[asterisk-users] Security - What inbound variables can attackers populate or use when calling?
I am setting filters, etc. on variables that attackers can send asterisk when they call (for example when they initially call into asterisk). So far, I am filtering: exten CALLERID(name) CALLERID(num) What other fields or variables would an attacker be able to use in the packets that they send when placing the call to asterisk? Further, I am assuming that in the case that an attacker, first, simply dials in normally and then after reaching voice prompts or other, starts his/her attack, then all I need to filter in that case is exten. Anything else here as well? Thanks!! -- _ -- 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 reuse mysql connection between AGI's
Hi Faheem, You need to build some daemonized application, here FastAGI will help you Regards On Fri, Aug 6, 2010 at 10:54 AM, Faheem wrote: > Hey, Is there any way to share MySQL connection between different agi's. > Actually when call comes to asterisk box it executes various agi scripts > sequentially. Each script checks various values by making a > new MySQL connection and then execute query and then disconnects. > > So, Ideally there should be one connection, and it should be reused between > each agi and when a call is over it should be disconnected. Is there > any mechanism to reuse single MySQL connection between agi scripts? > The agi scripts are written in Perl > > Thanks, > Faheem, M. > > > > > -- > _ > -- 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 > -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- 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] How to reuse mysql connection between AGI's
Hey, Is there any way to share MySQL connection between different agi's.Actually when call comes to asterisk box it executes various agi scripts sequentially. Each script checks various values by making a new MySQL connection and then execute query and then disconnects. So, Ideally there should be one connection, and it should be reused between each agi and when a call is over it should be disconnected. Is there any mechanism to reuse single MySQL connection between agi scripts?The agi scripts are written in Perl Thanks, Faheem, M. -- _ -- 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] Missing Mailboxes on SIP
Suddenly a couple days ago all of our SIP registrations are missing the Mailbox entry. We are using MySQL Add-on for realtime. Anyone have any idea why? Mailbox is still in the mysql tables. -- _ -- 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] Codec Conversion
On 08/06/2010 05:40 AM, Jeff Brower wrote: > Miguel- > >> El 05/08/10 14:50, Tim Nelson escribió: >>> - "michel freiha" wrote: Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice >>> Quality Regards >>> Again, iLBC is poor quality to begin with. You can't take a poor audio >>> sample and make it better by converting it to a codec with better >>> 'resolution'. An audio sample full of robot voice is going to sound >>> like the same robot voice even if you transcode it to a better quality >>> codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs. >>> >>> --Tim >> This just made me remember some comment on the iax.conf sample file... >> >> disallow=lpc10; Icky sound quality... Mr. Roboto. > LPC10 is a very old codec, from early 1980s. LPC10 doesn't do a good job > with pitch detection so it tends to have a > 'robotic' sound. With advent of MELPe, anyone needing bitrates 2400 or less > should not be using LPC10. > > -Jeff MELPe is patent encumbered, so there is still a place for LPC10. LPC10 should sound a lot better than the one in Asterisk. The Asterisk codec is broken. Steve -- _ -- 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 1.6 without DAHDI
Kevin P. Fleming wrote: > On 08/05/2010 03:52 PM, Roderick A. Anderson wrote: >> I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6 >> installed from the asterisk.org and digium.com repositories. >> >> I have Asterisk starting (service asterisk start) but see errors about >> dahdi in /var/log/asterisk/messages. >> >> ... ERROR[25658] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No >> such file or directory >> >> Linux-Vservers don't allow, under normal circumstances, guests to fiddle >> with /dev. I could create all the entries in /dev/dahdi but as far as I >> can determine I have no need of dahdi -- Asterisk 1.6.2, SIP only >> connections, and currently no conference call needs. >> >> Is there a way