RE: [Asterisk-Users] User-Oriented Management of Asterisk
Im def interest let me know if you want a beta tester =) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Hobbs Sent: 27 July 2004 2:26 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] User-Oriented Management of Asterisk While I was away on vacation, buried deeply in another thread (New Asterisk bounty: SIP simultaneous), Olle E. Johansson raised a question which is close to my heart - Asterisk's management model. A management model which simply manages telephone extensions and dial plans is irrelevant to most organisations. We need a model which manages users and their interaction with the PBX. I am currently constructing a CIM model of Asterisk from a user view-point so that Asterisk could offer a WBEM management interface, thus fitting in with the other enterprise equipment in the office. There is some interest in this model within the DMTF. Is anyone else interested? Chris -- Chris Hobbs Nortel Networks Tel: +1 613 765 5386 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help! BT UK has striken again ! I lost CLID!
Could be a faulty adsl filter had that with my metor. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vassilis Konstantinou Sent: 22 July 2004 10:09 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help! BT UK has striken again ! I lost CLID! I am a very happy Asterisk user/installer for the last couple of months. I regularly compile teh latest CVS. I got RC1 working and I as using the UK patches for the BT caller ID with no problems ...that is until this morning :-( I have an Asterisk X100 card going through the master socket of an engineer installed ADSL connection. Up to this morning all was trouble free. Then BT upgraded my 512K connection to 1mb. Everything seems to be working ok apart from the caller ID identification. In the past Asteriks used to detect it properly but now I get: NOTICE[294930]: callerid.c:245 callerid_feed: Caller*ID failed checksum I have seen in the past some people reporting some twicks for the rtgain to resolve this but I can't find what I need to change. Can somebody in a similar position help? Vassilis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where can i get an UK SIP account with UK number?
What I want to know is why you can port mobile numbers from network to network but say you have a local std code and you wanted to use that with a VoIP provider in the UK - in most instances you cant. Some are available to offer STD codes in certain towns but not all. Surely someone in the UK needs to address this issue for VoIP to become a more common? Its probably more of a oftel/ofcom thing. Kind Regards, Chris Bond -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Turner Sent: 16 July 2004 3:50 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Where can i get an UK SIP account with UK number? Dameon D. Welch-Abernathy wrote: > > There are probably others. Such as www.intervivo.net. Cheers, Mark. p.s. I work there. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
I have to agree with andy's comments the lack of documentation is * biggest downfall. Andy gave me a lot of help getting * up and running, without much of the help I probably would have not been able to see the full potention of *. Kind Regards, Chris Bond > Late last year I was approached by a publisher asking if I would be interested in writing an > asterisk book. I said a polite no (after some discussion) for a number of reasons: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
Please see my previous post - if you install identd it will give you a valid name. Identd is quite common service and usually very safe to open remotely. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Antti Lohikoski Sent: 09 July 2004 1:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net Hi! And thanks for helping me out here. Ok, I have an invalid username - how do I get a valid username? Thx Antti -- Terveisin: Antti Lohikoski Sinipiianpolku 12 02100 ESPOO GSM +358 (0) 50 337 5999 koti +358 (0) 9 46 16 84 [EMAIL PROTECTED] -- >>> [EMAIL PROTECTED] 07/09/04 3:17 PM >>> On Fri, 9 Jul 2004, Antti Lohikoski wrote: > and "No identd (auth) response" followed with "Closing Link: StiX > (Invalid username [~antti.loh])" Maybe your username is invalid. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IRC channel #asterisk on irc.freenode.net
On Fri, 9 Jul 2004, Antti Lohikoski wrote: > and "No identd (auth) response" followed with "Closing Link: StiX > (Invalid username [~antti.loh])" > Maybe your username is invalid. Install identd and allow TCP port 113 inbound access and it'll work - if you play about with your username it'll probably work too. Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel, Line Impedence and Echo
