Re: [asterisk-users] Good by asterisk 1.4? Please not.
On 16/04/11 12:33 AM, Kristijan Vrban wrote: Security only fixes: 2011-04-21 So in six days, no more bugfix patches will committed into 1.4-branch :( Is a prolongation possible? Because 1.4 is so reliable now. It would be a great loss. And no, 1.8 is not (yet) a replacement. It actually brings up a good point. We've just reverted a couple of installs from 1.6.2 because of deadlocks. What version should we be going to? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems With DAHDI on Ubuntu
On 14/04/11 1:48 AM, Shawn L wrote: I have 2 separate Asterisk servers that are both exibiting this problem. 1 has a 4 port FXO digium card, the other an 8 port. For some reason when the machine reboots, the dahdi drivers are not properly loaded. Then asterisk To fix it, all i have to do is login and run dahdi_cfg /etc/init.d/asterisk restart but that's a pain to have to do after every reboot. I've never had this Don't know why it's happening, but add those lines to /etc/rc.local as a quick hack in the interim until you find out what's causing it. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crashes on high IO load
On 6/04/11 12:39 AM, Maximilian Grobecker wrote: Hi, the log files contained (sometimes) lines about refcount -1 in astobj.c. I also generated core dumps and analyzed them - but there were always errors in another module. Mabye I found the solution: Asterisk seems to crash when a required module cannot be loaded fast enough due to heavy disk usage. When I move the modules directory to another hard disk Asterisk runs fine. I'm using autoload=yes in modules.conf and have several noload lines in it. Is there a possibility to say asterisk to load all modules to RAM at start time and not on demand? You could compile Asterisk with embedded modules? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Number Conversion
On 5/04/11 1:00 PM, Flavio Miranda wrote:
Hi all,
Please, could somebody point me out what is going wrong in this line below?
exten = _00XX.,1,Dial(DAHDI/G0/021${EXTEN:4},45,rT)
As I know, such line must convert any number dialed to 021, therefore,
as we can see, it's kept the number dialed!
It must not be running that line - have you done a dialplan reload?
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Re: [asterisk-users] How to send Hold invite from asterisk to other
On 16/03/11 5:43 PM, Nikhil wrote: ok..that means I have to modify chan_sip . I wondering why this is not available in asterisk. Because you haven't completed the patch yet! :P -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Executing shell commands via AMI
On 17/03/11 9:53 AM, Vinícius Fontes wrote:
No increased security, lots of hassle, all because there's an
undocumented feature that is supposed to increase security but just
takes functionality away.
If you really want to you could add some dialplan like:
[dangerous]
exten = s,1,System(${somecommand})
and use the manager to set the somecommand variable on a call you send
to the dangerous context.
Up to you.
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Re: [asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)
On 11/03/11 7:52 AM, Nick Ustinov wrote: These are the same for sip users and trunks disallow=all allow=ulaw allow=alaw allow=gsm allow=g729 Who is asking to transmit frame type slin ? Maybe transcodeviaslin or something with a Local channel? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone have BRI working with Asterisk 1.8, Latest DAHDI, LibPRI?
Hi, We have a site where we'd like to move from mISDN/chan_lcr to DAHDI with a b410p card. We've tried everything we can think of to get it working but we never seem to receive any calls etc - even though the card has no alarms. We've tried replacing the card, changing the jumpers etc but no go. The cards both work with mISDN and chan_lcr, but we get reasonably frequent crashes. Does anyone have BRI working at all with the latest Asterisk, DAHDI, LibPRI? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Bandwidth Calculator
On 4/03/11 12:15 AM, Dan Journo wrote: Hi, Does anyone have a good VoIP Bandwidth Calculator? http://www.asteriskguru.com/tools/bandwidth_calculator.php -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to use qualify times to route calls
On 3/03/11 11:29 AM, sean darcy wrote: I'm using 1.8.3, and have 2 sip providers. Both are set with qualify=yes. Each of them sometimes have qualify times 10+ times the other. For instance, one will be at 10-15ms, the other at 200ms. Is there a way I can route an outgoing call to the provider with the lower qualify time? Traditionally you'd use a value you consider to be good enough for calls and set qualify to that. I.E. if you think 30ms is ok then set qualify=30 and then just route via the first then the second depending on status. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to use qualify times to route calls
On 3/03/11 11:34 AM, Danny Nicholas wrote: getprov.agi does sip show peers and gets the qualify time from status. The low value is returned in the variable BESTPROV. If you're going to do that, you could probably knock something up with the SIPPEER function - SIPPEER(status). -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] B410P: DAHDI BRI PTMP HDLC Abort (6) on Primary D-channel of span 1 (TEI Errors)
On 2/02/11 7:05 AM, Olivier wrote: Hi Matt, Too bad I can't be more helpful on this but could work around this issue ? Nah, in the end I just learnt how to use LCR with mISDN. I upgraded DAHDI, LibPRI, Asterisk to latest versions and still no go - although the errors stopped happening. I'm going to try again this weekend - with a different b410P card - even though it works with mISDN :) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gtalk/Jabber Issue
On 11/02/11 6:54 PM, William Stillwell wrote: I was getting unable to make channel.. We couldn't get it to work properly until we upgraded to Asterisk 1.8 at which stage it magically started working (with the same configs etc). -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
On 18/02/11 1:00 PM, Kevin P. Fleming wrote: On 02/17/2011 05:51 PM, Albert wrote: Linksys SPA921, SPA922, SPA941, SPA942 are also working pretty well. ... and have all been discontinued by Cisco. Kinda, they've pretty much just rebadged them Cisco SPA303 etc - all the same options in the website, pretty much same location, phones look the same etc etc. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] B410P: DAHDI BRI PTMP HDLC Abort (6) on Primary D-channel of span 1 (TEI Errors)
Hi all, So, we reverted the LibPRI version and tested it, and then tried with the latest version of everything. Still no changes. The BRI line is in PTMP. If we set the configs to PTMP in the genconf_parameters and try it, we get the following: [Jan 21 17:32:20] ERROR[20341]: chan_dahdi.c:12645 dahdi_pri_error: Unable to receive TEI from network! If we set it to PTP (which it is not) we get the following message: [Jan 21 17:33:42] ERROR[20418]: chan_dahdi.c:12645 dahdi_pri_error: Received MDL/TEI managemement message, but configured for mode other than PTMP! So, with PTMP it says we don't get a TEI message and without it, it says we do! :) Either way we also get the following message non stop in the console: [Jan 21 17:33:42] NOTICE[20419]: chan_dahdi.c:12946 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Jan 21 17:33:42] NOTICE[20419]: chan_dahdi.c:12946 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 If we change the hardhdlc to dchannel instead the message goes away, but obviously it doesn't work :) So, anyone have any ideas? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On 17/01/11 4:29 PM, jon pounder wrote: Surely there is some mail client smart enough to be able to flip around the levels of indenting so most recent is top or bottom. If not quit bitching and make one - I will continue top posting since I don't seem to be alone in preferring it. I'm definitely more keen on inline replies - if you reply to 20 points in someone's email you quote the part you're replying to then reply to it. In a long email it's the only way. Otherwise you'd scroll down to find the question, scroll up to find the answer, scroll down to find the next question, scroll up for the next answer etc - crazy. Much easier when replies are inline with the questions. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI
On 17/12/10 5:56 PM, Olivier wrote: Hi, Did you use libpri 1.4.11.5 or 1.4.12-beta ? Recently l tried 1.4.11.5 on a live system and it failed (Asterisk 1.6.1.18 and dahdi trunk, Junghanns QuadBRI, PtmP lines). Going back to 1.4.11.2 solved it. Unfortunately, I couldn't note what error message were then generated. Heh, latest everything - so LibPRI trunk. I did try going backwards in terms of DAHDI, but not LibPRI - will try that on Monday. By the way, Kevin/Russell etc, any chance we could get a test added to bamboo for physical connectivity? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI
) Card 0 Span 2 AMI/ccs span=2,2,0,ccs,ami # termtype: te bchan=4-5 hardhdlc=6 echocanceller=mg2,4-5 # Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 AMI/ccs RED span=3,3,0,ccs,ami # termtype: te bchan=7-8 hardhdlc=9 echocanceller=mg2,7-8 # Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 AMI/ccs RED span=4,4,0,ccs,ami # termtype: te bchan=10-11 hardhdlc=12 echocanceller=mg2,10-11 # Global data loadzone= nz defaultzone = nz == Here's the dahdi-channels.conf: ; Autogenerated by /usr/sbin/dahdi_genconf on Thu Dec 16 18:33:59 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS group=2,11 context=external switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 context = default group = 63 ; Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 AMI/CCS group=2,12 context=external switchtype = euroisdn signalling = bri_cpe_ptmp channel = 4-5 context = default group = 63 ; Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 AMI/CCS RED group=2,13 context=external switchtype = euroisdn signalling = bri_cpe_ptmp channel = 7-8 context = default group = 63 ; Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 AMI/CCS RED group=2,14 context=external switchtype = euroisdn signalling = bri_cpe_ptmp channel = 10-11 context = default group = 63 == Anyone have any ideas? Obviously it doesn't receive or make any calls - pri is in provisioned, down, active. I had to reinstall the old box (mISDN etc) temporarily until we can get this sorted. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI
hardhdlc=3 echocanceller=mg2,1-2 # Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 AMI/ccs span=2,2,0,ccs,ami # termtype: te bchan=4-5 hardhdlc=6 echocanceller=mg2,4-5 # Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 AMI/ccs RED span=3,3,0,ccs,ami # termtype: te bchan=7-8 hardhdlc=9 echocanceller=mg2,7-8 # Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 AMI/ccs RED span=4,4,0,ccs,ami # termtype: te bchan=10-11 hardhdlc=12 echocanceller=mg2,10-11 # Global data loadzone= nz defaultzone = nz == Here's the dahdi-channels.conf: ; Autogenerated by /usr/sbin/dahdi_genconf on Thu Dec 16 18:33:59 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS group=2,11 context=external switchtype = euroisdn signalling = bri_cpe_ptmp channel = 1-2 context = default group = 63 ; Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 AMI/CCS group=2,12 context=external switchtype = euroisdn signalling = bri_cpe_ptmp channel = 4-5 context = default group = 63 ; Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 AMI/CCS RED group=2,13 context=external switchtype = euroisdn signalling = bri_cpe_ptmp channel = 7-8 context = default group = 63 ; Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 AMI/CCS RED group=2,14 context=external switchtype = euroisdn signalling = bri_cpe_ptmp channel = 10-11 context = default group = 63 == Anyone have any ideas? Obviously it doesn't receive or make any calls - pri is in provisioned, down, active. I had to reinstall the old box (mISDN etc) temporarily until we can get this sorted. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ADA: DOA?
On 8/10/10 6:02 AM, Danny Nicholas wrote: FWIW, open source is only truly dead when you can't find anywhere to download the source. It wasn't ever Open Source, and source was never provided. I checked a while ago and that's never likely to happen, so yep (at least as of last time) it is a dead project. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DISA does not accept pause from cellphones when upgrading from 1.4 to 1.6
On 5/10/10 8:17 AM, Alejandro Recarey wrote: I just upgraded my asterisk box from 1.4 + Zaptel to 1.6 + DAHDI and services I was using perfectly before are suddenly broken. I have a DISA access configured, and my companies employees use if to dial into the companies extension from their cell phones. For example they would dial DISA-ACCESS-NUMBER(pause)EXTENSION. Has anybody else experienced this problem? Any tip would be welcome. Nope, but maybe you should try a double pause? Also, maybe try enabling DTMF logging in /etc/asterisk/logger.conf and doing a logger reload in the console? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3rd party app store
On 20/09/10 3:06 AM, Kevin P. Fleming wrote: There is no fee to list free products on AsteriskExchange. The main problem is the fee required to list non free products. If the fee was a percentage of the sale price then I'm sure it would work much better. Otherwise it becomes a catch 22. Nobody promotes the store because they can't afford to put their products on there, so nobody sells their products when listed on the store, so nobody list their products etc etc. If everybody who had products available was listing the products there, and Digium was taking a percentage cut, you'd see much better success from it, because people would redirect there. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk News Accepting Submissions
Hi all, Sorry for the crosspost but I assume this may be of interest to both businesses and users. The Daily Asterisk News (running since 2004) is now accepting article submissions. Basically I've created a submission form where you specify whether your post is commercial or non commercial and I'll be reviewing each article to check what it falls under. You'll be able to specify whether you want to hide commercial posts or not. What we're looking for is anything cool you're doing with Asterisk or any products you've created that work with Asterisk. If you have any ideas or suggestions, feel free to mail me on them. Oh, and we've moved the Daily Asterisk News web server to Dallas, TX, so it should be a bit quicker for those of you in the states - well, anywhere except New Zealand really :) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High volume BLF - Suggestions?
On 13/09/10 11:03 PM, Steve Davies wrote: On 13 September 2010 11:43, Olivieroza_4...@yahoo.fr wrote: 2010/9/13 Steve Daviesdavies...@gmail.com [snip] Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar As Snom phones have a parameter to express a time period during which BLF's SUBSCRIBE messages are spread and sent to Asterisk, I thought Snom phones would handle this load more easily. [snip] The SUBSCRIBE is handled fine, it is the NOTIFY messages that cause a problem. Yeah, I would have said Snoms too, not because I know them to work in that situation, but because they seem to be sorting various people's problems via firmware upgrades pretty regularly. I'd advise flicking the people over at Snom and email, then posting back here with your results :) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP softphones answer but do not connect...
On 11/09/10 12:44 PM, Carlos Chavez wrote: The past few days I started having a problem with a small call center setup. All agents use Eyebeam 1.5 to receive calls from a queue. Eyebeam is configured to auto answer the call. The problem is that the agents claim that they get a call but no audio. From the logs I can see that it is calling the agent phone but after 10 seconds (the queue timeout for pickup) I get the message that nobody answered and the call is sent to the next available agent. This can happen with up to three agents (the third finally answers the call). This has happened at least 20 times in the past two days. At first the supervisor thought that the same call was ringing on three different agents at once but the logs say that the first two do not answer and the third does. What strategy are you using for the Queue? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with iax call (chan unavailable)
On 11/09/10 2:07 AM, isca...@free.fr wrote: Hi, I have a problem with my IAX softphones. After a call, when the softphone hangup, it remains unavailable for the other softphones. It can call anybody, but can not be reached... For example, if A call B, B answer, then A or B hangup, and C won't be able to call A or B after that (but A or B would be able to call C). The Dial function returns that the chan is unavailable. That is very annoying, the only solution till now is to restart my softphones I must say that sometimes it works fine and i do not encounter this problem. But it happens very regularly, too often i would say. Try it with Zoiper and see how you go. I've not seen the same thing happen. It may also be that you are using qualify and that the peer is too far away. What do you get when you type iax2 show peers? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channel Signalling
On 3/09/10 8:32 AM, Arnaldo Giacomitti Junior wrote: There´s a way to get the channel signalling in dialplan? I have changed the code in channels/chan_dahdi.c and includes: Upload it as a patch to the issue tracker: http://issues.asterisk.org -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logging the CID from the Privacy Manager
On 1/09/10 11:27 AM, Jaap Winius wrote:
exten = jw,1,Verbose(-- CID is${CALLERID(num)})
exten = jw,n,GotoIf($[${CALLERID(num)}=anonymous]?true:false)
exten = jw,n(true),Set(CALLERID(num)=)
exten = jw,n(false),NoOp()
exten = jw,n,Verbose(-- CID is${CALLERID(num)})
exten = jw,n,PrivacyManager(3,10)
exten = jw,n,GotoIf($[${PRIVACYMGRSTATUS}=FAILED]?bad)
exten = jw,n,Verbose(-- CID is${CALLERID(num)})
exten = jw,n,Dial(SIP/1000,60,w)
Maybe you could do:
Set(CDR(userfield)=${CALLERID(num)})
Before dialing SIP/1000
--
Cheers,
Matt Riddell
___
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Re: [asterisk-users] Mobile answer machine cut off
On 25/08/10 7:35 PM, Julian Lyndon-Smith wrote: Hey Matt, thanks for the response. I know it sounds impossible. Hell, I sound like a user :) But it *is* happening. And only on the cisco phones. We're trying to lab it up right now. What should I be looking for in the sip debug ? Just something happening when the call gets cut off. Is there any DTMF being transmitted, why was the call disconnected etc. Or just take a snippet and put it up on pastebin/post here -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.8 problem with one-way audio with no nat
On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote: Hi. I have a soft phone -- expresstalk-- on a computer in my network and I use the internal ip address of the asterisk box to register the phone. But using asterisk-1.8 between revisions 281912 and 281982 it breaks -- after a few seconds of the call, I lose audio from the asterisk box to my soft phone, but not the other way around. This looks like one commit, but obviously I would like to know what's going on here? What's in the commit? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Early media and IAX2
On 28/08/10 10:18 AM, Russ Dill wrote: My IAX2 trunk provider, Teliax, seems to be forcing early media. Early media is cool and all, but my Asterisk install doesn't seem to be fully supporting it. My initial setting was using Dial() to call all of my dahdi (TDM400P) extensions. The results were that incoming calls would not hear any ringing tones and the call would be ended by Teliax after 21 seconds. You could just answer the call before dialling your internal extensions. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digest Username/auth name mismatch
On 30/08/10 2:48 PM, kawanobe tomohito wrote: Hi I want to know how to solve below an error case. Uac cant's change username of from and digest header. I tried to put a...@192.168.0.1 on username of sip.conf.but same error returned. You don't need to have the @192.168.0.1 in there - just make sure the username and password are correct in the user's device. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 - Separate Signaling and Media?
