages to the phones (We're using Polycom's). The phones don't seem to have a problem with these extraneous messages so far. A better solution would have been to use t_replicate() in OpenSER, which absorbs these messages, but you can only call t_replicate once. We may still end up sending all calls BACK through OpenSER again to terminate the call, as it knows the location of all the phones as well. This is easy from a simple dial plan perspective, but I'm not sure yet how some of the more advanced Asterisk features such as hints and ACD Queues will work when specifying @proxy for their location. I'd prefer to leave OpenSER out of the equation though. Just trying to get it to do failure_route() etc to Asterisk is a huge pain considering the docs on it are s bad. Oh yeah check out the use of failure_route with t_relay() when sending calls to Asterisk in a redundant fashion. It seems to be working well so far. Failover is very fast. I also saw a post on the OpenSER list last night saying that the dispatcher (which we had looked at before) now supports failure_route too. We liked it initially because it can load balance on call-id and give you a roughly even call distribution. Don't try using realtime either it's hard to believe but you can't use it for sharing a common contact database between Asterisk systems. Digium have admitted to this. Doug. -Original Message- From: Tyler [mailto:[EMAIL PROTECTED] Sent: Fri 12/16/2005 2:13 PM To: Douglas Garstang Cc: Subject: [Asterisk-Users] Shutting down Asterisk when not in RTP Stream Doug, I've been reading a lot of your posts on the Asterisk list and the OpenSER list. You seem to be in the same situation I am in. I need to get a highly-availably and scalable solution up and running. I know Asterisk very well and am learning OpenSER now. What sort of high availability solution do you have running right now with OpenSER and asterisk? Do your users register to OpenSER or are you forwarding registrations? Just thought I'd throw you a couple questions as you seem to be fighting in the trenches right now and may be able to offer me a few do it this way tips to save me some time ;-) Thanks again, tf. -- Tyler [EMAIL PROTECTED] ___ --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] Is this possible in Asterisk?
On Sunday 18 December 2005 10:09, Christian wrote: Great, do you know where I can find info about this? Many thanks! There is nothing canned that does this. You need to break the problem down into sections and implement each section. Elmar's already broken it down for you. If you have any specific questions on the implementation, feel free to ask. Otherwise... try some stuff out, experiment and LEARN the system... This is exactly what makes Asterisk so powerful! -A. ___ --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] Is this possible in Asterisk?
Hi Elmar and all others, Will have a look and if I can't get it working I will post here! many thanks! - Original Message - From: Elmar Haneke [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, December 18, 2005 5:17 PM Subject: Re: [Asterisk-Users] Is this possible in Asterisk? Let's say an office has 20 people with 20 extensions and they want to enter a code on their phone when they leave for lunch and a voice will tel lthe caller like: The person you are calling is out of the office and will return at 1 pm. Is this something that is possible? I'm tot shure if there is any documentation regarding this specific topic. For Realisation I would suggest three parts: - Define an Pseudo-Number to be dialed on going to / coming back from lunch - The dialplan for this numbers should be modifiyng the state and playing an appropriate message. - The general dialplan has to read the current stat for the dialled target and act corresponding to this. - To Store the state there are DB-like functions in asterisk - or you can write an AGI. Elmar ___ --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 ___ --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] ACD with polycom ip phones
Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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] SIP Watchdog
Is there anything I can set or any scripts you guys have where if it sees certain connections (my upstreams) are down, it attempts to reconnect them say every minute or 5 minutes? If a provider reloads something, the connection some times drops and I have to do a "sip reload" to get it to come back. Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --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] SIP Watchdog
gate:/etc/asterisk/.sys# cat astdog.sh #!/bin/sh # # sleep 60 # while [ 1 ] ; do BEZI=`ps auxx|egrep 'asterisk -p'|egrep -v 'grep'|wc -l`; if [ $BEZI = 0 ]; then `killall -9 mpg123`; `asterisk -p`; fi sleep 10 done gate:/etc/asterisk/.sys# --- turby Is there anything I can set or any scripts you guys have where if it sees certain connections (my upstreams) are down, it attempts to reconnect them say every minute or 5 minutes? If a provider reloads something, the connection some times drops and I have to do a sip reload to get it to come back. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- S pozdravem, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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] Is it me, or is 1.2.1 slower than 1.0.9?
Hi all, I just wiped my system and did a clean Asterisk 1.2.0 install with Bristuff 0.3 Pre 1c. (It doesn't work with 1.2.1 yet!) :-( Is it my server or is 1.2.0 considerably slower than 1.0.9 was? It seems to me that all actions take noticably longer than before! Also, despite setting DYNAMIC_FEATURES=automon in the extensions.conf globals section and uncommenting automon=*1 in features.conf, nothing happens when pressing *1 When I change blinsxfer in features.conf to anything different than #, it no longer works. It only works with my softphones anyway, as my ZAP connected ISDN phone never transfers to begin with! I'm getting depressed, because I know all these nice features are there, and I cannot get any of them working! (Once it works, I can deploy it at 2 other locations and really start saving money...) Any suggestions? TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1c - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] New voicemail alert options for Cisco 7960 SIPphones
I just converted 5 7960's to the latest SIP firmware, used the Cisco example configuration files, and nothing custom within Asterisk and my message lights work fine. Kerry Garrison Publisher - GeekGazette.com - VOIPSpek.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Sunday, December 18, 2005 9:25 AM To: Asterisk-Users Subject: [Asterisk-Users] New voicemail alert options for Cisco 7960 SIPphones I'm looking for ideas on how to implement voicemail notification on Cisco 7960 SIP phones. Something like a light on the legacy pbx-phone would be perfect. Even maybe go so far as a quick ring to the extension every 15 minutes or so, but then that would increment the on-screen missed call count. How about a debug-test call where we telnet into the users phones, open a test call on speakerphone back to some extension which simply plays a soundfile like You've got mail. Any suggestions? Is there any simple way to check the voicemail application to see which mailboxes have new messages waiting? Is there simple way to notify users on phones like the Ciscos ? Thanks Tim ___ --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 ___ --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] ztdummy problem !!!
