[asterisk-users] VOIP at BerkeleyTIP-Global meeting on Sunday April 18 12N-3P, & April 27
Come discuss VOIP. :) Join via VOIP or come to Berkeley http://sites.google.com/site/berkeleytip/voice-voip-conferencing FSCafe at Moffitt at UCBerkeley, opens 1pm, but can connect from outside at 12N. Hot topics: Ubuntu 10.04, Free Culuture, VOIP, Set up the web server & mail list & asterisk/freeswitch on the BTIP box with Ubuntu 10.04? Tues April 27 5-6P VOIP online meeting also. http://sites.google.com/site/berkeleytip/ Join the mail list, tell us what you're interested in. http://sites.google.com/site/berkeleytip/mailing-lists -- _ -- 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] Changing storm-prevention behaviour in logger.conf
On Sat, Apr 17, 2010 at 11:14:23PM +0200, Remco Bressers wrote: > Dear List, > > According to https://issues.asterisk.org/view.php?id=14905 there is a storm > prevention mechanism in newer Asterisks. If i look in my logfile, i see : > > [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Registration from '"" > ' failed for 'xx.xx.xx.xx' - Wrong password > [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Last message repeated 3 > times > > This IS a good thing to do, but i want to disable this behaviour. We are > using fail2ban to ban scripts and people from the Asterisk system. On > version 1.4.23 this worked fine, but now this mechanism is in place, i > cannot use fail2ban anymore. > > Is there any option to disable this behaviour, or even better, add it to > logger.conf so anybody can decide what to do? I just want all logging and it > seems impossible now. > Maybe a patch on the source? If you use a newer version of rsyslogd to do your logging, there is a global configuration directive: $RepeatedMsgReduction off that will do what you are asking. The issue #14905 patch you mention is not in 1.6.2.x. -- Barry -- _ -- 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] Changing storm-prevention behaviour in logger.conf
Tilghman Lesher wrote: > On Saturday 17 April 2010 16:14:23 Remco Bressers wrote: > >> Dear List, >> >> According to https://issues.asterisk.org/view.php?id=14905 there is a storm >> prevention mechanism in newer Asterisks. If i look in my logfile, i see : >> >> [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Registration from '"" >> ' failed for 'xx.xx.xx.xx' - Wrong password >> [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Last message repeated 3 >> times >> >> This IS a good thing to do, but i want to disable this behaviour. We are >> using fail2ban to ban scripts and people from the Asterisk system. On >> version 1.4.23 this worked fine, but now this mechanism is in place, i >> cannot use fail2ban anymore. >> >> Is there any option to disable this behaviour, or even better, add it to >> logger.conf so anybody can decide what to do? I just want all logging and >> it seems impossible now. Maybe a patch on the source? >> > > That's not Asterisk doing that. That's your system logger. AFAIK, there's no > way to turn that off, as it's a defense mechanism against an attacker filling > your disks, causing lost messages and possible crashes (on some platforms). > > If running syslog-ng, check syslog-ng.conf and the summary option. Setting summary to 0 turns off that behavior. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] AGI, FASTAGI or Windows Voice Server
FastAGI will do everything you need. You just need to code the Windows
AGI. BTW, #6 is incorrect if you code your AGI properly
On Sat, Apr 17, 2010 at 1:43 PM, Edwin Quijada
wrote:
> I did using FTP. This is the problem and the solution that I did but
> doesnt work
>
> 1-When the call in to asterisk I play one prompt if this prompt doesnt
> exist I create it
> 2-In windows I have a program listen on a port waiting for request from
> asterisk
> 3- I sent by this socket the text and name for the file
> 4- In windows server create the file and convert to 8khz using sox
> 5- From windows try to copy this file to asterisk using FTP protocol
> 6- There is no syncronize between AGI script and copy to FTP
> 7- I did a loop to wait for copy of file to my sound directory but it never
> happenned because it couldnt create the file
> 8- if I put off the loop while (!existfile) { } so it can create the file
> in windows I really dont know why this behaviour
>
> My plan was so simple
> A server waiting request for asterisk and the copy this file to asterisk to
> play it
> but doesnt work, for this reason i am trying to do everything using FastAgi
> in a windows server.