On 24/08/10 11:45 PM, Kevin P. Fleming wrote: On 08/23/2010 10:30 PM, Tim Nelson wrote: - Tim Nelsontnel...@rockbochs.com wrote: Greetings all- Here's an odd question. Supposedly, IAX2 now has the ability to operate with signaling and media in separate streams, very much like SIP. I've read about this feature here[1] and there[2], but I have yet to see how to actually implement or test it. There are no options in the iax.conf sample configs with Asterisk. All suggestions welcome, except those telling me to jump off a bridge because separated signaling and media makes IAX pointless when compared to SIP. :-) Ugh, and let me specify references as originally intended: [1] http://tools.ietf.org/search/rfc5456 [2] http://www.voip-info.org/wiki/view/IAX+versus+SIP I believe you are misinterpreting the RFC. IAX2 cannot use a separate signaling and media stream to setup a call, but it *can* optimize a media stream for a bridged call so that the media does not have to make as many hops as the signaling does. The media still moves on the same ports as the signaling packets, using the same protocol. And in that case the options are: transfer=no ; Disable IAX native transfer ;transfer=mediaonly ; When doing IAX native transfers, transfer ; only media stream Well, it depends on what version. The above is from a 1.4 system - earlier systems had notransfer=yes, but not the mediaonly option. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to debug this specific issue?
On 24/08/10 4:42 AM, Steve Davies wrote: Hi, I am happy with the usual GDB backtrace methods and so forth, but have an issue that I cannot work out how to trace on 1.6.2.10. If I use either the Bridge() app, or the manager Action: Bridge() in a certain scenario (Basically to bridge 2 SIP channels, like an attended transfer, resulting in 2 other SIP channels being discarded) then the whole server locks solid. The console stops, the network stops, something is hammering the box and nothing (including debug tools) seem to be able to do anything about it. If I 'nice' asterisk to lowest priority, and 'nice' a copy of 'top' to highest priority, everything still locks. After a short period, the box recovers, seemingly due to the 60 second RTP timer. Anything that was being logged is lost. Do you have high priority enabled in asterisk.conf? Are you using DAHDI? (maybe kernel not happy) I really thought that the canary should have sounded if Asterisk got in a loop - or maybe that only happens with high priority? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD message
On 20/08/10 1:52 AM, Tino wrote: Hello, Is there a way to capture the answering machine message when the dialer detects the answering machine. Record? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] billsec exceeds duration on some calls
On 21/08/10 5:24 AM, A J Stiles wrote: Can anyone else with a similar setup try running a query such as SELECT COUNT(*) FROM cdr WHERE calldate LIKE 2010-08-20% AND billsecduration ; and seeing if they have any calls like this? Nothing on this box (1.4) mysql SELECT COUNT(*) FROM cdr WHERE billsecduration; +--+ | COUNT(*) | +--+ |0 | +--+ 1 row in set (0.31 sec) mysql SELECT COUNT(*) FROM cdr; +--+ | COUNT(*) | +--+ | 190052 | +--+ 1 row in set (0.00 sec) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mobile answer machine cut off
On 22/08/10 10:38 AM, Julian Lyndon-Smith wrote: Crap, sorry, meant to add that we are running 1.4 svn head Julian On 21 August 2010 23:38, Julian Lyndon-Smithaster...@dotr.com wrote: We are having some strange issue where a call from asterisk dials a mobile number. If the number answers, we put the call through to an agent SIP phone. All works fine. If, however, the call goes straight through to the mobiles voicemail service *and* the agent phone is a Cisco 79xx, then the call is dropped (from the mobile end) about 1 second into the call. If the SIP phone is an Aastra9133i, then there is no problem. Has anyone seen anything like this ? Heh, seems impossible! Um, maybe the voicemail beep is the same tone as a * and * is used to disconnect a call or something? Try doing a SIP debug and see what turns up. Also make sure it's 100% repeatable :D -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD message
On 25/08/10 7:14 PM, Tino wrote: Yes, we need to record the message :D So use the Record() application :D -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to debug this specific issue?
On 25/08/10 7:20 PM, Tilghman Lesher wrote: I really thought that the canary should have sounded if Asterisk got in a loop - or maybe that only happens with high priority? The canary only runs in high priority mode, and it's only able to do anything if high priority scheduling is the culprit. If it's something else, like memory swapping, there's nothing the canary can do to fix that. Aha, explains why I've never seen the canary die :D -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Context
We don't use a context for that. We set up dialplan code in a non asterisk part of MySQL called routing types. When a customer selects a DDI number they can choose a routing type to use with it. These routing types allow for variable substitution - i.e. if someone adds the routing type Direct Routing With Failover, there is a variable with this type called failover routing. The call is then sent to their SIP or IAX2 device (if not on the same machine that has the DDI number it uses DUNDI to find the appropriate machine). If somebody is unavailable on all machines, it sets the account code to the customer and goes to the outbound context for dialling. The difference is that in the routing type we don't use extensions (just applications). When someone adds the routing type to their DDI realtime extensions are created with extension 5551234 (or whatever their DDI number is) and priorities which increase. It makes some things a bit harder, but you can always use labels and the read application. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor asterisk
On 17/08/10 6:34 PM, Hans Witvliet wrote: On Mon, 2010-08-16 at 13:35 -0400, Jamie A. Stapleton wrote: Might be worth your time to check out: http://www.humbuglabs.org/ Though they write: ... insight into the enterprise’s telephony infrastructure. Utilizing a set of none-intrusive analytical technologies, Humbug is capable of interfacing directly with your PBX system, analyzing its traffic, plotting it and providing ... It looks (!) like an online-service. Who would give an outsider access to your phone-usage info? :) Take it you don't use Google Analytics, Facebook insights, Feedburner, Amazon EC3 etc etc. Sure you have to decide who you want to trust (personally I trust the humbuglabs guys) and what their level of protection is (are they looking after their own security), but it seems to be the way things are going at the mo. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Convert wav-file to alaw-file
On 18/08/10 1:15 AM, Danny Nicholas wrote: As I interpret what I have read in this forum over the last several months, there is not really an “ALAW” format. If you do a sox (don’t remember if you can do in native Asterisk) convert to RAW (headerless) format, Asterisk is happy to consider that as ALAW. That would be signed linear (SLIN). -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does deny/permit work in sip.conf?