You have to make sure that the uhciusb driver is not compiled in the kernel but just loaded as a module, and during boot you could load it using modprobe before you modprobe ztdummy. On 12/18/05, Doug Lytle [EMAIL PROTECTED] wrote: Gabriel Sartor wrote: Hey, I´m trying to *modprobe ztdummy, *but when i make modprobe, return one error. I use kernel 2.4 and have UHCI USB Controller allowed in my kernel. http://bugs.digium.com/view.php?id=5236 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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Is this possible in Asterisk?
Hi There, I can suggest you to check the dial status variable in dial plan and if its NO_ANSWER guide the caller to voicemail with 'u' option, and if they leave and get back on a fixed time you can take a look for day time night time topic in asterisk documents. HTH, -- M. Shokuie Nia. From: Christian [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Is this possible in Asterisk? Date: Sun, 18 Dec 2005 18:36:14 +0100 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc12-f19.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Sun, 18 Dec 2005 09:48:30 -0800 Received: from arizona.digium.com (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 7E14C4337;Sun, 18 Dec 2005 10:38:02 -0700 (MST) Received: from psmtp.com (unknown [64.18.0.41])by lists.digium.com (Postfix) with SMTP id 85E3E4334for asterisk-users@lists.digium.com;Sun, 18 Dec 2005 10:36:39 -0700 (MST) Received: from source ([64.233.162.192]) by exprod5mx127.postini.com([64.18.4.10]) with SMTP; Sun, 18 Dec 2005 09:36:21 PST Received: by zproxy.gmail.com with SMTP id o1so1103021nzffor asterisk-users@lists.digium.com;Sun, 18 Dec 2005 09:36:21 -0800 (PST) Received: by 10.64.3.11 with SMTP id 11mr451159qbc;Sun, 18 Dec 2005 09:36:20 -0800 (PST) Received: from uwv ( [82.182.183.11])by mx.gmail.com with ESMTP id a29sm1762599qbd.2005.12.18.09.36.19;Sun, 18 Dec 2005 09:36:20 -0800 (PST) X-Message-Info: LGjzam7y+LsP0A1T+4Y1PvJh+HFLKLA3pfw57wIWqLo= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com;h=received:message-id:from:to:references:subject:date:mime-version:content-type:content-transfer-encoding:x-priority:x-msmail-priority:x-mailer:x-mimeole;b=od0eK3XRc1NMxk1DtuLWp8refLZSJl1MqRQ6J8VP/+5XgiSxwZm+ucWQwOiCoNRGoKXQAwYGyCyF5pe+iUfcdGF+0CfCHFEDIBXltT9dL3HZ7b2Os8FCSiQZxEmzKUhydM4WWMpz4qoHrRBXX2m6fXooyczHXAXr8dDlwFL9XwY= References: [EMAIL PROTECTED] [EMAIL PROTECTED] X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2900.2670 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2900.2670 X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 18 Dec 2005 17:48:30.0140 (UTC) FILETIME=[41BFC7C0:01C603FB] Hi Elmar and all others, Will have a look and if I can't get it working I will post here! many thanks! - Original Message - From: Elmar Haneke [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, December 18, 2005 5:17 PM Subject: Re: [Asterisk-Users] Is this possible in Asterisk? Let's say an office has 20 people with 20 extensions and they want to enter a code on their phone when they leave for lunch and a voice will tel lthe caller like: The person you are calling is out of the office and will return at 1 pm. Is this something that is possible? I'm tot shure if there is any documentation regarding this specific topic. For Realisation I would suggest three parts: - Define an Pseudo-Number to be dialed on going to / coming back from lunch - The dialplan for this numbers should be modifiyng the state and playing an appropriate message. - The general dialplan has to read the current stat for the dialled target and act corresponding to this. - To Store the state there are DB-like functions in asterisk - or you can write an AGI. Elmar ___ --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 ___ --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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/
Re: [Asterisk-Users] SIP and echo cancel
Dear pals, As a matter of fact im serious to know where is the source of echo in a pure VoIP connection, i think the most of echo problems come from hybrid circuits which are not an issue in pure VoIP sessions. Regards. --- M. Shokuie Nia. From: Luki [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] SIP and echo cancel Date: Sat, 17 Dec 2005 21:45:57 -0800 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc11-f10.