to stop Asterisk (safe_asterisk) from even trying to load >> dahdi? > > Yes; don't load codec_dahdi.so in Asterisk. Use 'noload' in your > modules.conf file. What packages have you installed from the > asterisk.org and digium.com yum repositories? Thanks Kevin. I made that entry and now there are no more dahdi errors in the log file. Here is the command I used to install Asterisk. yum install asterisk16 asterisk16-configs asterisk16-voicemail And here are some RPM queries # rpm -qa | grep asterisk asterisk-sounds-core-en-gsm-1.4.19-1_centos5 asterisk16-core-1.6.2.10-1_centos5 asterisk16-configs-1.6.2.10-1_centos5 asterisk16-dahdi-1.6.2.10-1_centos5 asterisk16-voicemail-1.6.2.10-1_centos5 asterisk16-doc-1.6.2.10-1_centos5 asterisk16-1.6.2.10-1_centos5 There are also these packages. # rpm -qa | grep dahdi kmod-dahdi-linux-2.3.0.1-1_centos5.2.6.18_194.8.1.el5 dahdi-firmware-tc400m-MR6.12-1_centos5 dahdi-firmware-oct6114-128-1.05.01-1_centos5 dahdi-firmware-2.0.2-1_centos5 asterisk16-dahdi-1.6.2.10-1_centos5 kmod-dahdi-linux-fwload-vpmadt032-2.3.0.1-1_centos5.2.6.18_194.8.1.el5 dahdi-firmware-oct6114-064-1.05.01-1_centos5 dahdi-firmware-hx8-2.06-1_centos5 dahdi-linux-2.3.0.1-1_centos5 There may be more but I haven't taken the time to figure out how to get "yum info" to work like it used to -- showing the source repo instead of just the current (installed) repo -- so some of these are probably not from the asterisk.org or digium.com repos. There may be more that were installed as dependencies. \\||/ Rod -- > -- _ -- 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] Codec Conversion
>> This just made me remember some comment on the iax.conf sample file... >> >> disallow=lpc10; Icky sound quality... Mr. Roboto. >> > LPC10 is a very old codec, from early 1980s. LPC10 doesn't do a good job > with pitch detection so it tends to have a > 'robotic' sound. With advent of MELPe, anyone needing bitrates 2400 or less > should not be using LPC10. > > -Jeff > > OK, on years I have working with asterisk I never have used, tested or even heard that old codec. I was just quoting the funny comment... Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- 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] rolling over Master.csv CDR File
logrotate ~ Andrew "lathama" Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Thu, Aug 5, 2010 at 4:26 PM, Ujjval Karihaloo wrote: > Is there a setting to roll over the Master.csv CDR File in > /var/log/asterisk/cdr-csv, from and ZIP the older file once its gets a > certain size? > > -- > _ > -- 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 > -- _ -- 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] Codec Conversion
Miguel- > El 05/08/10 14:50, Tim Nelson escribió: >> - "michel freiha" wrote: >> > >> > Dear Sir, >> > >> > I tried to convert ilbc to ulaw and get the same result...Bad Voice >> Quality >> > >> > Regards >> > >> >> Again, iLBC is poor quality to begin with. You can't take a poor audio >> sample and make it better by converting it to a codec with better >> 'resolution'. An audio sample full of robot voice is going to sound >> like the same robot voice even if you transcode it to a better quality >> codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs. >> >> --Tim > This just made me remember some comment on the iax.conf sample file... > > disallow=lpc10; Icky sound quality... Mr. Roboto. LPC10 is a very old codec, from early 1980s. LPC10 doesn't do a good job with pitch detection so it tends to have a 'robotic' sound. With advent of MELPe, anyone needing bitrates 2400 or less should not be using LPC10. -Jeff -- _ -- 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 1.6 without DAHDI
On 08/05/2010 03:52 PM, Roderick A. Anderson wrote: > I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6 > installed from the asterisk.org and digium.com repositories. > > I have Asterisk starting (service asterisk start) but see errors about > dahdi in /var/log/asterisk/messages. > > ... ERROR[25658] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No > such file or directory > > Linux-Vservers don't allow, under normal circumstances, guests to fiddle > with /dev. I could create all the entries in /dev/dahdi but as far as I > can determine I have no need of dahdi -- Asterisk 1.6.2, SIP only > connections, and currently no conference call needs. > > Is there a way to stop Asterisk (safe_asterisk) from even trying to load > dahdi? Yes; don't load codec_dahdi.so in Asterisk. Use 'noload' in your modules.conf file. What packages have you installed from the asterisk.org and digium.com yum repositories? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com & www.asterisk.org -- _ -- 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 1.6 without DAHDI