On 7/08/10 3:47 PM, Frank Church wrote: On 7 August 2010 03:54, Bruce Ferrellbferr...@baywinds.org wrote: On 08/06/2010 07:30 PM, Bruce Ferrell wrote: On 08/06/2010 02:16 PM, Frank Church wrote: On 6 August 2010 16:21, Bruce Ferrellbferr...@baywinds.org wrote: On 08/06/2010 07:45 AM, Frank Church wrote: I have been seeing some attempts to register devices on my Asterisk and I want to reconfigure it so that devices will be registered only if they are from the correct address, ie 192.168.1.8/255.255.255.255. I thought using a config like deny=0.0.0.0/0.0.0.0 permit=192.168.1.8/255.255.255.255 but it is not working the way I thought? Does that need a host=static.ip entry to work, rather than the deny/permit option? Does using a host=dynamic setting override any deny/permit and port=5060 options? Does being a peer or a user make a difference here? I had this same problem once. host=ip addressor host=dynamic if you want to use permit/deny. Permit/deny and host=dynamic allows a sip peer or user to have a range of addresses. -- Does permit/deny have any influence on registration, or is it related to the destinations it can call to or receive call from? How do you stop an asterisk server from accepting registrations when the IP is outside a subnet even if the username and secret are correct? When host=dynamic registrations are accepted even if the pemit IP is different from the registered device's IP address. Does permit/deny work on a single IP address eg 192.168.4.111/255.255.255.2555 The same seems to apply in the [general] section, with contactdeny and contacnt permit When I set contactdeny=0.0.0.0/0.0.0.0 contactpermit=192.168.4.111/255.255.255.255 Devices whose IP is not 192.168.4.111 are able to register. When I've used permit/deny, I did it in conjunction with insecure set to port,invite to allow gateways that didn't register and don't use username/secret to originate calls but only from the ip range in permit. In fact it was for a provider that had gateways on a large number of IP addresses, all in the same CIDR block and I didn't want to do an entry for each of more than 100 gateways. contactpermit/contactdeny *should* work as you are suggesting that you want I've never tried that. I may attempt it tonight and see on my 1.4 system. To follow up on my own reply. I just tried this with one of my standard peers that I use for a softphone on a 1.6.2.10 and see the registration attempt come in at the console and a warning comes up : Host '192.0.2.40' disallowed by contact ACL (violating IP 192.0.2.40) : Registration denied because of contact ACL The peer does show in sip show peers and the softphone (twinkle) shows a Registration Fails with a 603 denied. So I'd say it's working -- I am using 1.4.27 and it doesn't seem to work. I should probably try the 1.6 series Are you using deny before permit? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What TERMINAL software do you use for MS Windows platform and WHY?
On 30/06/10 1:53 AM, bruce bruce wrote: Hi Everyone, I am accustomed to PUTTY and it's very nice as in it allows many many SSH profiles to be saved and allows tunneling etcbut it's not very good when it comes to scrolling up and down, colors, text size, and specially it doesn't give a title to the opened instance. Maybe giving the IP address as the title of the window would help a lot if you have many different servers opened at the same time. Can you please weigh in and tell me what your favorite terminal software is and why? Late response, and I don't use Windows any more, but SecureCRT with tabbed SSH windows and buttons which can be set up for things like nano /etc/asterisk/extensions.conf make life pretty simple. On Mac I now use iTerm (similar thing). -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good script to make appointment?
On 16/07/10 4:40 AM, Gilles wrote: Hello I'd like to write a script that would make it easier for people to call in, listen to the IVR, and make an appointment (eg. When? ASAP? A given day? - Morning? Afternon, etc.) I assume I'm not the first one to try and write this type of IVR, so would appreciate any feedback on writing this. http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+Wake-Up+Call+PHP -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow-me to my answering machine :-(
On 23/04/10 10:31 AM, Bryan Jacobs wrote: Don, No, I'm not trying to say there's a problem with generating the tones. The issue is that my phone is still holstered, connected to the car via Bluetooth. I have steering-wheel buttons for receiving calls and hanging up, but I don't have a safe way to press buttons. Why not just use followme for everything but the car, and if that fails, send the call to the car normally? -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP over TCP
On 25/04/10 7:00 AM, bruce bruce wrote: Adobe Air and Adobe FMS are good examples of VoIP working flawlessly over TCP. We are actually developing a flash phone which needs only TCP to transmit both signal and audio. Ok, let's look at that (UDP vs TCP for realtime stream). Let's call the sender A and the received B. UDP === A sends packet 1 to B. Arrives ok. No problem A sends packet 2 to B. Doesn't arrive. No problem (dropped packet) A sends packet 3 to B. Doesn't arrive. No problem (dropped packet) A sends packet 4 to B. Arrives ok. No problem TCP === A sends packet 1 to B. Arrives ok. No problem A sends packet 2 to B. Doesn't arrive. TCP starts retransmit A sends packet 3 to B. Doesn't arrive. TCP starts retransmit A sends packet 2 to B. Arrives but is now 20ms too late (dropped packet) A sends packet 4 to B. Arrives ok. No problem A sends packet 3 to B. Arrives but is now 20ms too late (dropped packet) So, in the worst state of the network (when packets aren't getting though), TCP is sending even more data than is required for the actual conversation. And it's doing this at a time when the network is struggling. If we assume that there is a jitter buffer on B which is throwing away packets which are out of order then it's going to somewhat improve the situation. If it's not then it's going to be a disaster! Most of the flash based conferencing solutions (voice/video) I've used have all had the same problem (increasing delay and over utilisation of bandwidth). The reason people are using TCP for this is because flash doesn't allow you to do it with UDP. However, IIRC Adobe is working on a UDP based protocol for exchanging real time data and this should resolve the situation. If there is a great multiplexing video conferencing app which uses flash (or similar) that you can recommend, I'd love to know about it! Moral of the story: UDP is designed for realtime traffic or data where timing is more important than accuracy TCP is designed for important data (i.e. where accuracy is more important than timing) -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing For AsteirskAddon
On 27/04/10 7:33 PM, 675842709 wrote: when i install asterisk addon ,i got error here chan_ooh323.c:1934: error: dereferencing pointer to incomplete type chan_ooh323.c:1935: error: dereferencing pointer to incomplete type chan_ooh323.c:1937: error: dereferencing pointer to incomplete type chan_ooh323.c:1938: error: dereferencing pointer to incomplete type chan_ooh323.c:1940: error: dereferencing pointer to incomplete type chan_ooh323.c:1943: error: dereferencing pointer to incomplete type Do you need OpenH.323? If not, run make menuconfig and disable it -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow-me to my answering machine :-(
On 29/04/10 2:00 PM, Bryan Jacobs wrote: This is fine, except that it imposes a delay on connecting my call. If I were to do steps 12 simultaneously, then my cell phone being off would stop the phones in step #1 from working. If you play a message telling someone that you are being located, surely they'd prefer this delay than to not get hold of you? If you can't dial DTMF in your car, then there's really no other option - unless of course you can hum two tones at the same time :) I'd just call the sip phones etc, then play a message saying Please hold while you are transferred to my cell number. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [PATCH] Make Queue announcements more consistent (1.4.26.2)
On 27/04/10 2:21 AM, James Lamanna wrote: Hi, After playing around with queues a bunch on 1.4.26.2, I noticed a few things, which the patch below addresses. It addresses: - Callers in position 0 will hear periodic/position announcements at a very different rate than all other callers. -- Announcements while in position 0 could be delayed up to timeout+retry seconds. -- This patch reduces that possible delay to only timeout seconds - The say_position and periodic_announcement times are in elapsed time that _includes_ the time of the announcement. -- This patch changes those times to be the time _between_ playing of those announcements Please post this to issues.asterisk.org. Unfortunately developers are unable to look at or add patches without knowing the license. When you create an account on issues.asterisk.org you can file a disclaimer for the code and then the patch can be added to the base Asterisk install (assuming it meets coding guidelines etc). -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD reporting NOTSURE most of the time
On 24/03/10 3:06 PM, Steve Moran wrote: I am running Asterisk and using Answer machine detection with call files on a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over 50,000 outbound calls last week, and 70% said NOTSURE). I have a suspicion that the problem may be due to the timing source on virtual server when its under load delivering lots of asterisk calls, since the AMDSTATUS always reports things such as:- AMDSTATUS:NOTSURE-AMDCAUSE:TOOLONG-5500 Looks like it's missing the first word - some VoIP providers take a while to pass audio - might be that there is a delay in your dialplan or that the first words of audio are simply not transmitted. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD reporting NOTSURE most of the time
On 24/03/10 3:06 PM, Steve Moran wrote: I am running Asterisk and using Answer machine detection with call files on a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over 50,000 outbound calls last week, and 70% said NOTSURE). I have a suspicion that the problem may be due to the timing source on virtual server when its under load delivering lots of asterisk calls, since the AMDSTATUS always reports things such as:- AMDSTATUS:NOTSURE-AMDCAUSE:TOOLONG-5500 Alternatively your threshold might be too high - do a few tests to your own phone and make sure it recognizes the individual words. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] new server install errors starting asterisk
Just try running: asterisk -vcd And you'll see the error. Alternatively you can edit /etc/asterisk/logger.conf to allow you to have a full log. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.711a or G.711u ???