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Sat, 17 Dec 2005 21:46:37 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id C2F0D2FC360;Sat, 17 Dec 2005 22:46:04 -0700 (MST) Received: from psmtp.com (exprod5mx123.postini.com [64.18.0.37])by lists.digium.com (Postfix) with SMTP id 891FC2FC355for asterisk-users@lists.digium.com;Sat, 17 Dec 2005 22:46:01 -0700 (MST) Received: from source ([64.233.184.204]) by exprod5mx123.postini.com([64.18.4.10]) with SMTP; Sat, 17 Dec 2005 23:46:00 CST Received: by wproxy.gmail.com with SMTP id i13so363732wrafor asterisk-users@lists.digium.com;Sat, 17 Dec 2005 21:45:57 -0800 (PST) Received: by 10.54.65.16 with SMTP id n16mr103252wra;Sat, 17 Dec 2005 21:45:57 -0800 (PST) Received: by 10.54.119.12 with HTTP; Sat, 17 Dec 2005 21:45:57 -0800 (PST) X-Message-Info: LGjzam7y+LspoNBQ/UDNaVvtG42BIjDD5YEU0+10Zno= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com;h=received:message-id:date:from:to:subject:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references;b=gqvH8XlVAQ3n1XKvRfERCjsu4Sw5l6Hs8ypOEpn/YT7wgM+Cu89/2rsLdnLtryXX6cFOSZewO4OJDWzUJ+TPn4iHLCivfP7HgWUtZndd45RaQyh1waRM3xz7TeC8eu4C76YDK1mY1V4lcH7UFNE0KO/a6mS/JqXL+GFHEYPJSBM= References: [EMAIL PROTECTED] X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 18 Dec 2005 05:46:37.0982 (UTC) FILETIME=[69B0F3E0:01C60396] Before I start hacking this into asterisk 1.2.1 I would like to known if others are running into this kind of problem ? Asterisk doesn't do any echo cancellation in the setup you describe; it just passes the audio data, and transcodes if necessary. The endpoints (the 841 phone and the 2002 and 3000 ATAs) are responsible for cancelling echo. The Sipura ATA's generally do a good job cancelling echo. You may want to play with the gain settings in the admin web config for the Sipura ATA. As far as the 841 is concerned, if the handset volume is too loud I noticed you may be getting acoustic echo. Hasn't been a problem for me for PSTN calls or SIP to SIP calls though. If you really want to patch asterisk to apply echo cancellation on the RTP stream on pure VoIP calls, that would be interesting to see how well it works. --Luki ___ --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 _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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] SIP and echo cancel
Dear pals, As a matter of fact im serious to know where is the source of echo in a pure VoIP connection, i think the most of echo problems come from hybrid circuits which are not an issue in pure VoIP sessions. Regards. --- M. Shokuie From: Luki [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] SIP and echo cancel Date: Sat, 17 Dec 2005 21:45:57 -0800 MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc11-f10.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Sat, 17 Dec 2005 21:46:37 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id C2F0D2FC360;Sat, 17 Dec 2005 22:46:04 -0700 (MST) Received: from psmtp.com (exprod5mx123.postini.com [64.18.0.37])by lists.digium.com (Postfix) with SMTP id 891FC2FC355for asterisk-users@lists.digium.com;Sat, 17 Dec 2005 22:46:01 -0700 (MST) Received: from source ([64.233.184.204]) by exprod5mx123.postini.com([64.18.4.10]) with SMTP; Sat, 17 Dec 2005 23:46:00 CST Received: by wproxy.gmail.com with SMTP id i13so363732wrafor asterisk-users@lists.digium.com;Sat, 17 Dec 2005 21:45:57 -0800 (PST) Received: by 10.54.65.16 with SMTP id n16mr103252wra;Sat, 17 Dec 2005 21:45:57 -0800 (PST) Received: by 10.54.119.12 with HTTP; Sat, 17 Dec 2005 21:45:57 -0800 (PST) X-Message-Info: LGjzam7y+LspoNBQ/UDNaVvtG42BIjDD5YEU0+10Zno= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com;h=received:message-id:date:from:to:subject:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references;b=gqvH8XlVAQ3n1XKvRfERCjsu4Sw5l6Hs8ypOEpn/YT7wgM+Cu89/2rsLdnLtryXX6cFOSZewO4OJDWzUJ+TPn4iHLCivfP7HgWUtZndd45RaQyh1waRM3xz7TeC8eu4C76YDK1mY1V4lcH7UFNE0KO/a6mS/JqXL+GFHEYPJSBM= References: [EMAIL PROTECTED] X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 18 Dec 2005 05:46:37.0982 (UTC) FILETIME=[69B0F3E0:01C60396] Before I start hacking this into asterisk 1.2.1 I would like to known if others are running into this kind of problem ? Asterisk doesn't do any echo cancellation in the setup you describe; it just passes the audio data, and transcodes if necessary. The endpoints (the 841 phone and the 2002 and 3000 ATAs) are responsible for cancelling echo. The Sipura ATA's generally do a good job cancelling echo. You may want to play with the gain settings in the admin web config for the Sipura ATA. As far as the 841 is concerned, if the handset volume is too loud I noticed you may be getting acoustic echo. Hasn't been a problem for me for PSTN calls or SIP to SIP calls though. If you really want to patch asterisk to apply echo cancellation on the RTP stream on pure VoIP calls, that would be interesting to see how well it works. --Luki ___ --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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] Is this possible in Asterisk?