Modprobe wcfxs opermade=UK is what I was using - if my card worked =) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Yager Sent: 02 July 2004 1:19 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Zaptel, Line Impedence and Echo Hi, I'm not sure if I just missed something somewhere along the way, but I noticed while I was going through the CVS logs that there is an option in the wcfxs module to set an "opermode" - which apparently might help with echo issues around the globe (like the ones I'm seeing - some times). So I'd love to use it and give it a go. Apparently, there are some options that can be passed to the module to help set these values. The output of modinfo -p wcfxs is: debug int robust int _opermode int opermode string timingonly int lowpower int boostringer int I'm in Australia, and I know that _opermode should be set to 3 and opermode should be set to AUSTRALIA. Can someone with more knowledge enlighten me as to how I might go about doing this - or indeed if I need to? (If it helps, I'm running Fedora Core 1) Thanks in advance, Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re:Latest Echo changes
Just received it today - ultra fast shipping from digium. Will let people know the results of echo when I switch it tonight. -Original Message- From: Chris Bond [mailto:[EMAIL PROTECTED] Sent: 28 June 2004 4:44 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re:Latest Echo changes Just spoke to someone at telappliant and there not willing to sell the cards in the uk yet as there not ratified to the UK standard. I've just spoke to someone at digium direct and there forfilling backorders at the moment. I've just placed an order at http://store.yahoo.com/asteriskpbx/newitd1pofxo.html. The guy recokens I they should start shipping at the end of the week. Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re:Latest Echo changes
Just spoke to someone at telappliant and there not willing to sell the cards in the uk yet as there not ratified to the UK standard. I've just spoke to someone at digium direct and there forfilling backorders at the moment. I've just placed an order at http://store.yahoo.com/asteriskpbx/newitd1pofxo.html. The guy recokens I they should start shipping at the end of the week. Kind Regards, Chris Bond -Original Message- From: Chris Stenton [mailto:[EMAIL PROTECTED] Sent: 28 June 2004 10:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re:Latest Echo changes Yes but telappliant (the uk disti) have yet to get approval for it in the UK. I've just fired of an e-mail to them as they said they should have it by the end of the month. As you say though you can go direct ... Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re:Latest Echo changes
I believe its out if you call digium direct - im gonna give them a call later see what the latest is. From: taf taffey [mailto:[EMAIL PROTECTED] Sent: 28 June 2004 10:31 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Re:Latest Echo changes Cheers Chris! Any idea when the new FXO Module will be available? My setup = Grandstream/ATA186 > Asterisk > FXO
RE: [Asterisk-Users] Re:Latest Echo changes
> As you are in the UK I assume you are using the X101P like me. The best you can do with this > card is compile agressive echo cancelling on and not have the tx gain too high. I hope that > when the new FXO module is available here the issue will go away. Out of curiostity anychance you can list what you did? Settings etc, you say not have it too high etc, what have you got yours set to? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple X100P in Asterisk box?
You can use the new digium TDM cards with 4x FXO modules you'd only need 2 then. -Original Message- From: Yiannis Costopoulos, Web2Net Solutions Ltd. [mailto:[EMAIL PROTECTED] Sent: 27 June 2004 16:11 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Multiple X100P in Asterisk box? Hi, I am the "IT guy" at a small startup based in UK. At the moment we have 3 analogue (PSTN) lines and we will be adding another 2 or 3 soon. Later on we should be changing to ISDN30. One of the partners mentioned getting an analogue PBX now, and when we move to ISDN, then get a digital PBX. I though of Asterisk. I have seen the website in the past and I know that it can do the job (even better than a PBX). I need to know, is it possible to have multiple X100P (5 or 6) cards in the same Asterisk box? What phones should I use in the office? Any suggestions? Are soft phones reliable enough on Windows PC's for a small call-centre/telesales department (5-6 stations)? Thanks, Yiannis. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
> I do get echo, lots of it, I am waiting until the new patch they are all on about on the > list gets into a stable release, then I will upgrade and see if that does the trick. The patch didn't seem to work for me. > I am told that some of the echo may be to do with a mismatch in the impedance with the BT > line. Problem is do we really want BT messing with gain there end and impedance cos it might mess our ADSL lines up =) I know im on the limits. Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
> I am finding that I have to increase the txgain in zapata.conf to 8 when > my X101P is connected to my BT phone line, otherwise people can hardly > hear me. This then gives echo issues. Im having the same issue so far im on rxgain=2.0 and txgain=6.0. Seems to work perfectly apart from the echo issue. Im just about to checkout the latest cvs and apply the echotraining=800 Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
> Per the doc in the configs samples, you have to implement echotraining=800 (instead of > "yes") to take advantage of the new code from yesterday. Was this in the new samples from the CVS? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Thousands of contexts?