>
> *---*
> *-Edwin Quijada
> *-Developer DataBase
> *-JQ Microsistemas
> *-Soporte PostgreSQL
> *-www.jqmicrosistemas.com
> *-809-849-8087
> *---*
>
>
>
>
>
> --
> Date: Sat, 17 Apr 2010 13:23:22 -0400
> From: stot...@first-notification.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server
>
>
>
>
> On Fri, Apr 16, 2010 at 5:58 PM, Edwin Quijada > wrote:
>
>
>
>
> Why don’t you use sox to transform the windows audio file into the
> asterisk format – I do this with pretty good results.
>
> I did. But my problem is not conversion my problem is that I dont know how
> play the file from windows server or copy this to asterisk without my AGI
> continue and desyncronyze it.
>
> Can you explain me exactly what did you do /?
>
> Do you have something like this using AGI ?
>
> I use sox with good results too in windows. The problem is when create the
> file and convert it , how send to asterisk
>
>
> Edwin Jaws
>
>
>
> If you just need to transfer a file to a linux box, there are plenty of
> ways. FTP, SFTP, TFTP, Samba.
>
> Thanks,
> Steve T
>
> --
>
> --
> _
> -- 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
>
--
_
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asterisk-users mailing list
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Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server
On Sat, Apr 17, 2010 at 1:48 PM, Edwin Quijada wrote: > > > Just a shot in the dark, have you tried ExternalIVR? It was originally > developed for Unwired Buyer (Ebay), I believe Digium and Unwired Buyer > teamed up on this one. > > This option NO. > > > Another option would be FastAGI to your windows server. You write an app > for the windows box that interacts with the AT&T application and then pipe > the audio back to your asterisk box somehow. First thought is app_bridge or > meetme. > > This is the idea just I dont know how to do. You can give any direction to > start first. I am looking for information about app_bridge > > *---* > *-Edwin Quijada > *-Developer DataBase king > > *-JQ Microsistemas > *-Soporte PostgreSQL > *-www.jqmicrosistemas.com > *-809-849-8087 > *---* > "This option NO." is quite a rude reply when someone is giving you ideas for free. Maybe you can say why it is not an option but your response was rude and makes me not want to help you anymore. I can tell you are an ESOL by the way you write, so maybe you don't understand the best way to communicate. Also, if you tried FTP, then did you not post that first. What else have you tried? Why waste people's time when you have tried things that didn't work but don't convey them? Did you try Samba? As far as app_bridge, there is plenty of documentation, let me waste more of my time.. http://tinyurl.com/y73mp9s Sounds like you should pay for the Linux version or paid Asterisk support. I really appreciate helping you, thanks, Steve Totaro -- _ -- 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] Changing storm-prevention behaviour in logger.conf
On Saturday 17 April 2010 16:14:23 Remco Bressers wrote: > Dear List, > > According to https://issues.asterisk.org/view.php?id=14905 there is a storm > prevention mechanism in newer Asterisks. If i look in my logfile, i see : > > [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Registration from '"" > ' failed for 'xx.xx.xx.xx' - Wrong password > [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Last message repeated 3 > times > > This IS a good thing to do, but i want to disable this behaviour. We are > using fail2ban to ban scripts and people from the Asterisk system. On > version 1.4.23 this worked fine, but now this mechanism is in place, i > cannot use fail2ban anymore. > > Is there any option to disable this behaviour, or even better, add it to > logger.conf so anybody can decide what to do? I just want all logging and > it seems impossible now. Maybe a patch on the source? That's not Asterisk doing that. That's your system logger. AFAIK, there's no way to turn that off, as it's a defense mechanism against an attacker filling your disks, causing lost messages and possible crashes (on some platforms). -- 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] AGI, FASTAGI or Windows Voice Server
