Re: [asterisk-users] asterisk-users Digest, Vol 81, Issue 27
On Sat, 9 Apr 2011, darin iv wrote: 0) Don't re-post the entire digest back to the list it came from. Posting 36k of cruft to ask 'How to change SIP port number?' seems somewhat 'newbish.' 1) Try Google. Try 'How to change SIP port number in Asterisk?' 2) Re-post with a new, relevant Subject and you will get relevant responses. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No ringback even though progressinband=yes is set
On 7 April 2011 23:04, Douglas Mortensen wrote: > Steve. Thanks for the insight. I won't pretend to know what "early-audio" is, > but I guess I'm about to find out :-). > > Also, I believe that I have a nearly identical setup like this with the exact > same SIP provider w/o any trouble. However, I think that system must be > running asterisk 1.4 or 1.2 (my guess is 1.4, but I'll have to check to > confirm). Is there a significant difference between 1.2/1.4 & 1.6 in this > scenario? > > Thanks a million!! :-) > > - > Doug Mortensen > Network Consultant > Impala Networks > P: 505.327.7300 > . > > > -Original Message- > From: Steve Davies [mailto:davies...@gmail.com] > Sent: Thursday, April 07, 2011 10:49 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] No ringback even though progressinband=yes is > set > > On 7 April 2011 17:02, Douglas Mortensen wrote: >> Any ideas on why callers who call into my customer's SIP trunk are not >> hearing a ringback tone? I had this on one other asterisk system, and wound >> up needing to set progressinband=yes in the SIP trunk config. >> >> I have set this on the current system & restarted asterisk, but to no avail. >> >> I am using: >> >> AsteriskNOW distro >> Asterisk build is 1.6 from AsteriskNOW repository: >> asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 >> >> Any help would be greatly appreciated! :-) >> >> - >> Doug Mortensen > > > In my personal experience with SIP and 1.6.x, that mostly depends on where > you are sending the call to. It depends on whether the next or subsequent leg > tries to use early-audio for the ring tone, or uses a Ringing event to signal > that is what is happening. It then depends on whether the originating > caller's equipment can understand early-audio ringing. > > We have a setup here where all our trunks support early-audio ringing except > one (an ISDN30 circuit) and we have to juggle things a bit sometimes to > ensure ringing occurs. > > Perhaps provide more details? Or you may find that tracing the SIP gives you > the clue that you need. > > Hope that helps, > Steve > > Early audio is audio that is sent before the call is "answered", usually in the form of a custom ring-tone or perhaps a "cannot connect, try later" message. Some systems do not support it as it can be abused to communicate at least basic information for free. We had a problem with this when connecting Asterisk 1.2 to Asterisk 1.6 via IAX. A 1.2 SIP system will automatically switch into early audio if it sees an early audio frame. 1.6 defaults to not doing this, but there is a parameter to re-enable it. In this case we solved the problem by upgrading to 1.6 everywhere :) Regards, 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] No ringback even though progressinband=yes is set
On 7 April 2011 17:02, Douglas Mortensen wrote: > Any ideas on why callers who call into my customer's SIP trunk are not > hearing a ringback tone? I had this on one other asterisk system, and wound > up needing to set progressinband=yes in the SIP trunk config. > > I have set this on the current system & restarted asterisk, but to no avail. > > I am using: > > AsteriskNOW distro > Asterisk build is 1.6 from AsteriskNOW repository: > asterisk16-1.6.2.17.2-1_centos5 > FreePBX 2.9 > > Any help would be greatly appreciated! :-) > > - > Doug Mortensen In my personal experience with SIP and 1.6.x, that mostly depends on where you are sending the call to. It depends on whether the next or subsequent leg tries to use early-audio for the ring tone, or uses a Ringing event to signal that is what is happening. It then depends on whether the originating caller's equipment can understand early-audio ringing. We have a setup here where all our trunks support early-audio ringing except one (an ISDN30 circuit) and we have to juggle things a bit sometimes to ensure ringing occurs. Perhaps provide more details? Or you may find that tracing the SIP gives you the clue that you need. Hope that helps, 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 meetme invalid extension
