[asterisk-users] Re: AsteriskNow console access
Took the easy way out. booted the system to single user mode by editing the grub menu and adding a 1 at the end. This game me shell access and i changed the root password. Geoff On 11/27/06, Geoff Karl <[EMAIL PROTECTED]> wrote: I just downloaded and installed the AsteriskNow appliance (http://www.asterisknow.org) . This looks like it has lots of promise. Anyone know what the secret is to being able to actually login to the root console? thanks, Geoff ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskNow console access
I just downloaded and installed the AsteriskNow appliance (http://www.asterisknow.org) . This looks like it has lots of promise. Anyone know what the secret is to being able to actually login to the root console? thanks, Geoff ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rtc: lost some interrupts at 1024 when loading ztdummy
I am running today's SVN of the 1.4 branch, on Ubuntu dapper. I compiled a custom kernel (2.6.15.7). Created modules of the rct and the rtc modlue loads fine. As soon as I load ztdummy the syslog fills up with: rtc: lost some interrupts at 1024 Hz. Any ideas what may be causing this? thanks, Geoff ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polling Asterisk for Life
On 3/2/06, Matt Riddell [NZ] <[EMAIL PROTECTED]> wrote: > Matt wrote: > > Yup.. that's the exact problem I'm having. I really can't explain > > what happens. If I don't restart asterisk it seems to happen after > > about 2 days. So I restart asterisk once a day at 3am. And it still > > goes down about once a month... > > Are you guys perchance using Local/[EMAIL PROTECTED] in your installations? > > -- > Cheers, > > Matt Riddell > ___ > Is there a known issue when using the Local/[EMAIL PROTECTED] thanks, Geoff ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialplan to try VOIP providers if they can't terminate call
I am trying to figure out how to try different VOIP providers if they aren't able to terminate the call because they don't offer service to that dialing area. The error that gets logged to the console is: Sep 13 00:01:43 WARNING[22093]: chan_iax2.c:6835 socket_read: Call rejected by x.x.x.x: No such context/extension The dialstatus returned is "No Answer" I could check for that dialstatus, but it is the same dial status as if someone didn't pick up the phone. How are ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] get dialstatus variable when returning No such context/extension
I have a list of VSPs that I use. Some are not able to terminate to
different locations. It appears they are returning this error
message:
Sep 13 00:01:43 WARNING[22093]: chan_iax2.c:6835 socket_read: Call
rejected by x.x.x.x: No such context/extension
I would like to find out what the dialstatus is on this so I can try a
different VSP that is able to terminate the call.
Right now I have this and it doesn't show the DIALSTATUS.
exten => _9X.,1,SetCallerID(6175551212)
exten => _9X.,2,NoOp(${EXTEN:1})
exten => _9X.,3,Dial(IAX2/VSP1/${EXTEN:1}|15|j)
exten => _9X.,4,NoOp(${DIALSTATUS})
exten => _9X.,104,Dial(IAX2/VSP2/${EXTEN:1}|15|j)
exten => _9X.,205,Dial(IAX2/VSP3/${EXTEN:1}|15|)
This works if the hosts are down, now I want to enhance it to try the
different providers if the aren't able to place the calls. If I can
figure out the DIALSTATUS then I can create a macro to handle the
logic.