Why not use mysql as it should be faster I'd suspect -Original Message- From: Manuel Wenger [mailto:[EMAIL PROTECTED] Sent: 18 June 2004 5:43 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Thousands of contexts? By reading the Wiki's I found out that an Asterisk server with many (>1) extensions and/or SIP users can become slow when reloading. But what happens when you also have many contexts in extensions.conf? More precisely, one context for each SIP user? I need this because I will have users with random usernames that they can choose, but I obviously cannot set that username as the outgoing caller ID when passing the call to our PSTN gateway. I need to change the CLI before dialling out. Now, every SIP user has his CLI, so I thought of creating a context for every user, where I would SetCallerID() before issuing the Dial() command. Obviously I would use some sort of script reading from a database to re-create the extensions.conf and sip.conf after making changes. Do you see any issues which could arise? Is Asterisk going to crash, or is it just going to be slow when reloading? Thank you for your help -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK install
We're thinking of doing the same with our argent office system at the moment. > The Argent system is running about 30 POTS phones. Can someone suggest > the cheapest option? Should I get some kind of large scale FXS box or > would the cost of doing that on a large scale work out the same as > getting cheap SIP phones? Best bet is to use an IAXy or supr to convert the phone into an IAX2. The supr's have passthrough Ethernet ports so easier to do. I suggest you get one to try first if you have headsets that's what we're in the process of doing soon. The other way to do it would be get an ADTRAN 650 or 750, you can pick them up cheap on ebay. But this requires an extra PRI interface for each ADTRAN box (unless there linked so they run via single T1 termination). > Our Telco is NTL offering us an ISDN 30 style package. I assume this is > a E100P card requirement? Any suggestions for good UK reseller or shall > I get it direct from Digium? Should work yes, digium direct are good or telappliant. Would be really good if you could post your config files on a website once you've got it all up wouldn't mind seeing the config, as we're about 3 months off before we think of converting ours. Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BT Caller ID - From Patch ?
Another solution to get uk caller id is to get a meteor unit, andy powell
kindly wrote a few agi scripts and the perl script for the meteor to
integerate it fully with *.
-Original Message-
From: Kannaiyan Natesan [mailto:[EMAIL PROTECTED]
Sent: 17 June 2004 6:20 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] BT Caller ID - From Patch ?
Any body used patch,
http://bugs.digium.com/bug_view_page.php?bug_id=0001719
to get the callerid for BT Line.
I applied the patch successfully but could not get it to work.
Any help.
Here are the logs:
-- Starting simple switch on 'Zap/1-1'
Jun 17 18:22:31 NOTICE[426000]: chan_zap.c:4811 ss_thread: Got event 2
(Ring/Answered)...
Jun 17 18:22:34 NOTICE[426000]: chan_zap.c:4811 ss_thread: Got event 2
(Ring/Answered)...
-- Executing MySQLput("Zap/1-1", "cid/cid=s") in new stack
-- mysqlput: family=cid, key=cid, value=s
-- Executing Dial("Zap/1-1", "SIP/12345|20|tr") in new stack
-- Called 12345
-- SIP/12345-c377 is ringing
== Spawn extension (default, s, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
and in my extensions.conf
exten => s,1,MySQLPut(cid/02085034825=${CALLERID}${EXTEN})
exten => s,2,Dial(SIP/12345,20,tr)
can anyone please help what could be the problem.
My Zapata.conf:
[channels]
busydetect=1
busycount=7
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
ukcallerid=yes
echotraining=yes
echocancel=yes
echocancelwhenbridged=yes
jitterbuffers=4
rxgain=0.0
txgain=0.0
group=1
pickupgroup=1-4
immediate=no
context=default
signalling=fxs_ks
callerid=asreceived
channel=1
Am I missing anything here?
Kannaiyan
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SMS in the UK
Fancy knocking up a howto for it? From: Gary Ruddock [mailto:[EMAIL PROTECTED] Sent: 16 June 2004 9:34 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SMS in the UK I managed to get sms text messaging working and integrated with our system. This was my last major task. I would like to take this opportunity and thank everyone involved in asterisk. I've been able to automate my business in every way I wished. When I've got a few quid I won't forget you! Want to block unwanted pop-ups? Download the free MSN Toolbar now!