On Sunday 18 December 2005 14:15, Mohammad Shokuie wrote: I can suggest you to check the dial status variable in dial plan and if its NO_ANSWER guide the caller to voicemail with 'u' option, and if they leave and get back on a fixed time you can take a look for day time night time topic in asterisk documents. That's not what he's asking for, from my interpretation. As I understand it he wants it to ring a lot fewer times than normal if a call comes in after the extension 'holder' has entered in the magic keycode to enable lunch mode. But yes, if you just used the unavailable message with some magic that GetDB'd before to determine the # of rings and SetDB'd on the magic keypress, you'd have something very close to what he wants, I think. -A. ___ --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] TDM01B answering issue
Hi there, As a matter of fact its an awfull issue specially when you are using auto announcement systems. As far as i know its possible to solve this problem on analog boards with tone detection and VAD algorithems but dont think there is anything out there you can use with asterisk and TDM boards, Regards. --- M. Shokuie Nia From: chawki hammoud [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] TDM01B answering issue Date: Sun, 18 Dec 2005 00:19:06 -0800 (PST) MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc12-f3.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Sun, 18 Dec 2005 00:19:39 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id D01952FC481;Sun, 18 Dec 2005 01:19:08 -0700 (MST) Received: from psmtp.com (exprod5mx155.postini.com [64.18.0.224])by lists.digium.com (Postfix) with SMTP id 0C6572FC47Afor asterisk-users@lists.digium.com;Sun, 18 Dec 2005 01:19:06 -0700 (MST) Received: from source ([66.218.94.75]) by exprod5mx155.postini.com([64.18.4.10]) with SMTP; Sun, 18 Dec 2005 02:19:07 CST Received: (qmail 19126 invoked by uid 60001); 18 Dec 2005 08:19:06 - Received: from [66.198.34.52] by web90104.mail.scd.yahoo.com via HTTP;Sun, 18 Dec 2005 00:19:06 PST X-Message-Info: LGjzam7y+LuUSNZFNK+DqnOnhvKURAK0CzR62Nl8W34= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=s1024; d=yahoo.com;h=Message-ID:Received:Date:From:Subject:To:In-Reply-To:MIME-Version:Content-Type:Content-Transfer-Encoding;b=cR/AIO9hQQQijoKTqE2va8Ar4Gk9QGe7ZREcQP0Qzl1Hs4bFEQrAnc4h1x6okO9M2QG8er1/jH4+e328SmtMAqIvUE3czGC8GPiKEeNkdSA8PTqWCpbyoQ7sHh9pLlTNGla1AL4wNjnA6dshViBmvDH53K/6jSpDa+oHqg0ryeE=; X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 18 Dec 2005 08:19:40.0140 (UTC) FILETIME=[CAAF12C0:01C603AB] Hi: I saw a hardware in callshops that attached to analoge line and begin counting from the time call is answered to the time it hangup ,So is there ant hardware or a software added to asterisk to solve this answering issue? --- Steve Underwood [EMAIL PROTECTED] wrote: Andrew Kohlsmith wrote: On Saturday 17 December 2005 22:13, Eric ManxPower Wieling wrote: *sigh* Analog Zap FXO ports consider the call answered as soon as it's finished throwing the DTMF at the telco. This is because a Zap port CAN'T tell when an analog call has been answered. Bah, you're absolutely correct. I keep forgetting about POTS; I think PRI when I think Zap. Its not absolutely correct, but its relatively correct. :-) The above is true for most analogue lines around the world. However, there are some places which provide a positive answer indication on analogue lines. The form varies, but it is typically a reversal of line power, or a short timed break in line power. Similarly, while most of the world's analogue lines no longer provide a positive indication of hangup, some still do. Again, this is usually by reversal or a short timed break. Steve ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or
Re: [Asterisk-Users] Too high volume on Music on Hold
Ran into this myself. Portage has a depend for mpg123 but it installs the one that's also in portage. The one in portage is broken (read: it doesn't work well with Asterisk). I avoid portage when it comes to anything asterisk related. I build from source. The asterisk source has mpg123 included in the distribution, and that version works fine. Not using the mpg123 included in the asterisk source will lead to various issues such as the one you are describing. On 18-Dec-05, at 8:25 AM, Alberto Sagredo wrote: Hi all. I have an asterisk box on gentoo , and when i try to play MOH, it get too much volume. At a point that it could damage my ear system :) If i normalize the music, decreasing the volume, it normalizes again and play at a volume that i could not use. What could it be wrong?. In other * box with gentoo too, it does not happen. Regards Alberto ___ --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 ___ --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] SIP and echo cancel
On Sunday 18 December 2005 14:32, Mohammad Shokuie wrote: As a matter of fact im serious to know where is the source of echo in a pure VoIP connection, i think the most of echo problems come from hybrid circuits which are not an issue in pure VoIP sessions. Easy. Get better endpoints. In a pure-voip loop you have echo due to acoustic coupling from the earpiece to the mic, or the speaker to the mic in a speakerphone. Easy way to tell: in a call with bad echo, have the other side mute. If your echo goes away, you've got your culprit. Also note that if your transmit level is too high or they have the volume up too loud on their end it could push the audio coupling over what the design specifications were. -A. ___ --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] TDM01B answering issue
Hi: So i think there is no possible way to terminate minutes using analoge lines, Is that true? --- Mohammad Shokuie [EMAIL PROTECTED] wrote: Hi there, As a matter of fact its an awfull issue specially when you are using auto announcement systems. As far as i know its possible to solve this problem on analog boards with tone detection and VAD algorithems but dont think there is anything out there you can use with asterisk and TDM boards, Regards. --- M. Shokuie Nia From: chawki hammoud [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] TDM01B answering issue Date: Sun, 18 Dec 2005 00:19:06 -0800 (PST) MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc12-f3.