On Wed, 6 Apr 2011, satish patel wrote: I have following dialplan for meetme and i want if someone type wrong meetme extension it should say invalid extension. But look like following doesn't work. its just hangup if i type wrong number. how to fix this code.. exten => i,n,Playback(pbx-invalid) The priority should be 1. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] voicemail call back loop
On Tue, 5 Apr 2011, Steve Edwards wrote: Use 'mailcmd' in voicemail.conf. On Wed, 6 Apr 2011, vip killa wrote: I have "externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl" so that when someone is left a voicemail it will call the person's mobile phone and prompt them with the new message. The perl script simply originates a call to a persons mobile phone and connects it to their voicemail using VoiceMailMain. Problem is when user hangs up from checking their messages, it runs the externnotify again causing an infinite loop. Has anybody encountered this problem or is there an option to not have it run externnotify after checking messages? Mailcmd? Also, storing programs that aren't AGIs in the AGI directory doesn't sound like a 'best practice' candidate to me. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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 voicemail callback
On Wed, 6 Apr 2011, vip killa wrote: What about a executing an AGI script with: [general] externnotify = /some_agi_script.agi Would that work? No. What makes a program (compiled or interpreted script) an AGI is that it follows the AGI protocol. Very simplistically, the AGI protocol consists of 2 things: 1) the AGI environment sent to the AGI's STDIN 2) a defined interface to ask Asterisk to do things in the form of a request and a response. Externnotify will work if the program is not an AGI. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Iptables configuration to handle brute, force registrations?
On Tue, 5 Apr 2011, Sherwood McGowan wrote: Why run fail2ban and add overhead when you can just do the same thing with iptables itself? On 4/5/2011 2:11 PM, Steve Edwards wrote: Because it's not the same? The iptables approach is great because it is 'light-weight' and it should already 'be there.' Also, it can react quicker because it doesn't have to read log files to make a decision. The 'downside' of the iptables approach is that the blocks go away when iptables is reloaded -- like when the host is restarted. Probably not an issue with Gordon since his hosts stay up for years. I'm thinking the iptables approach supplemented with a script to periodically save the block list to disk would allow persistent blocks as well as letting you accumulating blocks between all your hosts. Which would still be much 'lighter' than fail2ban. On Tue, 5 Apr 2011, Sherwood McGowan wrote: Agreed on all points Steve. I've already implemented an auto save function, to workaround the drawback you mentioned. Then you're already a couple of steps down the path further than me :) Are there possibly other drawbacks that I'm not seeing/remembering? I've been running an iptables based setup for some time, never really jumped into the fail2ban wagon I've never used fail2ban either. I don't think it's advantages are functional, but the more somewhat intangible: ) It's included with several of the all-in-one Asterisk distributions. ) It's documented. ) It's more flexible ) Somebody else gets to enhance and maintain the code. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Iptables configuration to handle brute, force registrations?