thanks,
Geoff
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Re: [Asterisk-Users] Multiple Instances of Asterisk (no contexts)
On 9/8/05, Matthew Boehm <[EMAIL PROTECTED]> wrote: > Geoff Karl wrote: > > I know I have seen something on the mailing list describing how to run > > more than one instance of Asterisk. I can't find it anymore. > > > > What are the things to look for when running more than one copy. > > > > Yes, I know about contexts. > > > > thanks, > > > > Geoff > > This begs a repeated question: Why? The entire 'point' of contexts is > so you don't have to run multiple instances of asterisk. > > -Matthew > > Because I am going to do clustering to protect against hardware / system failure. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple Instances of Asterisk (no contexts)
I know I have seen something on the mailing list describing how to run more than one instance of Asterisk. I can't find it anymore. What are the things to look for when running more than one copy. Yes, I know about contexts. thanks, Geoff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux-HA Heartbeat2 and Asterisk
On 9/2/05, Matt Riddell <[EMAIL PROTECTED]> wrote: > Geoff Karl wrote: > > The new version of heartbeat (http://linux-ha.org/GettingStartedV2) > > supports up to 16nodes. I was wondering if anyone has tried it with > > Asterisk. > > > > The biggest hurdle would be to configure multiple instances of > > Asterisk on the same box. Anyone configure more than one copy of > > asterisk on the same machine? Each instance would need their own set > > of configuration files. This would be a VOIP only box with a zaptel > > interface for timing. > > Um...doesn't running the multiple copies on the same machine kinda make the > aim of redundancy redundant? > > :) > > Use different contexts if you want to split by companies, then failover to > other PCs if you have hardware/network problems. > > -- > Cheers, > > Matt Riddell When you are building out a HA clustering solution you "fail-over" an instance to a another machine (node). You also tie a virtual IP address to the asterisk instance. Pretty much everything moves from one machine to another. Lots of times people design a system like Active/Active/Active/Passive; where the Active nodes are running an instance of Asterisk and you have a Passive node to accept any other failed nodes instance. This means that Passive nodes needs to be able to run any Asterisk instance. Really what you are protecting here is hardware failure and file system issues. If you had an Active/Passive design you could run different version of Asterisk. This would allow you to try out a new version and if it failed you could easily switch back to the "working" version. Geoff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linux-HA Heartbeat2 and Asterisk
The new version of heartbeat (http://linux-ha.org/GettingStartedV2) supports up to 16nodes. I was wondering if anyone has tried it with Asterisk. The biggest hurdle would be to configure multiple instances of Asterisk on the same box. Anyone configure more than one copy of asterisk on the same machine? Each instance would need their own set of configuration files. This would be a VOIP only box with a zaptel interface for timing. Geoff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Loop error when compiling CVS version of 1.2-Beta
I am still getting an error compiling the 1.2-Beta version. The tarball works fine, but I have never been able to compile the 1.2beta from CVS. I have been compiling CVS-HEAD on the machine for quite some time. It goes into this loop: if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c ast_expr2.c ast_expr2f.c asterisk.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c config_old.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c This is the contents of the include/asterisk/version.h file. /* * version.h * Automatically generated */ #define ASTERISK_VERSION "CVS-Nv1-2-0-beta1-09/01/05-11:06:01" #define ASTERISK_VERSION_NUM 99 Geoff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztdummy and zttest results
I am running the latest ztdummy on 1.2 beta1. This is the zttest results I have: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% --- Results after 10 passes --- Best: 99.975586 -- Worst: 99.975586 -- Average: 99.975586 Is this what I should expect, or are people getting better results? By default does it use the RTC kernel module on 1.2? thanks, Geoff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile problem with 1.2 beta 1
On 8/29/05, Doug Lytle <[EMAIL PROTECTED]> wrote: > Julian Lyndon-Smith wrote: > > > Has anyone else got 1.2 compiled from cvs ? I've posted the question > > below to the -dev list but got no answers: > > > > Mine complies fine under Mandrake and a kernel downloaded from > kernel.org, ztdummy won't load, but other then that no issues. > > Doug > I get the same compile errors on Debian Sarge. I have been compiling previous CVS HEAD versions. I was able to compile the tarball. Geoff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Variuos hangup codes in Manager API for failover
On 8/28/05, Matt Riddell <[EMAIL PROTECTED]> wrote:
> Steve Edwards wrote:
> >> Normally the way I do it is to program the failover into the dialplan
> >> and then
> >> send the call to Local/[EMAIL PROTECTED] to initiate it.
> >
> > How about a snippet? (Local channels somewhat escape me.)
>
> Ok,
>
> If you had something like this (we're assuming +101 jumping for arguments sake
> here):
>
> [outbound]
> exten => _9X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> exten => _9X.,102,Dial(IAX/myiaxprovider/${EXTEN:1})
> exten => _9X.,203,Dial(IAX/myiaxprovider/${EXTEN:1})
>
> Then you could originate a call with the following channel:
>
> Local/[EMAIL PROTECTED]
>
> which would do the whole failover thing for you.