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Sun, 18 Dec 2005 00:19:39 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id D01952FC481;Sun, 18 Dec 2005 01:19:08 -0700 (MST) Received: from psmtp.com (exprod5mx155.postini.com [64.18.0.224])by lists.digium.com (Postfix) with SMTP id 0C6572FC47Afor asterisk-users@lists.digium.com;Sun, 18 Dec 2005 01:19:06 -0700 (MST) Received: from source ([66.218.94.75]) by exprod5mx155.postini.com([64.18.4.10]) with SMTP; Sun, 18 Dec 2005 02:19:07 CST Received: (qmail 19126 invoked by uid 60001); 18 Dec 2005 08:19:06 - Received: from [66.198.34.52] by web90104.mail.scd.yahoo.com via HTTP;Sun, 18 Dec 2005 00:19:06 PST X-Message-Info: LGjzam7y+LuUSNZFNK+DqnOnhvKURAK0CzR62Nl8W34= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=s1024; d=yahoo.com;h=Message-ID:Received:Date:From:Subject:To:In-Reply-To:MIME-Version:Content-Type:Content-Transfer-Encoding;b=cR/AIO9hQQQijoKTqE2va8Ar4Gk9QGe7ZREcQP0Qzl1Hs4bFEQrAnc4h1x6okO9M2QG8er1/jH4+e328SmtMAqIvUE3czGC8GPiKEeNkdSA8PTqWCpbyoQ7sHh9pLlTNGla1AL4wNjnA6dshViBmvDH53K/6jSpDa+oHqg0ryeE=; X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 18 Dec 2005 08:19:40.0140 (UTC) FILETIME=[CAAF12C0:01C603AB] Hi: I saw a hardware in callshops that attached to analoge line and begin counting from the time call is answered to the time it hangup ,So is there ant hardware or a software added to asterisk to solve this answering issue? --- Steve Underwood [EMAIL PROTECTED] wrote: Andrew Kohlsmith wrote: On Saturday 17 December 2005 22:13, Eric ManxPower Wieling wrote: *sigh* Analog Zap FXO ports consider the call answered as soon as it's finished throwing the DTMF at the telco. This is because a Zap port CAN'T tell when an analog call has been answered. Bah, you're absolutely correct. I keep forgetting about POTS; I think PRI when I think Zap. Its not absolutely correct, but its relatively correct. :-) The above is true for most analogue lines around the world. However, there are some places which provide a positive answer indication on analogue lines. The form varies, but it is typically a reversal of line power, or a short timed break in line power. Similarly, while most of the world's analogue lines no longer provide a positive indication of hangup, some still do. Again, this is usually by reversal or a short timed break. Steve ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Sangoma E1 board Experience
Its not a limitation. Its an architectural design which is based on pulse code modulation (pcm) standards, which essentially says: - 8,000 audio samples per second, - each sample is an 8-bit value - resulting in 64,000 bits/second (like g711 codec standard) Thank you for your answer, but I don't think you read/understood my question? The Zaptel driver uses a 8 byte buffer for buffering G.711 in each direction. This has nothing to do with the sample rate, but how large packets you can buffer the voice for to be transported through the PCI interface. I get the impression that the Digium boards are limited to 8 bytes (1 ms ) by reading the zaptel driver. But, I don't know this cause I have no hardware manual that cover the PCI interface. I was hoping that the Sangoma boards offered a more variable buffer size??? jvb ___ --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] FOP led Colors
Hello guy´s
I´m trying to create a extension do modify the led colors of
a button on a FOP,
Via manager command, liked PHP, but I do not have a good
result, I have set de astdb family in op_astdb.conf, but never,
My Php script, and my extension.conf is bellow
Thanks for all
By
//php script
fputs($socket,
Action: Originate\r\n);
fputs($socket, Channel: Local/[EMAIL PROTECTED]/n\r\n);
fputs($socket,
Context: features \r\n);
fputs($socket, Exten:
*79\r\n);
fputs($socket, Priority:
1\r\n);
fputs($socket, Callerid:
$_SESSION[type]/$_SESSION[extension]\r\n); type like SIP or IAX
fputs($socket,
Timeout: 3\r\n\r\n);
//extensions.conf
[features]
exten = *78,1,UserEvent(ASTDB|Channel:
${CALLERID}^Family: dnd^State: On)
exten = *78,2,SetVar(temp=${CALLERID})
exten = *78,3,Cut(temp=temp,,1)
exten = *78,4,DBPut(dnd/${temp}=On)
exten = *78,5,Hangup
exten = 123,1,answer
exten = 123,n,hangup
Alex Montoanelli
___
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Re: [Asterisk-Users] Is it me, or is 1.2.1 slower than 1.0.9?
On Sun, December 18, 2005 20:05, Francesco Peeters (Asterisk) said: Hi all, I just wiped my system and did a clean Asterisk 1.2.0 install with Bristuff 0.3 Pre 1c. (It doesn't work with 1.2.1 yet!) :-( Is it my server or is 1.2.0 considerably slower than 1.0.9 was? It seems to me that all actions take noticably longer than before! Also, despite setting DYNAMIC_FEATURES=automon in the extensions.conf globals section and uncommenting automon=*1 in features.conf, nothing happens when pressing *1 Solved that... When I change blinsxfer in features.conf to anything different than #, it no longer works. That too... It only works with my softphones anyway, as my ZAP connected ISDN phone never transfers to begin with! And here we come to the root cause: The Siemens ISDN DECT station stubbornly refuses to do DTMF unless I manually go in to a menu 2 levels deep to temporarily turn it on... No preference setting, etc. It even gets worse with non Siemens DECT handsets (using the GAP protocol), as these do not even support the keypad switching, which means I first have to do a DECT transfer to a Siemens handset or the base, before I can xfer to a non-DECT extension or external peer... I'm getting depressed, because I know all these nice features are there, and I cannot get any of them working! (Once it works, I can deploy it at 2 other locations and really start saving money...) So my depression is somewhat lifted, as I will be proceeding with the other 2 locations, as I now have all features working, however I need to figure out how to properly work around the Siemens issue, as the other sites too use Siemens ISDN hardware! :rolls eyes: Time to bring out the AMD 1000 box and start prepping that one! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] iaxmodem through zaphfc
Hi Guys, I'm trying to send and receive some faxes using iaxmodem and hylafax through an hfc isdn board and a bristuffed asterisk. All seems to work fine, but the faxes are sent randomly truncated without any reported error. Any idea or suggestion ? Anybody owns a working hylafax/iaxmodem/zaphfc configuration ? Thanks in advance. ___ --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] Sangoma E1 board Experience
Its not a limitation. Its an architectural design which is based on pulse code modulation (pcm) standards, which essentially says: - 8,000 audio samples per second, - each sample is an 8-bit value - resulting in 64,000 bits/second (like g711 codec standard) Thank you for your answer, but I don't think you read/understood my question? The Zaptel driver uses a 8 byte buffer for buffering G.711 in each direction. This has nothing to do with the sample rate, but how large packets you can buffer the voice for to be transported through the PCI interface. I get the impression that the Digium boards are limited to 8 bytes (1 ms ) by reading the zaptel driver. But, I don't know this cause I have no hardware manual that cover the PCI interface. I was hoping that the Sangoma boards offered a more variable buffer size??? Spec sheets are available for the TigerJet 320 pci chipset as well as the Silcon Labs 3050, 3210 chip sets used on the TDM card. If you dig through those I think you'll find that it would difficult if not impossible to change the card's infrastructure since its based on the standards noted above. ___ --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] Sangoma E1 board Experience