On Tue, 5 Apr 2011, Sherwood McGowan wrote: Why run fail2ban and add overhead when you can just do the same thing with iptables itself? Because it's not the same? The iptables approach is great because it is 'light-weight' and it should already 'be there.' Also, it can react quicker because it doesn't have to read log files to make a decision. The 'downside' of the iptables approach is that the blocks go away when iptables is reloaded -- like when the host is restarted. Probably not an issue with Gordon since his hosts stay up for years. I'm thinking the iptables approach supplemented with a script to periodically save the block list to disk would allow persistent blocks as well as letting you accumulating blocks between all your hosts. Which would still be much 'lighter' than fail2ban. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 create mailbox
On Tue, 5 Apr 2011, vip killa wrote: Is it possible to create a voicemail box using AGI? An AGI executes as a child process when a channel executes agi() via the dialplan. Are you intending to call into Asterisk and let the caller create mailboxes? All the AGI needs to do is add a line to the appropriate stanza in voicemail.conf. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 voicemail callback
On Tue, 5 Apr 2011, vip killa wrote: I'm wondering if there is a simply way to perform a voicemail callback feature using AGI.For instance, a caller leaves a voicemail, the voicemail will then call the owner of the voicemailbox determined by a database look up. Use 'mailcmd' in voicemail.conf. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] dialplan is not finding my number asterisk 1.8.3
Oh, you are *not* going to like this, but you have a few different paths: 1. If the dialplan stuff is not really a memory corruption, but some sort of unplanned, but maybe accidentally programmed thing, either by you or something in the asterisk code, then: a. compile asterisk for debug. You can get in the menuselect stuff and make sure the debug compile options are turned on. Install it. b. Shut down asterisk c. fire it back up, under gdb: gdb br ast_context_remove_extension_callerid2 comm 1 where c end run Then use asterisk as normal and wait for the problem to re-occur. Look to see if any calls to ast_context_remove_extension_callerid2 occurred (they will occur with dial reloads-- lots of them). You'll have to search carefully! The gdb messages could be buried in your console output. If the problem reoccurs, but you didn't see any calls to ast_context_remove_extension_callerid2, then it could be assumed that you are suffering some sort of memory corruption. Where it dies, or things start looking strange, may not indicate where or why the corruption is happening. In such a case, it really gets tricky to debug. Then we might resort to stuff like dmalloc, and others, to help spot where/when corruption occurs. Let's cross that bridge if we come to it. murf On Tue, Apr 5, 2011 at 7:30 AM, Jerry Geis wrote: > Jerry Geis wrote: > >> >> Steve Murphy wrote: >> >>> Idea: >>> >>> If something is corrupting your dialplan, then this should >>> reveal the extent of the corruption: >>> >>> You might, when the system is working properly, do a: >>> >>> asterisk -rx "dialplan show" > somefile1 >>> >>> and then, when you are having problems, do a: >>> >>> asterisk -rx "dialplan show" > somefile2 >>> diff -u somefile1 somefile2 >>> >>> and see if this reveals anything juicy. >>> >>> murf >>> >>> >> Steve, >> >> That is a great idea. I did that the first time it happened. I dumped the >> dialplan, then I restarted >> and dumped again. it was the same. Being the first time I thought it was >> just a fluke but now it >> has happened a couple of times. I have not been able to narrow anything >> down. >> >> Thanks, >> >> jerry >> >> Steve, > > perhaps I did something wrong the first time. As I just got the error > again. I dumped the dialplan and my section: > > > [ Context 'smvoice-mediaport' created by 'pbx_config' ] > > is empty. > > when I restart and dump again. > > > [ Context 'smvoice-mediaport' created by 'pbx_config' ] > '1105' => 1. Goto(smvoice-mediaport-public-address,s,1) > [pbx_config] > 'mediaport_direct' => 1. Goto(smvoice-mediaport-public-address,s,1) > [pbx_config] > 'public_address' => 1. Goto(smvoice-mediaport-public-address,s,1) > [pbx_config] > > I have the correct data. > > The only thing I have in the dialplan for this box is: > > [smvoice-mediaport-public-address] > exten => s,1,System(/home/silentm/bin/smfunctions -stop) > exten => s,n,Playback(beep) > exten => s,n,Dial(Console/dsp) > exten => s,n,Hangup > exten => h,1,System(/home/silentm/bin/smfunctions -start) > > Can a system call be removing stuff from the dialplan? > > What next? > Oh, you are *not* going to like this, but you have a few different paths: 1. If the dialplan stuff is not really a memory corruption, but some sort of unplanned, but maybe accidentally programmed thing, either by you or something in the asterisk code, then: a. compile asterisk for debug. You can get in the menuselect stuff and make sure the debug compile options are turned on. Install it. b. Shut down asterisk c. fire it back up, under gdb: gdb br ast_context_remove_extension_callerid2 comm 1 where c end run Then use asterisk as normal and wait for the problem to re-occur. Look to see if any calls to ast_context_remove_extension_callerid2 occurred (they will occur with dial reloads-- lots of them). You'll have to search carefully! The gdb messages could be buried in your console output. If the problem reoccurs, but you didn't see any calls to ast_context_remove_extension_callerid2, then it could be assumed that you are suffering some sort of memory corruption. Where it dies, or things start looking strange, may not indicate where or why the corruption is happening. In such a case, it really gets tricky to debug. Then we might resort to stuff like dmalloc, and others, to help spot where/when corruption occurs. Let's cross that bridge if we come to it. murf > > Jerry > > -- Steve Murphy ParseTree Corporation -- _ -- 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] Iptables configuration to handle brute force registrations?