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6 installed from the asterisk.org and digium.com repositories. I have Asterisk starting (service asterisk start) but see errors about dahdi in /var/log/asterisk/messages. ... ERROR[25658] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory Linux-Vservers don't allow, under normal circumstances, guests to fiddle with /dev. I could create all the entries in /dev/dahdi but as far as I can determine I have no need of dahdi -- Asterisk 1.6.2, SIP only connections, and currently no conference call needs. Is there a way to stop Asterisk (safe_asterisk) from even trying to load dahdi? \\||/ Rod -- -- _ -- 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 - Segmentation Fault
Manmohan wrote: > I commented locale.php in defines.php and it perfectly worked well. > Now i am wondering what is this invite participants for, while adding > conference. wherein it asks for first name, lastname, emailaddress & > telephone number.. The 'Invite Others' option is mostly for installs that do not have a consistent e-mail environment, and are using the SERVER mailer. This feature lets the server send invite emails to multiple parties. In my environments we have Exchange and Outlook, so I prefer the CLIENT mailer, and I can manage the invitations in my mail client > Let me brief you how i had done this setup. I had created a SIP trunk > between Cisco Call manager and Asterisk server. And i am using webmeetme > for Audio conferencing. Sounds familiar. I put this package together after wasting too much money and time trying to make an expensive Cisco conferencing solution work. > Other than the invite participants, while the conf call is going on we > get couple of more options, when we click to the current ongoing conference > number. > End call -- To end the conference call Yes > Extend -- I am sure this is to extend the time of the call for which its > scheduled for, but not sure on how much time does it extends by default > OR is there any way to define the customized time on whatever required. 10 minutes is the default. I thought I had made it configurable in lib/defines.php, but no I have it hard coded in conf_add (to be fixed in the next release now). You can search for +600 and change it to any value you like. > Invite-- When i click this button it asks me telephone number. I assume this > is any number which asterisk server can reach as per the dialplan configured > in extension.conf in /etc/asterisk.. Though this invite button looks pretty > much interesting to use but whenever i enter any phone number it says > "System error" not sure if am understand this wrongly. You understand it correctly, but the default settings are likely not working. Check out the section 'Outcall defaults' in lib/defines.php. It is likely you need to change the OUT_CONTEXT at a minimum. Dan -- _ -- 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] Codec Conversion
El 05/08/10 14:50, Tim Nelson escribió: - "michel freiha" wrote: > > Dear Sir, > > I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality > > Regards > Again, iLBC is poor quality to begin with. You can't take a poor audio sample and make it better by converting it to a codec with better 'resolution'. An audio sample full of robot voice is going to sound like the same robot voice even if you transcode it to a better quality codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs. --Tim This just made me remember some comment on the iax.conf sample file... disallow=lpc10; Icky sound quality... Mr. Roboto. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- 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] Codec Conversion