On Sat, Apr 17, 2010 at 1:48 PM, Edwin Quijada wrote: > > > Just a shot in the dark, have you tried ExternalIVR? It was originally > developed for Unwired Buyer (Ebay), I believe Digium and Unwired Buyer > teamed up on this one. > > This option NO. > > > Another option would be FastAGI to your windows server. You write an app > for the windows box that interacts with the AT&T application and then pipe > the audio back to your asterisk box somehow. First thought is app_bridge or > meetme. > > This is the idea just I dont know how to do. You can give any direction to > start first. I am looking for information about app_bridge > > *---* > *-Edwin Quijada > *-Developer DataBase king > > *-JQ Microsistemas > *-Soporte PostgreSQL > *-www.jqmicrosistemas.com > *-809-849-8087 > *---* > "This option NO." is quite a rude reply when someone is giving you ideas for free. Maybe you can say why it is not an option but your response was rude and makes me not want to help you anymore. I can tell you are an ESOL by the way you write, so maybe you don't understand the best way to communicate. Also, if you tried FTP, then did you not post that first. What else have you tried? Why waste people's time when you have tried things that didn't work but don't convey them? Did you try Samba? As far as app_bridge, there is plenty of documentation, let me waste more of my time.. http://tinyurl.com/y73mp9s Sounds like you should pay for the Linux version or paid Asterisk support. I really appreciate helping you, thanks, Steve Totaro -- _ -- 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 and DTMF
Hi, I have 3 ISDN lines using the digium B410P card. Incoming and outgoing call are working. I use the following version : * libpri-1.4.10.2 * dahdi-2.2.1.1 * asterisk-1.6.2.6 On incoming calls, DTMF is not working, i can't see any logs. Here are the main configuration files : dahdi-channels.conf ;; Span 2: B4/0/1 "B4XXP (PCI) Card 0 Span 1" signalling=bri_cpe callerid=asreceived group=5 context=trunk_5_0 language=fr channel => 5-6 ;; Span 3: B4/0/2 "B4XXP (PCI) Card 0 Span 2" signalling=bri_cpe callerid=asreceived group=6 context=trunk_7_0 language=fr channel => 8-9 ;; Span 4: B4/0/3 "B4XXP (PCI) Card 0 Span 3" signalling=bri_cpe callerid=asreceived group=7 context=trunk_9_0 language=fr channel => 11-12 /etc/dahdi/system.conf # Span 2: B4/0/1 "B4XXP (PCI) Card 0 Span 1" span=2,1,0,ccs,ami # termtype: te bchan=5-6 hardhdlc=7 echocanceller=mg2,5-6 # Span 3: B4/0/2 "B4XXP (PCI) Card 0 Span 2" span=3,2,0,ccs,ami # termtype: te bchan=8-9 hardhdlc=10 echocanceller=mg2,8-9 # Span 4: B4/0/3 "B4XXP (PCI) Card 0 Span 3" span=4,3,0,ccs,ami # termtype: te bchan=11-12 hardhdlc=13 echocanceller=mg2,11-12 # Span 5: B4/0/4 "B4XXP (PCI) Card 0 Span 4" span=5,4,0,ccs,ami # termtype: te bchan=14-15 hardhdlc=16 echocanceller=mg2,14-15 Thank you ! Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.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] Changing storm-prevention behaviour in logger.conf
Dear List, According to https://issues.asterisk.org/view.php?id=14905 there is a storm prevention mechanism in newer Asterisks. If i look in my logfile, i see : [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Registration from '"" ' failed for 'xx.xx.xx.xx' - Wrong password [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Last message repeated 3 times This IS a good thing to do, but i want to disable this behaviour. We are using fail2ban to ban scripts and people from the Asterisk system. On version 1.4.23 this worked fine, but now this mechanism is in place, i cannot use fail2ban anymore. Is there any option to disable this behaviour, or even better, add it to logger.conf so anybody can decide what to do? I just want all logging and it seems impossible now. Maybe a patch on the source? Regards, Remco Bressers Signet B.V. -- _ -- 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 changes not reflected realtime