On Tue, 5 Apr 2011, Gilles wrote: I'm no expert of iptables, and it seems like it can handle banning IP's that are trying to register and fail too many times. Is there a good iptables configuration that I could use as reference? Gordon Henderson posted a link to his script that handled failures above a threshold and some other cool stuff a few months back. Try searching the archives. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] dialplan is not finding my number asterisk 1.8.3
Idea:
If something is corrupting your dialplan, then this should
reveal the extent of the corruption:
You might, when the system is working properly, do a:
asterisk -rx "dialplan show" > somefile1
and then, when you are having problems, do a:
asterisk -rx "dialplan show" > somefile2
diff -u somefile1 somefile2
and see if this reveals anything juicy.
murf
On Tue, Apr 5, 2011 at 5:44 AM, Jerry Geis wrote:
> Jerry Geis wrote:
>
>> I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a
>> speaker attached.
>>
>> When asterisk first starts this works. In fact it works for some time.
>> Then it just stops with this error on the CLI.
>>
>> [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite:
>> Call from 'mndemo_to_mediaport105' to extension '1105' rejected because
>> extension not found in context 'smvoice-mediaport'.
>>
>> When doing the "dialplan show" it clearly in the context.
>>
>> [ Context 'smvoice-mediaport' created by 'pbx_config' ]
>> '1105' => 1. Goto(smvoice-mediaport-public-address,s,1)
>> [pbx_config]
>>
>>
>> Its telling me it cannot find it. Its there - the dialplan shows its
>> there.
>> When I stop and start it works again for a little while.
>> Matter of fact I just issued "dialplan reload" and calling into 1105 works
>> again.
>>
>> Whats up? How do I get this to be consistent?
>>
>> Jerry
>>
>>
>> I just looked in my extensions.conf and I do not have
> extenpatternmatchnew at all. My understanding is that
> it is off by default.
>
> my sip.conf has:
> register => mndemo_to_mediaport105:secret@mndemo
>
> ; Description:
> [mndemo_to_mediaport105]
> type=friend
> defaultname=mndemo_to_mediaport105
> username=mndemo_to_mediaport105
> secret=secret
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> rtptimeout=60
> host=192.168.1.58
> context=smvoice-mediaport
>
>
> I was not aware I needed another context of :
>
> [mndemo_to_mediaport105]
> include => smvoice-mediaport
>
>
> The context is set above in sip.conf and that is what the CLI above is
> showing its using.