Michel- > I tried to convert ilbc to ulaw and get the same > result...Bad Voice Quality I think you have to be more specific when you say "bad voice quality". Like what? Worse than a cellphone call? Gaps of audio missing? Robotic or "cyborg" sound? Static? A background tone or buzzing? iLBC isn't any worse voice quality than other LBR codecs (GSM-AMR, EVRC, etc). If you want land-line quality and what you're hearing is cellphone quality, then you're asking too much. Otherwise, suggest to be specific and detailed in describing your problem. -Jeff > On Thu, Aug 5, 2010 at 4:13 PM, Tim Nelson wrote: > >> - "michel freiha" wrote: >> > >> > Dear All, >> > >> > i would like to ask please if someone tried to make a codec conversion >> from ilbc to g729, because i did that but the voice quality was too bad and >> a lot of disconnection.. >> > >> > Can i get your feedback regarding this issue please? >> > >> > regards >> >> I can't comment on your 'disconnection' as you don't say if that means the >> call is disconnected or you're getting stuttered audio. Regardless, iLBC has >> one of the lowest bitrates of the available codecs and as such the voice >> quality is not spectacular to begin with. Take 'not so good' audio and try >> to convert it to another audio format, and the deficiencies can be >> exacerbated. >> >> --Tim -- _ -- 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] rolling over Master.csv CDR File
Is there a setting to roll over the Master.csv CDR File in /var/log/asterisk/cdr-csv, from and ZIP the older file once its gets a certain size? -- _ -- 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] COnfig File question
1.7 for ASteriskNOw I will investigate..Thx for the ideas! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Thursday, August 05, 2010 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] COnfig File question On Thu, Aug 5, 2010 at 11:14 AM, Felipe Figueiredo mailto:felipe.figueired...@gmail.com>> wrote: Yes. Unless you use "make samples" while compiling the new Asterisk, you won't lose your confg files. I'm afraid there's no 1.7 version of Asterisk. [cid:image001.png@01CB34AA.02451E80] But there is a 1.7 version of AsteriskNow, which is what he was asking about. I'm not sure how you'd go about updating that though. -- Thanks, --Warren Selby http://www.selbytech.com <>-- _ -- 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] CDR report
Hi all! Are someone using a CDR report? I have an Asterisk 1.6 running perfect but I need a web based report of CDRs. Nothing big, only the basic. Have anybody a how-to or a link? Thanks in advance!! -- Atenciosamente, --- Dario Quiroz Analista de Suporte (71) 9275-9080 darioqui...@gmail.com --- -- _ -- 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 - Segmentation Fault
Hi Dan,
I commented locale.php in defines.php and it perfectly worked well.
Now i am wondering what is this invite participants for, while adding
conference. wherein it asks for first name, lastname, emailaddress &
telephone number..
Let me brief you how i had done this setup. I had created a SIP trunk
between Cisco Call manager and Asterisk server. And i am using webmeetme for
Audio conferencing.
Other than the invite participants, while the conf call is going on we get
couple of more options, when we click to the current ongoing conference
number.
End call -- To end the conference call
Extend -- I am sure this is to extend the time of the call for which its
scheduled for, but not sure on how much time does it extends by default OR
is there any way to define the customized time on whatever required.
Invite-- When i click this button it asks me telephone number. I assume this
is any number which asterisk server can reach as per the dialplan configured
in extension.conf in /etc/asterisk.. Though this invite button looks pretty
much interesting to use but whenever i enter any phone number it says
"System error" not sure if am understand this wrongly.
Please correct me if i am wrong.
--Manmohan Singh
On Wed, Aug 4, 2010 at 9:14 PM, Dan Austin wrote:
> Manmohan wrote:
> > I had tried the new version of webmeetme i.e., 4.0.2
> > The recording works very well.
> Great!
>
> > I see following php errors whenever i try to add in conference.
>
> > [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice:
> > Undefined variable: order in /var/www/html/web-meetme/meetme_control.php
> > on line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4
> You can ignore the Notices. They are fairly harmless, and only mean that
> variable is not set by the code or being passed in on the URL. You can
> turn off notices in /etc/php.ini if they bother you.
>
> > Also the Reports link doesnt display anything and in httpd error logs it
> gives me following php errors:
> > [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning:
> > include(locale.php) [function.include]:
> > failed to open stream: No such file or directory in
> /var/www/html/web-meetme/lib/defines.php
> > on line 3, referer: http://10.1.1.30/web-meetme/daily.php?