On Sat, Apr 17, 2010 at 11:14 AM, Jonas Kellens wrote: > > Is rtcachefriends=yes a wrong setting ?? > > > No, not if you want caching enabled. I enable sip realtime caching on all > of my systems. > > > What if I do not enable caching ? What would be the effect on my realtime > configuration with sip_buddies in my mysql-DB ? At the bottom of the page it talks a little about caching: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip I know that "sip show peers" doesn't work, and I believe that qualify does not work without caching (but I haven't tested that). I enable caching because I don't change the names of sip_accounts that frequently, and why have Asterisk hit the database constantly if you aren't changing the information? Asterisk will then save all of the results in RAM, and only do a look-up for an unknown account. If you have a web interface for updating information you could always use AMI to issue the prune/reload after committing the changes. -Jonathan -- _ -- 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 changes not reflected realtime
Is rtcachefriends=yes a wrong setting ?? No, not if you want caching enabled. I enable sip realtime caching on all of my systems. What if I do not enable caching ? What would be the effect on my realtime configuration with sip_buddies in my mysql-DB ? 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
Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server
Just a shot in the dark, have you tried ExternalIVR? It was originally developed for Unwired Buyer (Ebay), I believe Digium and Unwired Buyer teamed up on this one. This option NO. Another option would be FastAGI to your windows server. You write an app for the windows box that interacts with the AT&T application and then pipe the audio back to your asterisk box somehow. First thought is app_bridge or meetme. This is the idea just I dont know how to do. You can give any direction to start first. I am looking for information about app_bridge*---* *-Edwin Quijada *-Developer DataBase king *-JQ Microsistemas *-Soporte PostgreSQL*-www.jqmicrosistemas.com*-809-849-8087*---* _ -- _ -- 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] AGI, FASTAGI or Windows Voice Server
I did using FTP. This is the problem and the solution that I did but doesnt work
1-When the call in to asterisk I play one prompt if this prompt doesnt exist I
create it2-In windows I have a program listen on a port waiting for request
from asterisk 3- I sent by this socket the text and name for the file4- In
windows server create the file and convert to 8khz using sox5- From windows try
to copy this file to asterisk using FTP protocol 6- There is no syncronize
between AGI script and copy to FTP 7- I did a loop to wait for copy of file to
my sound directory but it never happenned because it couldnt create the file 8-
if I put off the loop while (!existfile) { } so it can create the file in
windows I really dont know why this behaviour
My plan was so simple A server waiting request for asterisk and the copy this
file to asterisk to play itbut doesnt work, for this reason i am trying to do
everything using FastAgi in a windows server.
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-Soporte PostgreSQL
*-www.jqmicrosistemas.com
*-809-849-8087
*---*
Date: Sat, 17 Apr 2010 13:23:22 -0400
From: stot...@first-notification.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server
On Fri, Apr 16, 2010 at 5:58 PM, Edwin Quijada
wrote:
Why don’t you use sox to transform the windows audio file into the asterisk
format – I do this with pretty good results.
I did. But my problem is not conversion my problem is that I dont know how play
the file from windows server or copy this to asterisk without my AGI continue
and desyncronyze it.
Can you explain me exactly what did you do /?
Do you have something like this using AGI ?
I use sox with good results too in windows. The problem is when create the file
and convert it , how send to asterisk
Edwin Jaws
If you just need to transfer a file to a linux box, there are plenty of ways.
FTP, SFTP, TFTP, Samba.
Thanks,
Steve T
_
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_
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Re: [asterisk-users] QUICKLY - What is the command to change .wmv to Asterisk compatible format using ffmpeg? or any other tool?