On 24/03/10 8:41 AM, Alejandro Cabrera Obed wrote: Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? It basically comes down to where the system is being used and what codecs you're using upstream. G.711a is aLaw and G.711u is uLaw. uLaw is predominantly used in the USA. aLaw is used in most of the rest of the world (although I think Japan might use uLaw). If you're using an ISDN card then it will be talking aLaw or uLaw depending on where you are. The idea is to avoid transcoding - i.e. converting between one format and another. So, if you're using a VoIP provider instead of ISDN, it will depend on what they're using. If your VoIP provider is outside of the US and accepts aLaw, then that's likely what you want to use (bear in mind that they might still use an upstream provider who uses G.729 etc). Easiest option is to just choose aLaw or uLaw based on your country. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free iPhone Asterisk Function and Application Reference
On 20/03/10 9:47 AM, Tzafrir Cohen wrote: On Fri, Mar 19, 2010 at 10:50:17AM -0400, Zeeshan Zakaria wrote: Hi Matt, This is very useful. But what about android platforms? Will it run on it? Just use an RSS reader. I guess browsers and RSS readers on the iPhone are too limited. There are a couple, the main diff with the Daily Asterisk News app is that you can vote, see similar articles etc. But no, it doesn't run on Android and I have no plans to write it for Android just yet :) Still getting to know iPhone/iPad SDK - once done I might start learning Android SDK :) -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app
On 20/03/10 3:46 AM, Zeeshan Zakaria wrote: Actually I might be wrong but haven't tried it yet because the download page is not available or the link is broken. I have however an iPhone too to try it. Which link is broken? I just clicked on it from the original email, and then clicked the download link. When done from an iPhone it brings up the app with a link to download, on my Mac it opens iTunes to the application page. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app
Hi all, I've released another free app for the iPhone and iPod touch - this one lets you read the Daily Asterisk News. Hope you enjoy it :D http://www.venturevoip.com/news.php?rssid=2371 -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue MOH
On 15/03/10 11:23 AM, Thomas Perron wrote: I want callers to enter a queue and then hear music on hold. does anyone have notes on how to integrate queuing to a dial plan that uses moh? You can just set the music on hold class for the Queue in queues.conf - you actually have to provide an option (r IIRC) to provide ringing instead of music on hold. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On 19/03/10 1:19 PM, Adrian Marsh wrote: Hello, I’m looking for some advice on securing Asterisk. Have a look at fail2ban: http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Management API
On 9/03/10 9:13 PM, Peter Childs wrote: Also is there some way to get the starting end to auto pickup, (or at least hit for this to happen (I'm using SIP if that helps)) When you make an originate request it works like this: 1. Call is made to the Channel parameter. 2. When the Channel answers it connects the other end to the application/context/extension. So, send the channel to the SIP device and then the other end won't start till the SIP device picks up. 2. Send DTMF to the far end, PlayDTMF looks like it should work but it seams to send the Play the DTMF to my end not the far end. I seam to be able to send it to the far end by finding far end channel's name and using that instead, but this does not work if the far end is not a channel, (eg the Answer phone) but I hope that will not really be a problem... Again, looks like you have the order of the channels round the wrong way. If you originated to a SIP device and sent the other end to the application PlayDTMF, then it would be sent to the SIP device (if that's what you want). -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls per second limit in manager
Also, why are you saying your name is Philip? On 24/02/2010, at 12:59 AM, CDR vene...@gmail.com wrote: My dear friend Matt Riddell insists that the Manager only can dial 5 calls per seconds, which I find ridiculous. Is there a way to prove him wrong and have him lift the limit that has been plaguing the life of us users of SineDialer and SmoothTorrque Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls per second limit in manager
The responses from the Asterisk manager on your machine start providing responses of no account code when calls are initiated at a higher rate. On 24/02/2010, at 12:59 AM, CDR vene...@gmail.com wrote: My dear friend Matt Riddell insists that the Manager only can dial 5 calls per seconds, which I find ridiculous. Is there a way to prove him wrong and have him lift the limit that has been plaguing the life of us users of SineDialer and SmoothTorrque Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls per second limit in manager
Yeah, the problem's not the origination. The problem is that calls originated asyn with accountcodes show up in show channels concise without details. Pretty simple to test with sipp and core show channels concise. I assume it's because the call origination happens at a faster rate than Asterisk can fill out the details. Apologies for top post, laptop is running a defrag. On 24/02/2010, at 9:32 AM, Tommy Botten Jensen tommy.jen...@freecode.no wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Olle E. Johansson skrev: 23 feb 2010 kl. 20.18 skrev Matt Riddell: The responses from the Asterisk manager on your machine start providing responses of no account code when calls are initiated at a higher rate. Where's the bug report id? I haven't heard about this limit. I don't know what it is, but we should at least be able to accept the originate requests in asynch mode, put them on a queue and process them in a separate thread (which can be configurable in manager.conf). This is just brainstorming - but first, let's try to find out if the limit you believe in exists in the code or is just the effect of something else. It seems to be the effect of something else - or perhaps an older asterisk version. I wrote a quick script that ran 80 calls in ~ 0.2 seconds with no problems what so ever. I am using asterisk 1.6.2.1, and the script authenticated for each time, and used the originate-application for it's calls. - - T On 24/02/2010, at 12:59 AM, CDR vene...@gmail.com wrote: My dear friend Matt Riddell insists that the Manager only can dial 5 calls per seconds, which I find ridiculous. Is there a way to prove him wrong and have him lift the limit that has been plaguing the life of us users of SineDialer and SmoothTorrque Philip -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEAREKAAYFAkuEO2EACgkQ573V05EH/pZDQwCfaotZoweNLI8cTQ+yxZ2tr7WK +YsAn3OxXc5ULAj4lPdiIhoBDG4Tm7Xp =XE12 -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls per second limit in manager
Yeah, so at say 10 calls per second originated from the manager with async on, you'd likely have about a thousand channels. Then if you type show channels concise you'll see about 20% of the calls are missing accountcode, destination etc. I wrote some code to just repeat this test over and over, and with 5 CPS you get maybe one or two channels in this state, but as you increase the CPS you end up with more. Initially I thought it might have been manager parsing, but did show channels concise from Asterisk console and got the same. On 24/02/2010, at 10:50 AM, Tommy Botten Jensen tommy.jen...@freecode.no wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Matt Riddell skrev: Yeah, the problem's not the origination. The problem is that calls originated asyn with accountcodes show up in show channels concise without details. Pretty simple to test with sipp and core show channels concise. I assume it's because the call origination happens at a faster rate than Asterisk can fill out the details. Apologies for top post, laptop is running a defrag. No worries. Did I misunderstand the bit about this being calls spawned from the AMI (Manager) only? And what details are missing? 'sip show concise' gives me the following: SIP/05-00ca!internal!!1!Ringing!(None)!!100!!3!26!(None)! 1266961653.567 ... and similar * n. Thanks, Tommy -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEAREKAAYFAkuETagACgkQ573V05EH/pb73QCgnUMlaghEOBjS9aVw5PKtJZy4 Tl0AoJ2kRTHr0tOJPxsXAdVPmoulGJJE =zS/n -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime extensions
On 19/02/10 8:15 AM, jonas kellens wrote: How about something like : [mycontext] exten = 100,1,NoOp(calling 100) exten = 100,n,NoOp(going realtime) switch = Realtime/mycont...@realtime_extensions mailto:mycont...@realtime_extensions ; from here on we use realtime And then my MySQL-DB contains : `extensions_table` VALUES (1, 'mycontext', '100', n, 'Wait', '2'); `extensions_table` VALUES (2, 'mycontext', '100', n, 'NoOp', 'into RealTime'); 'extensions_table` VALUES (3, 'mycontext', '100', n, 'Playback', 'my-sound-file'); I'm not sure that's likely to work - or if it does, not in the way you expect. Likely if you did a query for exten = 100, the n extensions would be returned in a random order. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime extensions
On 20/02/10 10:53 PM, jonas kellens wrote: I have read on this list that people do not get a reply if they ask stupid questions. Is this then a stupid question that I ask ? If nobody has ever combined extensions.conf and realtime in a way that I want to do, I wanna hear it too. Even if this means no solution for me. Then I know it's not doable. :) Maybe you should read the messages from the list then :) You've already been replied to. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Free iPhone Asterisk Function and Application Reference
Hi all, I've uploaded a free app for the iPhone called AsteriskRef to the Apple AppStore. This allows you to lookup applications and functions using your iPhone or iPod touch so you don't have to jump out of extensions.conf or open another terminal tab. It currently supports applications and functions from Asterisk 1.4, but I'm adding 1.6 and trunk at the moment. It currently requires OS3.1.3, but I've got another version under review at the moment which will run on 3.0. Hope you like it, let me know if you have any questions. More info here: http://www.venturevoip.com/news.php?rssid=2353 -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] strange asterisk behaviour on XEN
On 15/02/10 11:55 PM, Emre Kurnaz wrote: Hi all, Now a days we are planning to run two asterisk boxes on XEN with DNS Failover. But even using the default configuration asterisk shuts itself down at least 5 times in a day with an exit status of 139 (i think it should be 139-128=11 there may be a coding mistake). Thus what do you prefer to do? How can i examine the core dump file? http://www.voip-info.org/wiki/view/Asterisk+debugging#CoreSoftwareDebugging -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Losing local SIP phones when internet goes down?