[Asterisk-Users] DECT delay once hungup
I've got the following setup: IAXy -> Dect Base Station. When you dial from a SIP phone (cisco 7960), the rings with very little delay. However, if you hangup it takes 3-4 rings after hanging up before the dect base station phone stops ringing. The same applies when an incoming call is directed from PSTN FXO -> Dect Base. Is there a fix to this I've looked about on voip-info but cant find any information that might be causing it. Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] XML How To for Cisco 7960
What you got in your sip cnf files? -Original Message- From: Iain Stevenson [mailto:[EMAIL PROTECTED] Sent: 11 June 2004 2:37 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] XML How To for Cisco 7960 Ah, how? Which SIP version do you have - 'cos I've made innumerable tests of my own (and using Cisco code) containing SoftKey commands and the phone always barfs. Iain --On Friday, June 11, 2004 8:09 pm +1000 Simon Brown <[EMAIL PROTECTED]> wrote: > Yes they do !! > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson > Sent: Friday, 11 June 2004 19:53 > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] XML How To for Cisco 7960 > > > > --On Friday, June 11, 2004 10:46 am +0200 Stefan de Konink > <[EMAIL PROTECTED]> wrote: > >> http://ipphones.utelisys.net/ >> http://ipphones.utelisys.net/includes/cisco.inc.phps >> >> There are some perl classes on this topic too (even for image >> generation!). I didn't had the time to made a GD patch to use it >> inside PHP yet. But I hope this wil help. Anyway on Cisco.com you can >> find some PDF files with clear statements. Only thing that doesn't >> work is HTTP_PUSH :( >> > > SoftKeys don't work either :-( > > Iain > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to get the Called id with AGI
Helped me so don't worry =) -Original Message- From: Karl Dyson [mailto:[EMAIL PROTECTED] Sent: 11 June 2004 9:04 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How to get the Called id with AGI Just realised I answered completely the wrong question. Misread "called id" as "caller id". D'Oh. Sorry, Karl This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony
> Yes, you are right!! > However, GUI for newbie's will help some people to overcome the first > hurdles, and then plunge into more advanced stuff! One thing quote a lot of companies do is outsource the initial configuration, because they simply don't have the technical skills initially. But what you want then is a way to easily go in and add an extension, remove someone who's left, setup hunt groups, etc, etc. It's more the general day to day maintenance that needs to be addressed, editing really complex IVR's, dialplans, etc I think should be left to the people who know what there doing. (Although there's nothing stopping adding an advanced interface too..) Just my thoughts =) Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony
> The power of asterisk comes from its method of config. If one wraps it > with a GUI one will inherently limit the flexibility. > Then since the GUI is what gets 'seen' people ~may~ take the lack of > flexibility or even just the look and flow of the GUI to be a reflection > on the power of Asterisk. But if it was an official addon from the cvs tree (similar to the voicemail cgi stuff), it would make take-up a lot easier =) That way you wouldn't make people "stuck" to one GUI, if they don't want it they don't need to check it out. Its just at the moment, you've got sub projects for lots of different GUIs, what needs to happen is someone to consolidate what's out there and bring it all into one official project. It makes sense that the GUI becomes a web one, then it can run on a number of web browser platforms. Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NetworkWorld article on Open Source Telephony
> This is the traditional view of telecoms in large organisations. However > it seems in a lot of large companies they are dumping their existing > telecoms wholesale for an IP solution, on a site by site basis, as soon > as the maintainence contract renewal comes around. It surprises me to > see that, and maybe I have seen a very unrepresentative sample, but in > some places it does appear to be happening. Of course, right now things > like * do not have an adequate reputation to pick up much of that > business. There is, however, a preparedness there for radical change. I think one thing * is lacking at the moment is a web interface to manage and add users and do anything you can do via a shell interface. If it had that but on a simplified level (oblessly you can have an advanced mode too). It could also integrate with the CDR, meetup, sms, voicemail functions that exist in *. So rather than have different projects for over view of who's on the phone and to who, etc you have one management interface. Just my opinion, at the moment I don't know enough about * to start writing an interface like this. But im sure some of the guys on the list do =) Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
asterisk-users@lists.digium.com
Just wondering if anybody has got AT&T Voices working for text->speech on *? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DNS SRV records
www.xname.org =) -Original Message- From: Stephen Rosebush [mailto:[EMAIL PROTECTED] Sent: 02 June 2004 2:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DNS SRV records I have the same problem, I have a domain name but I do not want to pay for DNS services... I kept on trying to find a place where I can get SRV records from but none of the free DNS services provide them. I've tried ZoneEdit, DNS Park, etc. I've seen one which there might be a possibility, www.granitecanyon.com.. though they do not offer mail forwarding. Andrew Thompson wrote: >My DNS gui(Cpanel/WHM) only allows the following options for entry type: > >A6 > >CNAME >MX >NS >PTR >TXT >WRK > >Does anyone know if any of these options are acceptable substitutes for an >SRV record, or do I need to put in a ticket to have a SRV record >specifically created for me? > >- >Andrew Thompson >http://aktzero.com/ > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hyperthreading?