Thanks but there are tons of unncessary information that come up and nothing specific to asterisk with that type of search. I have already run through those. Anyhow, I won't want to convert anymore but I am wondering if echo() would work and would echo my cam pictuer back to me. I am trying the following: exten => 20,1,playback(beep)exten => 20,n,Record(/tmp/myvideo:wav)exten => 20,n,Hangup exten => 21,1,Answer()exten => 21,n,Background(/tmp/myvideo) * * Problem is that eyeBeam shows my camera on and my picture but on top says, "waiting for remote video" for ever. So, it seems asterisk doesn't send picture back to me. I have videosupport=yes in sip.conf [general] and I have allow=h263 in sip.conf How can I go about debugging the video transmission? Thanks On Sat, Apr 17, 2010 at 1:07 PM, Steve Totaro wrote: > > > On Sat, Apr 17, 2010 at 12:31 PM, bruce bruce wrote: > >> Hi Guys, >> >> I want to test my first video transmission call from Asterisk 1.6 to >> X-lite softphone. I set videosupport=yes in SIP [general] and I have place a >> .wmv file in /var/lib/asterisk/sounds/en and did a chown asterisk.asterisk >> on it. I guess I have to use Playback command for the file and before that I >> have to convert the file to h.263??!! >> >> I just installed ffmpeg (the conversion tool) but does anyone have a quick >> command to change .wmv file to h.263 or whatever the Asterisk compatible >> video format is? >> >> Thanks a lot >> >> http://tinyurl.com/yyr6tvx > > -- > _ > -- 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] AGI, FASTAGI or Windows Voice Server
On Fri, Apr 16, 2010 at 4:59 PM, Edwin Quijada wrote: > Hello! > I have developed an IVR using AGI and so far it works great. I'm using > Cepstral voices, but now want to use the voices from AT & T that are on a > Windows server to be heard best. With cepstral what I do is to generate > audio files from shipping and this text I reproduce this method it has > worked very well. > > > > Now, try to do the same by creating the audio file in windows with the > voices of AT & T, the problem is that there is no way to synchronize the > generation of the audio file and step Asterisk to be played, so it occurred > to me to use FastAGI to generate all Windows and play in the same window > the audio file generated. > > We buy Linux licenses for the voices but they are very expensive and > already bought windows for another project. How do you think would be the > best option? > > > > If you have another idea, please Tell me because I'm getting crazy with > this and can not solve. > > TIA > > Edwin > > Just a shot in the dark, have you tried ExternalIVR? It was originally developed for Unwired Buyer (Ebay), I believe Digium and Unwired Buyer teamed up on this one. Another option would be FastAGI to your windows server. You write an app for the windows box that interacts with the AT&T application and then pipe the audio back to your asterisk box somehow. First thought is app_bridge or meetme. Thanks, Steve T -- _ -- 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] AGI, FASTAGI or Windows Voice Server
On Fri, Apr 16, 2010 at 5:58 PM, Edwin Quijada wrote: > > > > Why don’t you use sox to transform the windows audio file into the > asterisk format – I do this with pretty good results. > > > I did. But my problem is not conversion my problem is that I dont know how > play the file from windows server or copy this to asterisk without my AGI > continue and desyncronyze it. > > Can you explain me exactly what did you do /? > > Do you have something like this using AGI ? > > I use sox with good results too in windows. The problem is when create the > file and convert it , how send to asterisk > > > Edwin Jaws > > > If you just need to transfer a file to a linux box, there are plenty of ways. FTP, SFTP, TFTP, Samba. Thanks, Steve T -- _ -- 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] DIALSTATUS variable and qualify=no
Hi! > > could anybody tell me if the info below is still correct: > > > > Note: In order to obtain useful DIALSTATUS information when dialing a > > peer you will need to have qualify=yes in that peer's definition (e.g. > > in sip.conf or iax.conf). > > http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS > > That's not correct. DIALSTATUS will be set whether or not you've got > qualify=yes in the peer definition. I think the emphasis of the quote above was on "useful". The answer does not 100% fit the question. :-) 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] QUICKLY - What is the command to change .wmv to Asterisk compatible format using ffmpeg? or any other tool?
On Sat, Apr 17, 2010 at 12:31 PM, bruce bruce wrote: > Hi Guys, > > I want to test my first video transmission call from Asterisk 1.6 to X-lite > softphone. I set videosupport=yes in SIP [general] and I have place a .wmv > file in /var/lib/asterisk/sounds/en and did a chown asterisk.asterisk on it. > I guess I have to use Playback command for the file and before that I have > to convert the file to h.263??!! > > I just installed ffmpeg (the conversion tool) but does anyone have a quick > command to change .wmv file to h.263 or whatever the Asterisk compatible > video format is? > > Thanks a lot > > http://tinyurl.com/yyr6tvx -- _ -- 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] QUICKLY - What is the command to change .wmv to Asterisk compatible format using ffmpeg? or any other tool?