>
> In lib/defines.php, either comment out the 3rd line or add ../ before
> locale.php-
>include("../locale.php");
>
> But that is not likely why you do not get the reports. The most likely
> cause is
> A PHP notice is being thrown while the GD code is rendering the graph,
> resulting in
> a corrupt image which your browser cannot display.
>
> Check these settings /etc/php.ini-
> error_reporting = E_ALL
> display_errors = Off
>
> Dan
>
> --
> _
> -- 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
>
--
Thanks & Regards
Manmohan Singh Jandu
--
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Re: [asterisk-users] Codec Conversion
- "michel freiha" wrote: > > Dear Sir, > > I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality > > Regards > Again, iLBC is poor quality to begin with. You can't take a poor audio sample and make it better by converting it to a codec with better 'resolution'. An audio sample full of robot voice is going to sound like the same robot voice even if you transcode it to a better quality codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs. --Tim -- _ -- 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] Codec Conversion
Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality Regards On Thu, Aug 5, 2010 at 4:13 PM, Tim Nelson wrote: > - "michel freiha" wrote: > > > > Dear All, > > > > i would like to ask please if someone tried to make a codec conversion > from ilbc to g729, because i did that but the voice quality was too bad and > a lot of disconnection.. > > > > Can i get your feedback regarding this issue please? > > > > regards > > I can't comment on your 'disconnection' as you don't say if that means the > call is disconnected or you're getting stuttered audio. Regardless, iLBC has > one of the lowest bitrates of the available codecs and as such the voice > quality is not spectacular to begin with. Take 'not so good' audio and try > to convert it to another audio format, and the deficiencies can be > exacerbated. > > --Tim > > -- > _ > -- 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 > -- _ -- 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] AMI Command
Danny Nicholas wrote:
>> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
> Lesher
>> Subject: Re: [asterisk-users] AMI Command
>
>> Actually, what you probably want is the CoreShowChannels command.
>
>> Tilghman Lesher
>
> To second this; core show channels doesn't require the maintenance/overhead
> that core show hints does. If you add new lines (and we mostly all do) you
> have to add a new hint to the dialplan to keep up with it using core show
> hints. If you change releases, your hints might do funny (not ha ha)
> things.
>
>
It depends on what asterisk version you are running. I am running 1.6
and just use a single line like the following to automatically add any
hints for extensions that exist.
exten => _9XX,hint,SIP/${EXTEN}
--
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Re: [asterisk-users] COnfig File question
On Thu, Aug 5, 2010 at 11:14 AM, Felipe Figueiredo < felipe.figueired...@gmail.com> wrote: > Yes. Unless you use "make samples" while compiling the new Asterisk, you > won't lose your confg files. > I'm afraid there's no 1.7 version of Asterisk. [?] > > > But there is a 1.7 version of AsteriskNow, which is what he was asking about. I'm not sure how you'd go about updating that though. -- Thanks, --Warren Selby http://www.selbytech.com <<325.png>>-- _ -- 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] COnfig File question
>From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo >Subject: Re: [asterisk-users] COnfig File question >Hi All: > If we upgrade asteriskNow from 1.4.18 to 1.7.0; just want to make sure all the config file functionality will reamin same >That is - everything in /etc/asterisk will still work the same way. >Users.conf >Provider.conf >Extensions.conf >Sip.conf >Etc. >Thx in advance. >Any answers would be appreciated The .conf files themselves won't change unless you do a "make samples" ; however , the handling of some items may. -- _ -- 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] COnfig File question
Yes. Unless you use "make samples" while compiling the new Asterisk, you won't lose your confg files. I'm afraid there's no 1.7 version of Asterisk. [?] On Thu, Aug 5, 2010 at 1:01 PM, Ujjval Karihaloo wrote: > Any answers would be appreciated > > > > Thx > > UK > > > > > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ujjval Karihaloo > *Sent:* Thursday, July 29, 2010 4:31 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] COnfig File question > > > > Hi All: > > > > If we upgrade asteriskNow from 1.4.18 to 1.7.0; just want to make sure > all the config file functionality will reamin same > > > > That is – everything in /etc/asterisk will still work the same way. > > > > Users.conf > > Provider.conf > > Extensions.conf > > Sip.conf > > > > Etc… > > > > Thx in advance. > > > > -- > _ > -- 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 > <<325.png>>-- _ -- 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] COnfig File question