>
> Note that this is slightly simplified. The jumping behaviour has now been
> changed and will require the 'j' option in the latest versions unless you use
> gotoif and check the dialstatus.
>
> Normally you'd want to connect the originated call with an extension/context
> so that once that number answers it is connected to say an agent or an
> application. This part should be pretty self explanatory.
>
> Make sense now? Feel free to ask if it doesn't!
>
> :)
>
> --
> Cheers,
>
> Matt Riddell
> ___
Thanks Matt, that is a good strategy.
Any idea on how to pass the reason a call failed back through the
Asterisk Manager Interface? It would be great to send something back
like Busy, NoAnswer, etc...
Geoff
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Re: [Asterisk-Users] DIALSTATUS for Originate
On 28 Aug 2005 10:35:34 -, saket setu <[EMAIL PROTECTED]> wrote:
>
>
>
> Hi all,
> I am from India and has been recently using asterisk for testing and
> enahncing my telephony knowledge. I am trying to use the originate Command
> from the Asterisk manager on both SIP and ZAP. The command works successfully
> but does not return any DIALSTATUS such as BUSY,ANSWER,NOANSWER as in case of
> command DIAL when used from the dial plan. Can some one guide me how to get
> the vaue of $DIALSTUATUS on originate or is there some other way to trap the
> status both on SIP and ZAP.
>
> I have also tried to write a dial plan in a manner such that i originate a
> call to my internal extension and jump to a context in the dial plan and
> execute the Dial command and trap all the statuses but this also does not
> work and it straight away bridges my internal extension to the external call
> without returning any dial status.
>
> Here is the example of what i did:
> 1. Originate:
> Action: Originate
> Channel: SIP/201 (Internal extension)
> Context: Airtel
> Extension: 26191341(External PSTN Number)
> Priority: 1
>
> 2. Dial Plan :
> [AIRTEL]
> exten => _XX.,1,Dial(SIP/${ETEN},15,t)
> exten => _XX.,2,NoOp(${DIALSTATUS})
> exten => _XX.,3,Goto(_XX.-${DIALSTATUS},1)
> exten => _XX.-Busy,1,Hangup
> exten => _XX.-NOANSWER,1,Hangup
> exten => _XX.-ANSWER,1,Goto(s,1)
> exten => s,1,Queue(Airtel|r|||300)
>
> thanks
> Saket
Stefan Tichy Wrote:
Response: Success
Message: Originate successfully queued
Indeed this response to a originate manager command is not what you
may have expected. You can listen to the events provided by the
manager interface and wait for something like this:
Event: Newstate
Channel: SIP/201-
State: Up
--
If you are using Async and the action ID for some reason the Event:
Newstate doesn't respond with the ActionID, but only a automatically
generated "Uniqueid".
Any ideas on how to determine which ActionID is being returned?
Thanks,
Geoff
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[Asterisk-Users] Variuos hangup codes in Manager API for failover
I am currently using the Manager API to place a few calls. I have more than one VSP available. I was wondering how to best tell a call failed to move on to the next VSP. I see messages like this, which is an obvious failure, and I would then move on to the next VSP. Event: Hangup Cause: 3 Cause-txt: No route to destination Do people have others that they check for? thanks, Geoff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Streaming MP3's from Asterisk with Ices
On 7/21/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Thu, Jul 21, 2005 at 07:04:43AM -0700, Geoff Karl wrote: > > On 7/20/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > > On Mon, Jul 18, 2005 at 05:44:17PM -0700, Geoff Karl wrote: > > > > > > > Being that my end goal is to stream an mp3 file any ideas on how this > > > > should be configured. > > > > > > Why stream an mp3 file in the first place? Is the network saurated? Do > > > you really need the quality that mp3 offers you, just so you can > > > transcode it to phone quality and waste CPU in the process? > > > > > > I wonder if it would be useful to stream music from another server using > > > simply asterisk or a similar voip server: a client holds a permanent > > > connection somewhere and provides a stream of sound. > > > > I would like a client to be able to listen to a meetme conference > > without the need of any VOIP software. I think most people have the > > MP3 codec installed on their local machine, but they don't have OGG > > installed. > > > > Do you have other ideas on how this could be done? > > Provide a simple, dumbed-down iax client that will connect to your > server to a specific extension. iaxclient comes with a simple > command-line program that has all the functionality you need, and you > just need to put some GUI around it. Alternatively, iaxcomm should be > hackable. > > -- > Tzafrir Cohen | [EMAIL PROTECTED] | VIM is The other beneift of not using a voip client is reducing server load, since you only need to encode once. Geoff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Streaming MP3's from Asterisk with Ices