Hi Guys, I want to test my first video transmission call from Asterisk 1.6 to X-lite softphone. I set videosupport=yes in SIP [general] and I have place a .wmv file in /var/lib/asterisk/sounds/en and did a chown asterisk.asterisk on it. I guess I have to use Playback command for the file and before that I have to convert the file to h.263??!! I just installed ffmpeg (the conversion tool) but does anyone have a quick command to change .wmv file to h.263 or whatever the Asterisk compatible video format is? Thanks a lot -- _ -- 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 changes not reflected realtime
On Sat, Apr 17, 2010 at 4:42 AM, Jonas Kellens wrote: > Do I need to 'sip prune realtime all' after every change ?? If you change a sip peer and you have caching enabled, then you need to prune that peer for the change to take effect. On 1.6.1 I issue the following: sip prune realtime sip show load That will only clear the caching for and not all of the peers. The load statement re-caches the peer immediately. I haven't tried this on 1.4, so I don't know if those options exist or not. > Is rtcachefriends=yes a wrong setting ?? No, not if you want caching enabled. I enable sip realtime caching on all of my systems. -Jonathan -- _ -- 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] DIALSTATUS variable and qualify=no
On Sat, 2010-04-17 at 17:38 +0400, Rustam Kovhaev wrote: > could anybody tell me if the info below is still correct: > > Note: In order to obtain useful DIALSTATUS information when dialing a > peer you will need to have qualify=yes in that peer's definition (e.g. > in sip.conf or iax.conf). > http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS > That's not correct. DIALSTATUS will be set whether or not you've got qualify=yes in the peer definition. -- Jared Smith Digium, Inc. -- _ -- 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] X-lite direct sip call - Is it possible?
Hi Guys, Wondering if anyone has tried to make a direct SIP peer to peer call using x-lite without any registrations of any sort. I can't seem to find the setting. Thanks, bruce -- _ -- 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] DIALSTATUS variable and qualify=no
Hi there, could anybody tell me if the info below is still correct: Note: In order to obtain useful DIALSTATUS information when dialing a peer you will need to have qualify=yes in that peer's definition (e.g. in sip.conf or iax.conf). http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS THANKS!! -- Regards, Rustam Kovhaev -- _ -- 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 changes not reflected realtime
Hello Steve, I don't really understand what you mean. Do I need to 'sip prune realtime all' after every change ?? Is rtcachefriends=yes a wrong setting ?? Kind regards, Jonas. Steve Howes wrote: On 17 Apr 2010, at 10:25, Jonas Kellens wrote: When changing the secret, the old secret is still the one to use until a sip reload. When changing the name, the old name is still the one to use for registrations until a sip reload. So it's being cached? Does 'sip prune realtime all' clear it too? rtcachefriends=yes By that? S -- _ -- 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 set up Fax on Asterisk - Using analog Fax machines and HT502 (or FXS of a Digium TDM410P)
hin lee wrote: > Can't upgrade the version. how about buying a FXS gateway and be > done with the issue. Go to ebay and search for AudioCodes. You can > get 1 FXS port gateway for around $30 to 2 FXS at $85. Probably the > best bet is to convince the customer to upgrade Asterisk. > > Seems to me at this point, unless YOU know of a technical reason why you should not upgrade, your best bet is to tell the client either to let you upgrade to the latest 1.4 version, or to get someone else to support them. I can understand legitimate legal reasons they my not want to use a fax to email service, but to insist on continuing to use a broken version of asterisk serves no one. They don't have the services they want or need, and you get a black eye because you cannot give them what they want. The customer is NOT always right. > > *From:* Danny Dias > *To:* asterisk-users@lists.digium.com > *Sent:* Thu, April 15, 2010 10:16:26 PM > *Subject:* [asterisk-users] How