>
>
> Also my extensions.conf section is :
>
> --
> [smvoice-mediaport-public-address]
> exten => s,1,System(/home/silentm/bin/smfunctions -stop)
> exten => s,n,Playback(beep)
> exten => s,n,Dial(Console/dsp)
> exten => s,n,Hangup
> exten => h,1,System(/home/silentm/bin/smfunctions -start)
>
> [smvoice-mediaport]
> exten => public_address,1,Goto(smvoice-mediaport-public-address,s,1)
>
> #include "/etc/asterisk/express.dnis.conf"
>
> --
> where express.dnis.conf has:
> ; Phone Caller ID & DNIS Manager screen
>
> ; MMPCGA: VISUAL PC ROOM 105 - exten =>
> 1105,1,Goto(smvoice-mediaport-public-address,s,1)
>
> ---
> Here is a call that works:
> == Using SIP RTP CoS mark 5
> -- Executing [1105@smvoice-mediaport:1]
> Goto("SIP/mndemo_to_mediaport105-0003",
> "smvoice-mediaport-public-address,s,1") in new stack
> -- Goto (smvoice-mediaport-public-address,s,1)
> -- Executing [s@smvoice-mediaport-public-address:1]
> System("SIP/mndemo_to_mediaport105-0003", "/home/silentm/bin/smfunctions
> -stop") in new stack
> -- Executing [s@smvoice-mediaport-public-address:2]
> Playback("SIP/mndemo_to_mediaport105-0003", "beep") in new stack
> -- Playing 'beep.gsm' (language
> 'en')
> -- Executing [s@smvoice-mediaport-public-address:3]
> Dial("SIP/mndemo_to_mediaport105-0003", "Console/dsp") in new stack
> << Call placed to 'dsp' on console >> << Auto-answered >>-- Called dsp
> -- ALSA/dummy answered SIP/mndemo_to_mediaport105-0003
> -- Executing [h@smvoice-mediaport-public-address:1]
> System("SIP/mndemo_to_mediaport105-0003", "/home/silentm/bin/smfunctions
> -start") in new stack
> << Hangup on console >> == Spawn extension
> (smvoice-mediaport-public-address, s, 3) exited non-zero on
> 'SIP/mndemo_to_mediaport105-0003'
> --
>
>
> As I mentioned starting asterisk all this works. There is some random time
> later - perhaps days where it then stops
> finding the exten.
>
> Is there something I have wrong in the config above?
>
> Jerry
>
> --
> _
> -- Bandwidth and Colocation Provi
Re: [asterisk-users] Dialplan matching
On Mon, Apr 4, 2011 at 8:09 AM, Asterisk User wrote: > > Hello all, I am trying to figure out the logic in on prefix matching for > Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT > calls to 011870, 01137455 and so on. > > exten => _011870.,1,Goto(intl-disabled,s,1) > exten => _01137455.,2,Goto(intl-disabled,s,1) > exten => _01137477.,3,Goto(intl-disabled,s,1) > exten => _0113749.,4,Goto(intl-disabled,s,1) > exten => _011.,5,Goto(intl-disabled,s,1) > exten => _011.,6,Playback(all-outgoing-lines-unavailable) > exten => _011.,7,Wait(1) > exten => _011.,8,Playback(please-hang-up-and-dial-operator) > exten => _011.,9,Hangup > > Is this correct or should it be: > > exten => _011870X,1,Goto(intl-disabled,s,1) > exten => _01137455X,2,Goto(intl-disabled,s,1) > > I tried searching for definitive information on voip-wiki, nerd vittles, > but there is a lot of confusion. > Assuming that 011870 is followed by more than digit, normally, I'd say your first set is more applicable. The . in the pattern at the end means any number of digits, followed by a timeout. If you know the number of digits, and it is fixed, then you could use _011870XXX or similar to avoid the timeout, and begin the Goto immediately on reception of the final digit. The X in the second set will match just one digit, and the Goto will be be executed. Does that help? > > > -- Steve Murphy ParseTree Corporation -- _ -- 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] SIP channel able to add codecs once up and running?
>From my observations, if a video capable device starts the call in non-video mode, it is never able to add video to the channel? Is this correct, or am I missing something? It looks as if the codec 'jointcapability' is calculated at the start of the call, and can never be added to (with exceptions for T.38 fax) as any SDP update is masked using the existing 'jointcapability' and knocks out the newly requested codec. Is that right? Thanks, 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] Is it possible to dial an automated message when Asterisk receives an email?
Un-top-posting... On Sat, Apr 2, 2011 at 3:39 PM, John Kiniston wrote: You could use a procmail recipe to create a call file and then move it to the /var/spool/asterisk/outgoing directory. Below is a untested example .procmailrc: :0: * ^to.trig...@example.com | /usr/local/bin/callout.sh where callout.sh would look like this perhaps: !/bin/bash sleep 5 CALL="callout.call"; echo channel: LOCAL/NUMBER@pstn-local > /tmp/$CALL; echo context: ivr-call-out >> /tmp/$CALL; echo exten: s >> /tmp/$CALL; echo priority: 1 >> /tmp/$CALL; echo mv /tmp/$CALL /var/spool/asterisk/outgoing done Again all untested writing by the seat of my pants type stuff. On Sat, 2 Apr 2011, Rafael Bermúdez wrote: John, Thanks for your reply. I will test this script. A couple of comments of the top of my head: ) If you construct the call file name using the PID you can accomodate more than a single event at the same (or really close) time. ) The 'mv' command has an extraneous 'echo' ) Pay attention to permissions. Sudo can help if needed. Personally, I prefer a restrictive sudo to the blunt hammer of wide open permissions :) -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Is it possible to dial an automated message when Asterisk receives an email?