On 7/20/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Mon, Jul 18, 2005 at 05:44:17PM -0700, Geoff Karl wrote: > > > Being that my end goal is to stream an mp3 file any ideas on how this > > should be configured. > > Why stream an mp3 file in the first place? Is the network saurated? Do > you really need the quality that mp3 offers you, just so you can > transcode it to phone quality and waste CPU in the process? > > I wonder if it would be useful to stream music from another server using > simply asterisk or a similar voip server: a client holds a permanent > connection somewhere and provides a stream of sound. > > -- > Tzafrir Cohen | [EMAIL PROTECTED] | VIM is I would like a client to be able to listen to a meetme conference without the need of any VOIP software. I think most people have the MP3 codec installed on their local machine, but they don't have OGG installed. Do you have other ideas on how this could be done? thanks, Geoff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Streaming MP3's from Asterisk with Ices
On 7/19/05, Stefan de Konink <[EMAIL PROTECTED]> wrote: > On Tue, 19 Jul 2005, Geoff Karl wrote: > > > >From what i can see in the ices configuration there is no way to get > > an input other than an mp3 playlist. In order to work with Asterisk I > > need to use the stdinpcm input module. > > > > I am sure someone has a solution to get mp3 audio out of asterisk. > > What about ezstream, you can pipe audio in and stream it to icecast2? > > > Stefan > > That looks like it might work. Anyone hack ezstream together with Asterisk? thanks, Geoff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Streaming MP3's from Asterisk with Ices
Thanks for the reply. You have to use ices in order to encode mp3 and ices2 only encodes ogg. I should have been more descriptive. I would like to stream a meetme conference using icecast. In order to do that i need to get the media stream from Asterisk to Icecast. I think the only way to do this is using ices2 with Asterisk. >From what i can see in the ices configuration there is no way to get an input other than an mp3 playlist. In order to work with Asterisk I need to use the stdinpcm input module. I am sure someone has a solution to get mp3 audio out of asterisk. thanks, Geoff On 7/19/05, Flu <[EMAIL PROTECTED]> wrote: > As you mentioned, you'll need to use Ices in order to stream MP3s, I > believe Ices2 is only for OGG. Ices is fairly easy to setup, however. > Once you have the streaming working properly, you can then hook any > extension you like into the stream by using the "mp3player" application. > You'll also need to install mp3 support for asterisk in order for this > to work. You can do this by running "make mpg123" from the Asterisk > source directory. > > Once this is setup, you can setup the stream on an extension like so: > > exten => 100,1,MP3Player(http://64.236.34.196:80/stream/1074) > > Now, after reloading Asterisk, you should be able to hear the stream on > extension 100. > > Enjoy! > > On Mon, 2005-07-18 at 17:44 -0700, Geoff Karl wrote: > > I am trying to stream an mp3 file from an Asterisk meetme conference. > > > > Ices and Icecast are streaming mp3 files on their own. > > > > I have looked at the info on the wiki and the various readme files. > > > > It appears you need to use Ices2 to work with the confirguration file > > that Asterisk gives you, but it calls Ices. Ices2 appears to only > > stream ogg files and i would like to stream an mp3 file. > > > > Being that my end goal is to stream an mp3 file any ideas on how this > > should be configured. > > > > thanks, > > > > Geoff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Streaming MP3's from Asterisk with Ices
I am trying to stream an mp3 file from an Asterisk meetme conference. Ices and Icecast are streaming mp3 files on their own. I have looked at the info on the wiki and the various readme files. It appears you need to use Ices2 to work with the confirguration file that Asterisk gives you, but it calls Ices. Ices2 appears to only stream ogg files and i would like to stream an mp3 file. Being that my end goal is to stream an mp3 file any ideas on how this should be configured. thanks, Geoff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