On Sunday 18 December 2005 18:01, Rich Adamson wrote: Spec sheets are available for the TigerJet 320 pci chipset as well as the Silcon Labs 3050, 3210 chip sets used on the TDM card. If you dig through those I think you'll find that it would difficult if not impossible to change the card's infrastructure since its based on the standards noted above. You're still not answering his question. :-) The TJ320 has no buffering capability but the Silabs parts have nothing to do with that, and neither do the framers on the higher-density TDM cards. I think his question is more why don't these cards have bigger PCM buffers and interrupt less often, or at least have deeper buffers so if an interrupt is delayed I don't get overruns? I believe the answer lies in latency. You do *not* want deeper buffers. Better interrupt handlers, perhaps, but not any deeper buffering on the hardware, as that just increases latency. -A. ___ --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] Sangoma E1 board Experience
[EMAIL PROTECTED] wrote: Its not a limitation. Its an architectural design which is based on pulse code modulation (pcm) standards, which essentially says: - 8,000 audio samples per second, - each sample is an 8-bit value - resulting in 64,000 bits/second (like g711 codec standard) Thank you for your answer, but I don't think you read/understood my question? The Zaptel driver uses a 8 byte buffer for buffering G.711 in each direction. This has nothing to do with the sample rate, but how large packets you can buffer the voice for to be transported through the PCI interface. I get the impression that the Digium boards are limited to 8 bytes (1 ms ) by reading the zaptel driver. But, I don't know this cause I have no hardware manual that cover the PCI interface. I was hoping that the Sangoma boards offered a more variable buffer size??? The current cards from Digium use bus mastering, so the amount of buffering on the board should not be a big issue, unless the PCI latency in really bad. The 1ms buffering is a driver issue, and was a design choice which made a lot of sense originally. It might make less sense now. - 1ms ensured the latency between E1/T1 ports bridged in the drievr was low, and EC would not be an issue. The latest revision of the 4 port E1/T1 cards from Digium now have on board bridging, so this is less of an issue. It still matters when bridging between cards, though. - 1ms ensured good EC convergence, using software EC. Adding delay really degrades the performance of an EC adaption loop. It may be a block size of 2 or 3 ms might have been a better compromise between adaption performance, and the necessary response time of the software. However, making the block size considerably bigger - say ta 20ms block, which is what * generally works with in its core - would significantly degrade EC adaption. Now there are cards with hardware EC, for which 20ms blocks seems to make a lot of sense. However, right now they still all work in the same 1ms block manner. Regards, Steve ___ --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] Extension processing misunderstanding
Hi,
I am trying to configure asterisk to send all calls which come in on the second port of my linksys pap2(which isattached to a fax)to send out my FXO trunk line.I have setup seperate sip profiles for both ports, the second port defaults to a [fax] context.
If I have the context configured as below, then it works for numbers which match the number patterns.
[fax]
exten = _,1,Dial(${TRUNK}/${EXTEN},45)exten = _,2,Congestionexten = _0X,1,Dial(${TRUNK}/${EXTEN},45)exten = _0X,2,Congestion
However, shouldn't I be able to use the s number pattern as such for one simple rule? It does not work for me.
[fax]
exten = s,1,Dial(${TRUNK}/${EXTEN},45)exten = s,2,Congestion
Thanks,
Bradley
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Re: [Asterisk-Users] Sangoma E1 board Experience
Spec sheets are available for the TigerJet 320 pci chipset as well as the Silcon Labs 3050, 3210 chip sets used on the TDM card. If you dig through those I think you'll find that it would difficult if not impossible to change the card's infrastructure since its based on the standards noted above. You're still not answering his question. :-) The TJ320 has no buffering capability but the Silabs parts have nothing to do with that, and neither do the framers on the higher-density TDM cards. I think his question is more why don't these cards have bigger PCM buffers and interrupt less often, or at least have deeper buffers so if an interrupt is delayed I don't get overruns? I believe the answer lies in latency. You do *not* want deeper buffers. Better interrupt handlers, perhaps, but not any deeper buffering on the hardware, as that just increases latency. I agree 100%. But, I don't believe any of that can be changed anyway on the digium cards; I'm 90% sure the card's buffering (and therefore interrupt frequency) is hardwired within the chip sets. (I've not tried to analyze the T1/E1 cards, only the TDM card.) ___ --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] Configuration of two Asterisk server
Mantu Jha wrote: Hi I am have two Asterisk server at two different location one is having static ip 203.101.42.14 and other is having static ip 10.42.16.1 how can i integrate both so that i can use the others dial plan. It's all here on this page. http://www.voip-info.org/wiki-Asterisk+-+dual+servers You can use the switch statement on server 203.101.42.14 to make the server 10.42.16.1's dial plan available. You can then dial extensions registered on 203.101.42.14 from 10.42.16.1 You cannot use the switch statement on both servers though. Only one. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] What is the best Dell Machine for Asterisk?
Then, how about Acer? Does it work well with asterisk? Simone Cittadini wrote: Matt Florell ha scritto: The best Dell for a production environment Asterisk server is no Dell at all. They make some great workstations, but I've had many problems with their servers(as have many others on this list) when trying to use them in production for Asterisk. Take a look at the Digium compatibility list: http://www.digium.com/index.php?menu=compatibility *I've installed a PowerEdge 2850 - Xeon 3.0GHz/2MB, 800FSB with two TE410P in it, the cards didn't worked out of the box, but they worked after a couple of hours googling around, and it is in production since 3 months, never gone down. * *(I'm not advocating dell, actually I don't even like dell as a society, only sharing my experience) * ___ --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 ___ --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] What is the best Dell Machine for Asterisk?