Any answers would be appreciated Thx UK From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo Sent: Thursday, July 29, 2010 4:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] COnfig File question Hi All: If we upgrade asteriskNow from 1.4.18 to 1.7.0; just want to make sure all the config file functionality will reamin same That is - everything in /etc/asterisk will still work the same way. Users.conf Provider.conf Extensions.conf Sip.conf Etc... Thx in advance. -- _ -- 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] AMI Command
>From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher >Subject: Re: [asterisk-users] AMI Command >Actually, what you probably want is the CoreShowChannels command. >Tilghman Lesher To second this; core show channels doesn't require the maintenance/overhead that core show hints does. If you add new lines (and we mostly all do) you have to add a new hint to the dialplan to keep up with it using core show hints. If you change releases, your hints might do funny (not ha ha) things. -- _ -- 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] AMI Command
On Thursday 05 August 2010 06:05:48 Ron wrote: > Thank you. i think i would go for this solution. > > On 8/5/10 4:53 PM, Gareth Blades wrote: > > Ron wrote: > >> Hi, > >> > >> Is there a way to check on AMI if a user is currently engage on the > >> phone? i would like to display on my portal whether a user is calling or > >> not. > >> > >> thank you > >> > >> regards > >> Ron > > > > You could get it to run a command and do 'core show hints' and parse the > > result. You will need to define hints for each extension though but you > > might have already done this as its required to get busy lamps working > > on most phones. Actually, what you probably want is the CoreShowChannels command. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] Codec Conversion
- "michel freiha" wrote: > > Dear All, > > i would like to ask please if someone tried to make a codec conversion from > ilbc to g729, because i did that but the voice quality was too bad and a lot > of disconnection.. > > Can i get your feedback regarding this issue please? > > regards I can't comment on your 'disconnection' as you don't say if that means the call is disconnected or you're getting stuttered audio. Regardless, iLBC has one of the lowest bitrates of the available codecs and as such the voice quality is not spectacular to begin with. Take 'not so good' audio and try to convert it to another audio format, and the deficiencies can be exacerbated. --Tim -- _ -- 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] Identify remote prompts: Partial audio matching?
On Wed, Aug 4, 2010 at 10:12 PM, Philipp von Klitzing wrote: > Ok, here's the challenge: > > I would like to be able to find, match - and then react - upon prompts > that are presented by the outbound/remote side of a call. Think mobile > phone and "This user is temporarily unavailable". > > Collecting a limited number of known prompt snippets should not be a > problem, but how would you then detect their presence in a longer > recording (or live audio stream)? > > Recently there was an at least slightly related posting on this list, if > I recall that correctly, but I have simply not been able to turn this up. > > Philipp > > P.S.: This is all about audio analysis, not about cause codes. Exact match: http://github.com/Motiejus/SoundPatty Regards Motiejus Jakštys -- _ -- 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] AMI Command
Thank you. i think i would go for this solution. On 8/5/10 4:53 PM, Gareth Blades wrote: > Ron wrote: >> Hi, >> >> Is there a way to check on AMI if a user is currently engage on the >> phone? i would like to display on my portal whether a user is calling or >> not. >> >> thank you >> >> regards >> Ron >> > You could get it to run a command and do 'core show hints' and parse the > result. You will need to define hints for each extension though but you > might have already done this as its required to get busy lamps working > on most phones. > -- _ -- 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] Codec negotiation : expecting G726, getting G711a (alaw)