to set up Fax on Asterisk - Using > analog Fax machines and HT502 (or FXS of a Digium TDM410P) > > Hello Asterisk users, > > We are having MANY but MANY problems configuring an analog fax machine > to work properly on Asterisk, the first thing we do was to plug in the > Fax analog machine to the FXS port of the Digium TDM410P, we set > echocancel=no on zapata.conf and also faxdetect=yes on general > section, but our Asterisk CRASH every time we try to send/receive fax! > > We are using Asterisk 1.4.21 and Zaptel 1.4.11; the card does not > shows any interrupt in /proc/interrupts > > We also tried with a Sangoma B600 on this machine and the same result! > Then we tried with the sangoma and digium card on another asterisk > box, with Asterisk 1.4.30 and DAHDI 2.2.x and the fax was not reliable > 100% but at least Asterisk vener went down > > We cant make any upgrade of Asterisk/Zaptel due to some rules of the > customer, the do not want to use fax2email, they need to use the > panasonic fax machine, this is driving me crazy! > > We also tried with a HT502 with passtrough fax mode and pcmu and pcma > enabled and the same result, asterisk does down when trying to > receive/send a fax > > What could solve our problem? what else should we try about > configuration? just faxdetect=incoming and set echocancel=yes and > that's all? Please your help, we really need to put this working > > Thanks in advance for all your help! > > -- > Saludos > Danny Dias > SkypeID: danny.dias1 > -- Dog is my co-pilot -- _ -- 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 changes not reflected realtime
On 17 Apr 2010, at 10:25, Jonas Kellens wrote: > When changing the secret, the old secret is still the one to use until a sip > reload. > When changing the name, the old name is still the one to use for > registrations until a sip reload. So it's being cached? Does 'sip prune realtime all' clear it too? > rtcachefriends=yes By that? S -- _ -- 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] Realtime changes not reflected realtime
Hello list, Using Asterisk 1.4.25.1 Using realtime sip_buddies I notice that when changing the sip_buddie name (field 'name' and 'username') or secret, this is not implemented until a sip reload. When changing the secret, the old secret is still the one to use until a sip reload. When changing the name, the old name is still the one to use for registrations until a sip reload. asterisk*CLI> sip show peers testcorp3/testcorp3 192.168.1.100 D N 5061 OK (28 ms) Cached RT -- now I change the secret in sip_buddies -- -- I restart my IP-phone -- asterisk*CLI> [Apr 17 11:19:45] NOTICE[24072]: chan_sip.c:16612 sip_poke_noanswer: Peer 'testcorp3' is now UNREACHABLE! Last qualify: 28 [Apr 17 11:19:55] NOTICE[24072]: chan_sip.c:12985 handle_response_peerpoke: Peer 'testcorp3' is now Reachable. (24ms / 2000ms) asterisk*CLI> -- IP-phone is still able to register with the old password -- -- I do a sip reload -- asterisk*CLI> sip reload [Apr 17 11:22:00] Reloading SIP [Apr 17 11:22:00] == Parsing '/etc/asterisk/sip.conf': [Apr 17 11:22:00] Found [Apr 17 11:22:00] == Parsing '/etc/asterisk/users.conf': [Apr 17 11:22:00] Found [Apr 17 11:22:00] == Parsing '/etc/asterisk/sip_notify.conf': [Apr 17 11:22:00] Found -- I restart my IP-phone -- asterisk*CLI> [Apr 17 11:22:58] NOTICE[24072]: chan_sip.c:12985 handle_response_peerpoke: Peer 'testcorp3' is now Lagged. (2025ms / 2000ms) [Apr 17 11:23:05] NOTICE[24072]: chan_sip.c:12985 handle_response_peerpoke: Peer 'testcorp3' is now Reachable. (31ms / 2000ms) [Apr 17 11:23:05] NOTICE[24072]: chan_sip.c:15889 handle_request_register: Registration from '' failed for '192.168.1.100' - Wrong password [Apr 17 11:23:08] NOTICE[24072]: chan_sip.c:12985 handle_response_peerpoke: Peer 'testcorp3' is now Reachable. (30ms / 2000ms) asterisk*CLI> -- Now the new secret is in place !! -- This is my realtime setting in sip.conf : ;- REALTIME SUPPORT ; For additional information on ARA, the Asterisk Realtime Architecture, ; please read realtime.txt and extconfig.txt in the /doc directory of the ; source code. ; rtcachefriends=yes ;rtsavesysname=yes ;rtupdate=yes ;rtautoclear=yes ;ignoreregexpire=yes 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