How big of RAM for Asterisk server? My production environment will be about 400 users in the office. Matt Florell wrote: The best Dell for a production environment Asterisk server is no Dell at all. They make some great workstations, but I've had many problems with their servers(as have many others on this list) when trying to use them in production for Asterisk. Take a look at the Digium compatibility list: http://www.digium.com/index.php?menu=compatibility You will notice that there are several Dells on there as well. The best solution is to build your own server with an Asus or SuperMicro Motherboard in it or buy a SuperMicro system from one of the many vendors that assemble systems with SuperMicro boards in them(just do a google search for supermicro servers). Hope that helps, MATT--- On 12/16/05, Hiu Yen Onn [EMAIL PROTECTED] wrote: hi all, What is the best Asterisk-compliant for Dell machine is recommended? I will have roughly 400 users in a production office. thanks!! ___ --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 ___ --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 ___ --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] Anybody having trouble terminating calls at Voxee? eom
-- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005 ___ --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] Odd problem with Encore 201-SA (r2 converter) with asterisk
Dear comunity: I use a r2mf converter called SignalPath model 201-SA from Encore Networks, i configure my Asterisk Box and i receive calls wothout problem, but when i try to make a Outgoing call, sound a busy signall after few seconds. I Think is a lost parameter in my zapata.conf file, but if anybody has experience with this kind of product and asterisk, your help are welcome. my zapata. looks this way: context=incalls switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe usecallingpres=yes usecallerid=yes callerid=asreceived hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=-5.0 txgain=-5.0 group=1 callgroup=1 immediate=no accountcode=Telmex-52766102 musiconhold=default channel = 1-15,17-46,48-62 My converter are PRI-Net, an take the syncronicity from R2mf Link. My zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 Thanks in advanced... Fernando Romo signature.asc Description: OpenPGP digital signature ___ --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] Odd problem with Encore 201-SA (r2 converter) with asterisk
Dear comunity: I use a r2mf converter called SignalPath model 201-SA from Encore Networks, i configure my Asterisk Box and i receive calls wothout problem, but when i try to make a Outgoing call, sound a busy signall after few seconds. I Think is a lost parameter in my zapata.conf file, but if anybody has experience with this kind of product and asterisk, your help are welcome. my zapata. looks this way: context=incalls switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe usecallingpres=yes usecallerid=yes callerid=asreceived hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=-5.0 txgain=-5.0 group=1 callgroup=1 immediate=no accountcode=Telmex-52766102 musiconhold=default channel = 1-15,17-46,48-62 My converter are PRI-Net, an take the syncronicity from R2mf Link. My zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 Thanks in advanced... Fernando Romo signature.asc Description: OpenPGP digital signature ___ --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] [Fwd: Odd problem with Encore 201-SA (r2 converter) with asterisk]
Dear comunity: I use a r2mf converter called SignalPath model 201-SA from Encore Networks, i configure my Asterisk Box and i receive calls wothout problem, but when i try to make a Outgoing call, sound a busy signall after few seconds. I Think is a lost parameter in my zapata.conf file, but if anybody has experience with this kind of product and asterisk, your help are welcome. my zapata. looks this way: context=incalls switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe usecallingpres=yes usecallerid=yes callerid=asreceived hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=-5.0 txgain=-5.0 group=1 callgroup=1 immediate=no accountcode=Telmex-52766102 musiconhold=default channel = 1-15,17-46,48-62 My converter are PRI-Net, an take the syncronicity from R2mf Link. My zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 Thanks in advanced... Fernando Romo ___ --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] SIP Remote Call Control
Hi All, I want to control a sip phone from my pc. I found a draft for this. http://www.faqs.org/ftp/pub/internet-drafts/draft-mahy-sip-remote-cc-01.txt Can someone let me know sip phones supporting this protocol or similar one? Thanks. Jason. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Asterisk Voice mail-reg
HI allHow to configure voice mail in asterisk . pls do the needful. regards ramakrishnan.n __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Recording Volume on Zap Channel
i have tried rsgain=100 txgain=100 recording volume improved but still not good --- Steve Totaro [EMAIL PROTECTED] wrote: Hi All I have a call center working on 8 FXO Channels, everything working fine except one little problem, I am using asterisk queues with monitor-format = wav49 and monitot-join = yes asterisk is recording all conversations but the problem is that the volume of Zap Channel is too low in most of the calls i am unable to understand what other person was saying (ZAP Channel) although Agent's (SIP Channel) vocie use to get recorded pretty good. Any suggession will be higly appreciated Thanks in Advance Did you try adjusting the gain in Zapata.conf? ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] SIP and echo cancel
On Dec 18, 2005, at 12:01 PM, Andrew Kohlsmith wrote: On Sunday 18 December 2005 14:32, Mohammad Shokuie wrote: As a matter of fact im serious to know where is the source of echo in a pure VoIP connection, i think the most of echo problems come from hybrid circuits which are not an issue in pure VoIP sessions. Easy. Get better endpoints. In a pure-voip loop you have echo due to acoustic coupling from the earpiece to the mic, or the speaker to the mic in a speakerphone. Easy way to tell: in a call with bad echo, have the other side mute. If your echo goes away, you've got your culprit. Also note that if your transmit level is too high or they have the volume up too loud on their end it could push the audio coupling over what the design specifications were. We're having some issues with a Budgetone, especially in speaker phone mode causing echo. I think I read the specs have a feature line item of Echo cancellation (pending), lol. No way to fix this other than buying new phone(s)? Phil ___ --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] sharing a line w/multiple extensions