What cards was it FXO - cos is it card related this HT problem? -Original Message- From: Manuel Wenger [mailto:[EMAIL PROTECTED] Sent: 01 June 2004 3:18 PM To: [EMAIL PROTECTED] Subject: R: [Asterisk-Users] Hyperthreading? That's the problem we had with Asterisk and HT on a 2.4 Kernel: whenever Asterisk was staying in the RTP stream, and HT was enabled (on a Dell Dual Xeon system), we had choppy audio. After disabling HT, everything was fine again. Nothing measurable, indeed, but you could definitely hear it. So there *must* be something. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Hyperthreading?
Ah - can any other user confirm that the FXO card works with hyperthreading enabled? -Original Message- From: Maron Kristófersson [mailto:[EMAIL PROTECTED] Sent: 01 June 2004 1:12 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Hyperthreading? Not using any cards at the moment here, However I will have an E100 card installed later this week. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Hyperthreading?
What cards you using currently I've just got one FXO card that I need to use with it. -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: 01 June 2004 12:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Hyperthreading? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hyperthreading?
Are they any issues still with hyperthreading processors, I've read and been told by a few people to make sure its disabled in bios if I want to use * on a hyperthreading machine. Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delay when routing PSTN -> IAXy dect phone
Just setup *, got a developers kit FXO where the incoming/outgoing pstn is
plugged in. I've then got an IAXy that is plugged into a Philips DECT
phone. * is setup so that the [bell] section rings the phone - exten =>
s,1,Dial(IAX2/myuser,30)
What's happening when someone calls my number is that the phone rings 3
times before you hear the dect phone ringing.
Anybody got any ideas as its driving me crazy?
Logs from * (don't worry about the agi script here I've tried it with those
lines commented and get the same results):
May 29 15:32:10 NOTICE[-1373635664]: chan_zap.c:4797 ss_thread: Got event 2
(Ring/Answered)...
May 29 15:32:12 NOTICE[-1373635664]: chan_zap.c:4797 ss_thread: Got event 2
(Ring/Answered)...
May 29 15:32:13 NOTICE[-1373635664]: chan_zap.c:4797 ss_thread: Got event 2
(Ring/Answered)...
May 29 15:32:15 NOTICE[-1373635664]: chan_zap.c:4797 ss_thread: Got event 2
(Ring/Answered)...
-- Executing AGI("Zap/1-1", "mclid.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/mclid.agi
-- AGI Script Executing Application: (SETCIDNAME) Options: (0143211)
-- AGI Script Executing Application: (SETCIDNUM) Options: (0143211)
-- AGI Script mclid.agi completed, returning 0
-- Executing Dial("Zap/1-1", "IAX2/myuser|30") in new stack
-- Called myuser
-- Call accepted by 192.168.0.60 (format ULAW)
-- Format for call is ULAW
-- IAX2[myuser]/4 is ringing
-- IAX2[myuser]/4 answered Zap/1-1
-- Hungup 'IAX2[myuser]/4'
Kind Regards,
Chris Bond
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web Based Frontend
Are there any web based frontends for asterisk, for mananging voice mail etc and asterisk in general? Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