On 9/02/10 12:59 PM, Tilghman Lesher wrote: add to the top of /etc/resolv.conf nameserver 127.0.0.1 If you're using DHCP on any of your interfaces, you'll need to configure dhclient (or whatever dhcp client you're using) to prepend in the configuration with (e.g. /etc/dhcp3/dhclient.conf): prepend domain-name-servers 127.0.0.1; Otherwise, your entry in resolv.conf will be overwritten on each DHCP lease renewal. Yeah, although if you're using DHCP, then dnsmasq is possibly a better option. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Losing local SIP phones when internet goes down?
On 6/02/10 4:06 AM, Dave Cotton wrote: On 05/02/10 16:01, Jeff LaCoursiere wrote: On Fri, 5 Feb 2010, Vinícius Fontes wrote: I solved similar issues by setting srvlookup=no, having bind running locally and just the line nameserver 127.0.0.1 on /etc/resolv.conf. Your local bind is what solved the problem. The srvlookup=no didn't actually help IMO. Given the choice between configuring bind and dnsmasq I know which I'd go for. They're both pretty easy - bind9 easier I reckon. To set up on debian do: apt-get install bind9 add to the top of /etc/resolv.conf nameserver 127.0.0.1 Then it's done. Dnsmasq is probably overkill for this type of thing, though some people in the office prefer it to bind. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MATH
On 31/01/10 6:27 PM, Thomas Perron wrote:
what is wrong with this please:
;exten = 4,1,WaitExten(3)
exten = 4,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten = 4,n,WaitExten(3)
exten = 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
exten = 2,n,Waitexten(3)
exten = 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
exten = 3,n,WaitExten(3)
exten = 9,1,SayNumber(${TOTAL})
Heh, you might need to say what you're expecting and what you're getting :D
Straight off, all I can see is that 2 does 200, 3 does 300 and 4 does 500.
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Re: [asterisk-users] Unable to create channel of type 'DAHDI'(cause 0- Unknown)
On 30/01/10 11:48 AM, sean darcy wrote: Sigh. OK you don't like asterisk - sorry. Obviously some other software works better for you. I'm glad. Don't worry, he/she's trolling, second post like that for the day :) Obviously has an issue with something, but rather than try and get it sorted he/she'd rather just bitch. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax client for symbian s60
On 28/01/10 9:14 PM, Asterisk - thinking:systems wrote: Hi all, I searched for a long time and know that here this question also was asked in the past, but ... Is there any iax client for s60 now? Or still no client available? There are so many people asking for it, but nobody seems to get it done :-( Not that I'm aware of - best place to ask would be the IAXClient mailing list, but I'm pretty sure I'd remember if someone had written one. Probably the closest would be Tim Panton's work - maybe hunt him down :D -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jitterbuffer and PLC
What user are you running Asterisk as? -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jitterbuffer and PLC
On 16/01/10 12:56 AM, nak...@02.246.ne.jp wrote: Hi, I have a question about jitterbuffer and PLC. Do you get the same results if you use: iax2 test losspct x Where x is the loss percent you'd like to test? -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cheap ip phone with auto-answer
On 29/12/09 10:22 AM, Leif Neland wrote: I want some cheap ip-phones with auto-answer, to work as paging system at dinnertime. Options, please. Use some of the Chinese PA1688 or AR1688 phones - support auto answer, IAX/SIP etc. Prices around $45 -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA9x2 echo problem
On 7/12/09 8:11 AM, Dubravko Caric wrote: Hi all, we have this annoying problem with Linksys SPA9x2 phones and echo cancellation. I have read posts on other sites about this problem but they are more than one year old and people were using older firmware. Linksys/Cisco has released 6.1.5a firmware but we still experience the same problems. SPA phones have low sound volume of handset microphone and people on the other side (PSTN, GSM) are complaining that they hear us very badly. I increased Handset gain to +6 and lower Additional handset gain to -3 to get amplification of +3 what gives higher volume on the other side but not as much as I would like. That is one problem, the other one is that now VoIP to VoIP calls in our company are much louder than those to PSTN and GSM, and if we receive two calls immediately one after another there is a big difference between PSTN and VoIP calls in the sound volume on our side and our users are complaining. We are using SIP trunk towards PSTN (so I can’t use txgain and rxgain in zapata.conf), for GSM calls we are using PORTech SIPtoGSM GW (but I can’t increase volume on it anymore because I am getting echo) and I can’t increase volume on Linksys phones anymore because our internal VoIP calls would then be much too loud. Does anyone have same issues even with this new firmware and if so how do you handle those problems? Sounds to me like you need to speak with the company providing you a SIP trunk. If the calls between VoIP-VoIP are too loud and the calls to PSTN are too quiet, then likely the provider needs to check their gain settings. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Source-IP on Asterisk DRBD/-HA-Cluster wrong
On 4/12/09 9:28 AM, Scott L. Lykens wrote: Apologize for not directly answering your questions, however, I'm considering playing with Remus and Xen in the future to deal with high availability without dropping calls. See http://dsg.cs.ubc.ca/remus/ for some details. I have no idea if it will work or what the implications are but I noticed that in doing research for some other projects and made a note of it to try in the future. Yeah, I was looking at that too - haven't had much time to work on it further, but if it can handle running Asterisk cleanly it looks like a pretty nice solution. I'm just not 100% convinced that it will work in a real time environment rather than for hosting web sites. I'm sure that it would be perfect if you wanted failover web sites without any downtime, but wonder how it would work with Asterisk. Post your progress as you move through it :) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio cuts out during IVR
On 25/11/09 5:47 AM, Dr. Michael J. Chudobiak wrote: Hi all, I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the audio vanishes in the middle of listening to an IVR background prompt. This happens with both analog (Digium card) and IAX2 incoming calls. The prompts are stored in ulaw format (and the IAX2 calls use ulaw). The asterisk console claims that the IVR prompts are proceeding in the expected fashion, but I can't hear anything. The logs don't report anything interesting. Has anyone seen anything like this? Suggestions? Is the machine running a GUI? I.E. Gnome/KDE/XFCE etc -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On 13/11/09 3:59 AM, Danny Nicholas wrote: Without the allowguest=no, Asterisk doesn't put up any defense against an unauthorized guest. You still have NAT/Firewall/IPTABLE defenses, for what they are worth. The trick is to get what you need without allowing what you don't want. A slight clarification - I wouldn't say it's defences. By default these calls are sent to the default context (which should not have the capability to make calls other than test the system). So, yes you are allowing unauthenticated calls, but to the echo test etc. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On 13/11/09 8:30 AM, SIP wrote: Eh... if VoIP fraud weren't so rampant, and I didn't constantly see mailings to the Asterisk list about How do I secure my system from the people who've been costing me tons of money lately, I would say that having a lax stance on security in exchange for additional usability might be a good thing. But as is, that's simply not the case. The 'usability' you get from this is really only questionably essential in its ability to save time, but the security one would get from a change could save some people actual money -- not just time. The problem there is normally lax usernames and passwords. Not that there is default access to the echo test. As someone who used to design systems and networks, I would vote for security over nebulous desire to keep the status quo. Because you're already using Asterisk. If it had been too hard at the start maybe you wouldn't. True, you can't keep stupid people from doing stupid things, but given a choice between protecting the ignorant from a bad situation or catering to those who want to avoid an extra step or two on installation, I'd side with protecting the ignorant every time. There's always a trade-off between usability and security, and I'm of the opinion that security is the more important of the two when dealing with systems connected to the Internet. Call me a cynic. :) The ignorant won't have changed the default context - they likely won't even know how to edit a config file - so they're safe. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On 13/11/09 9:37 AM, Lee Howard wrote: Michiel van Baak wrote: When I started working with asterisk, and found my first issue, I created a patch, put it on the tracker, followed up on the comments, and stuff got in. I'm sincerely pleased to know that you've had a different experience than have I. I've had an experience which is a little of both. I've had some patches accepted, and other not accepted (MySQL userfield2-5). I think it's really important that not every patch gets accepted, and I really like the discussion which has taken place on this one. Basically the two sides of the argument are: For: I put stuff in my default context, now people can use it without authentication - I didn't expect this. Against: I'm a new user, I tried to get Asterisk working but had authentication problems, now I'm moving to Microsoft OCS (or 3cx or whatever). I kinda think that you want to make it as easy as possible for new users to at least run an echo test (and maybe make a call through to Digium). Once they've done that they're going to need to edit config files. If there is strong wording in the config files explaining that they shouldn't be adding anything here without first reading the security document I think it would suffice. Maybe the best way would be to make it that the default context only provides the info from the examples unless you provide an option: read_security_document=yes or whatever. I know that it seems really easy for most of us to chuck a couple of sip devices into the config and set up some extensions, but for a new user, any step at all they need to make before getting a call working is bad. The average new user won't know much about VoIP, nor much (if anything) about Linux, and seeing some text interface provide some random error when they try it for the first time will just turn them away. If you read the page about contributing code to asterisk, it clearly states that the dev mailinglist is the place to discuss development. If you post comments there, people will read it, comment on it, and if more people agree with the ideas it will get implemented. It's how all OpenSource projects work. I truly wish it were. I've seen more than a few that didn't. :) just consider yourself lucky it's not glibc or something you're trying to commit to :) The people with commit access tend to just say no. Even if the change stops something from breaking on multiple platforms (see eglibc discussion). Basically to get a change into Asterisk, you need a reasonably good percentage of people agreeing that the change is worthwhile (and the best way to implement it). Don't get me wrong, I understand the change you're proposing, just that it may not be the 100% best way to do it, and it needs to be carefully thought out before proceeding with something which may have a large impact on new users. Think what it's like for the 3G video people who have a huge patchset that they wrote before bringing it up for discussion only to hear it was the wrong way to do it. At least the patch is small :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowguest defaults to yes for SIP
On 13/11/09 12:33 PM, Tzafrir Cohen wrote:
On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote:
Maybe the best way would be to make it that the default context only
provides the info from the examples unless you provide an option:
read_security_document=yes
Asterisk used to require that you set have 'TELEPHONY=yes' in
/etc/{sysconfig,default}/asterisk to start running. This is no longer
the case. Such requirements are not the thing that will make the user
read the documentation, and they get in the way of automating the
installation.
Yeah, but would you automate an install with additional contents in the
default context?
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Re: [asterisk-users] SIP source address error
On 12/11/09 9:04 AM, Jaap Winius wrote: Hi all, My Asterisk problem today involves getting a SIP client on a private net to register with a server somewhere else on the Internet. This worked for me about a year ago no problem, but now I see an error message on the remote server every time the client attempts to connect (the server is running Debian lenny with Asterisk 1:1.4.21.2~dfsg-3). Here's an example: [Nov 11 14:29:47] WARNING[6365]: chan_sip.c:1787 __sip_xmit: sip_xmit of 0xb63d5694 (len 444) to 192.168.8.30:5060 returned -1: Operation not permitted 192.168.8.30? At first I thought maybe the local NAT (iptables SNAT) wasn't doing its job properly, but it seems fine for the rest. Also, the same client, going through the same NAT, has no problem connecting to my ISP's SIP server. Then I thought it might be the SIP client (a Siemens Gigaset S675IP phone), but I get exactly the same problem when using an old analog phone with a Linksys SPA-3000 instead. Has anyone encountered this problem before? If so, what caused it and what solved it? Are you binding to an address that the box doesn't own? Check the top of sip.conf. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Import
On 11/11/09 12:56 AM, Philipp Kempgen wrote: Khaled W Chehab schrieb: how to write the cdr directly to the databse (Mysq)instead of importing Master.csv to table using a php script. Noting that I load asterisk_addons_mysql cdr_mysql from Asterisk Addons. Configuration file: /etc/asterisk/cdr_mysql.conf Also, the status check is cdr mysql status -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gradstream Budge Tone-201
On 10/11/09 1:12 PM, bilal ghayyad wrote: Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? I wouldn't recommend the BudgetTone - it's been a while since I used it, but there are better phones around (even from Grandstream). -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is an extension is use
On 10/11/09 1:02 PM, Conklin, Tom wrote: Have you taken a look at the following? http://www.astassistant.com/ Also: http://www.asternic.org and the newer version: http://www.fop2.com -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText
On 10/11/09 12:58 PM, Thomas Perron wrote: Does anyone have any success with sending a text message from extensions.conf to an PSTN endpoint such as a cell phone? If so, kindly send configuration for this part. I am working on an IVR and want callers to get a text message at a particular part of the call, after dialing a defined character (such as 22). We use clickatel. Basically we use the PHP API and call it via an AGI which sends texts. Therefore the extensions.conf is pretty sparse: exten = s,1,Read(destination) exten = s,2,AGI(agi://127.0.0.1/send_sms.php) Pseudo code for send_sms is: 1. Read AGI variables 2. Get destination variable 3. Include clickatel API file 4. call send_sms function We also provide an API from our telephone exchanges, but to be fair you're likely better off just using clickatel yourself :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to configure softphones in asterisk server
On 10/11/09 4:08 AM, C. Savinovich wrote: He wrote me too. I would have helped him, but the name on the email address threw me off. Poor guy - language/cultural barrier maybe? Here's some tips: 1. Read Asterisk The Future of Telephony (buy a copy or download from http://asteriskdocs.org) 2. Set up sip.conf/iax.conf based on what type of softphone 3. Download a softphone - I've listed a few here: http://www.venturevoip.com/news.php?rssid=2188 4. Make calls :D The most important step is number 1 - once you get the hang of Asterisk the rest will be easy :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText
On 10/11/09 4:19 PM, Thomas Perron wrote: Will text messages work to non-SIP enpoints using your logic/code? thank you If you mean SMS, yeah. Basically use SendText for devices which can display them (i.e. SIP/IAX phones) and Clickatel or the like for disconnected devices (i.e. SMS to mobile). If you wanted to extend it you could also use the Jabber functions to send to instant messaging clients. Here at the offices we basically do the following: SMS Messages for urgent notifications, payments received, support requests. Jabber Messages for incoming support call details, long Post Dial Delay warnings, congestion warnings. MRTG displaying IAX2 and SIP peer response times. Custom graphs to display inter country links. We use a system of circles around an international link. Each of our servers gets a circle. The larger the circle, the higher the delay, and if the host is unreachable the circle goes red. That way you can see from a quick glance if an international link is totally down (lots of red circles), a problem for one of our servers (one red circle), or if one of our servers is having trouble connecting to all remote links (one red circle on each link). We do the same circles for a couple of key customers to make sure their systems are always connected to multiple of our exchanges. Oh, the other thing we display on the dashboard is our Jabber statuses, and the number of tickets open in any of our support queues, and who they are assigned to. That way if someone is getting overloaded with support requests you can move jobs to another staff member. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendJabber question sending Links