On Thu, 2005-12-15 at 09:33 -0800, John Biundo wrote: I'm particularly worried about acceptance of this shared line (or lack thereof) aspect of the system. My wife will get the idea of extensions, transfers, parking, etc. because she uses a PBX at work, though I worry that the habits of how the phone is supposed to work at home may die hard with her. And the kids are a whole 'nuther story. IMHO, kids are the ones likely to use the more advanced features... ie, conferences, and anything else likely to tie up multiple in/outbound calls at the same time... You might consider permitting them to throw their own mp3's onto the server for their personal MoH I thought that having some common area phones share a single extension (wired into a single ATA FXS port) might ease the transition, but I'm also afraid it might be confusing (you can just pick up from these extensions, but you have to transfer or park to/from these extensions. Huh?). Program one of the speed-dial keys to transfer the call to a meetme with MoH. Then tell them that to put the call on hold, just press this button. Then label the next closest speed dial as Pickup which simply dials the meetme room. Of course, if you have multiple 'lines' you might need some 'custom' dialplan magic to ensure a hold will always add you to a new empty meetme, while an unhold will take you to the oldest meetme which hasn't been 'un-holded' yet :) Otherwise, you might as well teach them about parking Still, a button for Hold which does a #700, and then they just walk to any other phone and dial 701 (or whatever) should get you most of the way there. Just remember to set a short-ish parking timeout which will call-back the phone . The huge selling point, which I'm hoping will overcome any initial resistance, is the idea that one person will no longer tie up the whole phone system for the house when they make/take a call. And deploying one of my free DIDs to give my 16-year-old his own phone number that rings only in his bedroom is the real ace up my sleeve! Yep, neat + his own direct VM etc... Sure, Asterisk will come with a lot of other neat features, but frankly most of them have more geek appeal (though I have high hopes for my favorite feature -- announced caller id over the stereo/tivo while we're making dinner -- to revolutionize the way we deal with (or at least who answers ;-) ) phone calls at that hour), and in some cases I think may face similar that's not the way it's supposed to work objections. For example, while they will acknowledge that voicemail is cool, I suspect they'll miss the simplicity of walking into the kitchen, seeing if the answering machine is blinking, and just pressing the button. Use phones that have a VM indicator then program a speed dial for your VM extension. I'm excited AND anxious about starting a real beta test with them! Maybe that's why I'm already 3 weeks behind my original schedule. ;-) Well, looks like you are close, I think the biggest one to test thoroughly is the echo. That is probably the hardest one to ask people to live with... Regards, Adam ___ --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] SIP Trunk please help
Hi,
I already contacted what I inputed on my softphone but we
both can't hear each other. I used X-lite and the other is a hardware
SIP phone. What could be the problem?
Thanks,
Ryan
At 03:03 PM 12/16/05, you wrote:
yes
$AGI-exec('Dial', SIP/[EMAIL PROTECTED]);
Diyanat
From: Ryan Pagquil [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com, asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] SIP Trunk please help
Date: Fri, 16 Dec 2005 13:56:09 +0800
MIME-Version: 1.0
X-OriginalArrivalTime: 16 Dec 2005 05:58:00.0170 (UTC)
FILETIME=[AB7B14A0:01C60205]
Hi,
Thanks for the reply... Actually I'm using AGI to do it
instead of defining it on extensions.conf... Would it be the same
in extensions.conf? Should I write $AGI-exec('Dial',
'SIP/[EMAIL PROTECTED]'); to dial it from AGI script (perl), is this correct?
Thank you very much,
Ryan
At 01:45 PM 12/16/05, Diyanat Ali wrote:
in the sip.conf have the following enteries
; for regsitering with ser
register:seruser:[EMAIL PROTECTED]:5060;(put ser machine ip:port)
;add a user for the ser machine
[seruser]
type=friend
host=0.0.0.0 ;(put ser machine ip here)
nat=no ;(change as needed )
canreinvite=yes ;(change as needed)
insecure=very ;(change as needed)
disallow=all
allow=ulaw
allow=gsm
context=sip
dtmfmode=rfc2833
in extensions.conf under contect [sip]
[sip]
;replace extension and the priority to macth your dial plan
exten = _X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ;(seruser is defined
in sip.conf)
Diyanat
From: Ryan Pagquil [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP Trunk please help
Date: Fri, 16 Dec 2005 10:31:24 +0800
MIME-Version: 1.0
Hi,
I've been setting up asterisk for prepaid use. I'm
testing to call a SER registered user from the Asterisk just to
simulate the prepaid calls. Now, I can already contact Asterisk
and it prompts me to input my call card number and after that I
dial in the number I want to call (a SER registered device). My
question is how can I implement on sip.conf to use my SER as the
trunk line? So that calls will be forwarded to it. Do I also
need to register asterisk on SER?How?
Please help!
Thanks,
Ryan
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Re: [Asterisk-Users] Configuration of two Asterisk server
JP Carballo wrote: Mantu Jha wrote: Hi I am have two Asterisk server at two different location one is having static ip 203.101.42.14 and other is having static ip 10.42.16.1 how can i integrate both so that i can use the others dial plan. It's all here on this page. http://www.voip-info.org/wiki-Asterisk+-+dual+servers You can use the switch statement on server 203.101.42.14 to make the server 10.42.16.1's dial plan available. You can then dial extensions registered on 203.101.42.14 from 10.42.16.1 You cannot use the switch statement on both servers though. Only one. Doh, slight correction. I should have said that you could then dial extensions registered on 10.42.16.1 from 203.101.42.14 -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Anybody having trouble terminating calls at Voxee? eom
Tom Lynn wrote: Their trunk works fine as of the time this email is sent. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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