> Only when I configure my Grandstream to use only G726 (I have 8 > choices), I see that the g726-codec is used. > When I configure 7 x g726 and 1 x alaw, then again alaw is used ! > > Is it normal that Asterisk has such a great preference for alaw ?! The > moment the peer suggests codec alaw (even if it is last choice), alaw is > chosen by Asterisk for the communication. Please look at the first part of my last message (order of codecs in the [general] section) and apply changes there, followed by a "sip reload". Philipp -- _ -- 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] Can ChanIsAvail return status from sip uri using router ip
Hi! > Although my previous posts in this forum have not received satisfying > answers, here is another question from me. You might want to consider to reqest a refund. ;-> > my question is can i use ChanIsAvail function to get the status of a user > in the format SPI/user-id if i provide user in sip uri like this > > ChanIsAvail(SIP/u...@153.18.x.x:5062) > > calling user with this sip uri works fine. That will not work, you need a known peer combined with qualify= for a useful response. You can read details on the Wiki: http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail Philipp -- _ -- 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] Codec Conversion
Dear All, i would like to ask please if someone tried to make a codec conversion from ilbc to g729, because i did that but the voice quality was too bad and a lot of disconnection.. Can i get your feedback regarding this issue please? regards -- _ -- 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] AMI Command
Ron wrote: > Hi, > > Is there a way to check on AMI if a user is currently engage on the > phone? i would like to display on my portal whether a user is calling or > not. > > thank you > > regards > Ron > You could get it to run a command and do 'core show hints' and parse the result. You will need to define hints for each extension though but you might have already done this as its required to get busy lamps working on most phones. -- _ -- 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] AMI Command
On Thu, Aug 5, 2010 at 11:28 AM, Ron wrote: > Hi, > > Is there a way to check on AMI if a user is currently engage on the > phone? i would like to display on my portal whether a user is calling or > not. > # Asterisk Manager API Action CoreShowChannels: List currently active channels (Priv: system,reporting,all) From: http://www.voip-info.org/wiki/view/Asterisk+manager+API And maybe this (check output): http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+SIPshowPeer Regards Motiejus Jakštys -- _ -- 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] AMI Command
Hi, Is there a way to check on AMI if a user is currently engage on the phone? i would like to display on my portal whether a user is calling or not. thank you regards Ron -- _ -- 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] Can ChanIsAvail return status from sip uri using router ip
hello, Although my previous posts in this forum have not received satisfying answers, here is another question from me. my question is can i use ChanIsAvail function to get the status of a user in the format SPI/user-id if i provide user in sip uri like this ChanIsAvail(SIP/u...@153.18.x.x:5062) calling user with this sip uri works fine. I once tried but status returned was "unknow host 153.18.x.x". what is wrong here? thanks Nasir Javaid -- _ -- 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] Codec negotiation : expecting G726, getting G711a (alaw)
On 08/03/2010 04:21 PM, Philipp von Klitzing wrote: > Also: > > There are at least two implementations of the g726 codec, i.e. g726 and > g726aal2. For this also look at the g726nonstandard setting in sip.conf. > It is quite possible that your problem is here. > I have the following setting in sip.conf : g726nonstandard = no ; If the peer negotiates G726-32 audio, use AAL2 packing ; order instead of RFC3551 packing order (this is required ; for Sipura and Grandstream ATAs, among others). This is ; contrary to the RFC3551 specification, the peer _should_ ; be negotiating AAL2-G726-32 instead (so it uses RFC3551) > For quick testing to see if the codec works at all: Configure your phones > to do g726 only (so no alaw/ualaw at all). > Only when I configure my Grandstream to use only G726 (I have 8 choices), I see that the g726-codec is used. When I configure 7 x g726 and 1 x alaw, then again alaw is used ! Is it normal that Asterisk has such a great preference for alaw ?! The moment the peer suggests codec alaw (even if it is last choice), alaw is chosen by Asterisk for the communication. Kind regards, Jonas. -- _ -- 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