On 6/11/09 10:21 PM, Stefan Schmidt wrote: thanks for your answer, i will try to say it in an easy way ;) i send now a jabber message which looks like this: snip Customer Nr [1234] Person ABC from Company XYZ CRM: https://crm.x.y/getcustomer?customer=1234 Ticket: https://rt.x.y/getticketsfromcustomer?customer=1234 /snip but what i want to have should look like this: snip Customer Nr [1234] Person ABC from Company XYZ CRM [URL] Ticket: [URL] /snip I´ve tried to send it in html style witha href=text/a and so on, but i didnt get it working. The Problem is that the links i send has around 400 Chars each which make the message long and hard to read. i hope its now clear what i want. Heh sounds like you need tinyurl.com. Or maybe just make a page a.php?1234 which loads getcustomer.php?customer=1234 Or with an Apache rewrite. It seems that what you're wanting is more on the jabber client side - you're wanting one that can receive messages and display them as pure HTML. There may be one - I don't think Adium (the client I use) does it, but if you had a look at a few different clients, maybe one will. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent cell phone voice mail capturing call
On 6/11/09 3:25 PM, Darrick Hartman wrote: Russell Horn wrote: Hi, I've a DID number that gets passed to three internal phones and a cell phone via my outbound IAX trunk. If the cell phone is off or out of coverage, its voice mail captures the call. What's the best way to avoid this? Is there a recommended way to force the cell phone user to press 1 before the call is passed there ala google voice? Or is there another way to detect the presence of the answering machine rather than a human? Thanks, Russell. Require the cell phone user to press a button to accept the call (much the same way that the followme app does). In fact it sounds like what he's actually wanting is the followme app: http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 remote pickup
On 6/11/09 3:37 AM, Antony Stone wrote: On Thursday 05 November 2009 14:28, Danny Nicholas wrote: Hi. I have several Asterisk 1.4.21 machines, each with ISDN cards in them, and Polycom SIP phones on people's desks. I'm trying to work out how to provide a remote pickup facility along the following lines: The normal (as defined in features.conf) way to pick the call would be *82233. Features.conf defines *8 as a global pickup to be followed by an extension. Thanks, I'll investigate this and see if that works instead. What we do is create an Asterisk database entry: Pickup/NUMBER/GROUP Where NUMBER is the extension, and Group is the Pickup Group. We then set pickup mark variable in the macro that dials the extension. Then if someone dials *79 (or whatever) it picks up the group that the person dialling *79 is in. I.E. * Call goes to Jon (who is in group 3) * He is away from his desk * Jane dials *79 (also in group 3) and picks up the call If Fred (in group 5) were to dial *79 he would not pick up the call. Names have been changed to protect the innocent :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-stat! - help needed (once again due to mailserver problem)
On 26/10/09 3:47 AM, Lukasz Pakula wrote:
Dear all,
I'm trying to install Asterisk-stat (ASTERISK CDR ANALYSER) following:
http://www.voip-info.org/wiki/index.php?page=Asterisk+CDR+Areski+GUI
however it fails to run properly - lots of lines like:
*Notice*: Undefined variable: s in
*/home/lukasz/DATA/www/asterisk-stat/cdr.php* on line *26*
*Notice*: Undefined variable: t in
*/home/lukasz/DATA/www/asterisk-stat/cdr.php* on line *27*
That's not an error - it's a notice - it means you have error_reporting
set to E_ALL in php.ini.
Depending on which version of Linux you use the file could be in a few
places.
If you are using Debian it would be in:
/etc/php5/apache2/php.ini
You'll need to restart Apache after changing the setting.
If you're brave you could surround the lines creating the problem with:
if (isset($s)) {
// Do something with $s
}
(replacing the commented line // with the line in question)
--
Cheers,
Matt Riddell
Director
___
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Re: [asterisk-users] Concurrent calls including mysql taking lot of time for execution
On 23/10/09 6:11 AM, jonas kellens wrote: On Thu, 2009-10-22 at 13:45 +1300, Matt Riddell wrote: It's really simple you just read from standard input and write to standard output. If you tell us a programming language you'd like to use (i.e. php/c/perl/bash etc) we can give you a link to some docs and examples. Might I highjack this thread to ask for this documentation ? I want to use php. :) Sorry been moving house for the week - easiest one to use for PHP is PHPAGI: http://phpagi.sourceforge.net/ -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX jitterbufer oddity
On 27/10/09 2:07 AM, Steve Davies wrote: Hi, First a confession - The box in question is a 1.2.35 box, so this may be solved in a newer version as I know the JB code is all hugely changed, but... It may be worth checking into. Scenario: - IAX outbound call from Asterisk, which rings okay. - Remote end sends ANSWER, which we immediately ACK. - The ANSWER control packet gets put into the JB (that's how I read the code) - The remote end is clustered, and we receive a TXREQ within 1ms of our ACK - chan_iax2 starts to process the TXREQ correctly. What I think happens at this point is that the ANSWER control frame now leaves the JB in order, but is not processed because the channel state has moved into the new transferring state, so ANSWER has no meaning, app_dial never forwards the ANSWER control event to the calling channel, and the bridge is never fully completed, so it all eventually times out. Disabling the JB in IAX does resolve the issue, but is not ideal. I have tried to follow the code in the various versions 1.2, 1.4 and 1.6, but it is just too complicated. Does anyone know if this was addressed since 1.2, or can it still happen in 1.4 or 1.6? Just a shot - all boxes using NTP? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - mISDN and B410P questions
On 25/10/09 11:52 AM, Paul Hales wrote: I have used both misdn and dahdi_bri over the last year, and would happy take dahdi if for no other reason that it's much easier to install. A patch is available to allow dahdi_bri to work with Asterisk 1.4, and I have used that successfully. Ooh really? Where would I find that? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - mISDN and B410P questions
On 25/10/09 11:52 AM, Paul Hales wrote: I have used both misdn and dahdi_bri over the last year, and would happy take dahdi if for no other reason that it's much easier to install. A patch is available to allow dahdi_bri to work with Asterisk 1.4, and I have used that successfully. Which brings me to another question - what does Digium recommend people use on a 1.4 system with their b410p card these days? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendJabber question sending Links
On 5/11/09 9:14 PM, Stefan Schmidt wrote: Hello, i use sendjabber notifications when a call is answered to send the answering user information about the caller also with links to our CRM or ticket system. My problem is that i dont know how i can make a link like CRM and not have to use http://crm.x.y/fubar?user=1234. i´ve allready googled for this question, but i´ve only found how to xml format an url, but not how i can send it with sendjabber application. Does anybody have an idea how i can do this? It might pay to rephrase your question. You're trying to send a link, and what's going wrong? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to dial multiple extensions at once likeinaring group and put them in conference?
On 29/10/09 2:15 AM, Danny Nicholas wrote: Mea Culpa?? Since I’ve only been dabbling with AMI for about 6 weeks, I hadn’t stumbled upon the Async parameter. A “more correct” dissertation of the sentence would be “The AMI originate by default operates in a synchronous or threaded fashion, unless you specify Asynchronous mode using Async: true”. Guess I’ll never be as smart as you, Matt. :D I should hope not!! If everyone was as smart as me, how would I take over the world? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