On Sat, 2 Apr 2011, Rafael Bermúdez wrote: I have a server that sends a preformatted email when an event occur. What I need is that when Asterisk receives this email automatically dial a pre-recorded message. It doesn't have to dial ride away, maybe a scheduled cron job will be enough. Procmail, call files and a little scripting would be one approach. Can you take a step back up the chain and have your server: ) Execute the script to create the call file? ) Create the call file on a shared drive? ) Connect to Asterisk via AMI to originate the call? Thanks, and sorry for my lousy English Probably much better than my ish. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Registration from '"000000" x 1000
On 2 April 2011 09:46, Jonas Kellens wrote: > Hello list, > > I often see the following in my message log : > > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > > And there are hundreds of them... > > > Is there a setting so I can make Asterisk not respond to SIP PEER > registrations which are not in my sip.conf or my realtime MySQL DB ?? Yes, you add a rule to your firewall! Even better, get it filtered further out so that it does not waste your inbound Internet bandwidth, because in my experience, once those SIP spammers start, they continue for weeks at the very least. IIRC, the way SIP registrations works basically requires than an failed/un-authorised attempt is responded to, so that the other party knows to authenticate. If you stop sending that response, no-one can authenticate. Hope that helps. 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 and fail2ban
On 03-29-2011 19:25, Steve Edwards wrote: Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? On Tue, 29 Mar 2011 23:09:06 +0200, ad...@3a.hu wrote: after reviewing last week's log i'd say around 25-28k/min :) On Tue, 29 Mar 2011, Gilles wrote: So it looks like I should check out sshguard instead of relying on blocks of IP's :-) It's not A or B, think A AND B. Security should be in layers -- my pocket GPS is in my locked glove box, in my locked car, in my locked garage, in my gated community. If there is never a need to accept callers from North Korea, how will you explain to your boss that some NK script weenie discovered some weakness in A or B and racked up a bazillion minutes to Libya? What if you misconfigure A or B? What if A or B has a 'window of opportunity' during system restart? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 and fail2ban
On Tue, 29 Mar 2011 12:34:04 -0500, Sherwood McGowan wrote: Remember guys, there's a LOT of IP blocks out there that are almost definitely not going to be somewhere you expect to receive SIP traffic from. On Tue, 29 Mar 2011, Gilles wrote: I agree. Is there a list I could use to check which blocks have been allocated to which countries so I can add them to Asterisk's blacklist? I posted this several months ago: http://www.voip-info.org/wiki/view/allocated-class-a-ip-address-blocks -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 and fail2ban
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan First thing I'd do is restrict the ip blocks your sip endpoints can register/call from in sip.conf (or your database's table for sip endpoints) On Tue, 29 Mar 2011, Gilles wrote: Thanks for the idea, but it's not possible, as the Asterisk must be accessible for road warriors and receive SIP calls from anyone. Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] DAHDI, IAX2 and SIP considerations for Early-Media / Alerting
Hi, Short version: Is it possible or even legal to convert an IAX2 PROGRESS/EARLY-MEDIA indication into a DAHDI/q.931 ALERTING signal when your ISDN provider does not pass early media on receipt of an PROGRESS(8) indication? Long version: I have an Asterisk 1.6.2.18-rc1 based system with a DAHDI trunk (UK E1 line), also, the system has IAX2 trunks, and several SIP handsets. All 3 protocols (q.931/IAX2/SIP) have a mechanism to indicate either ALERTING/RINGING, or to specify PROGRESS/EARLY-MEDIA. Based on this you'd think call setup would all work happily all of the time :) What happens based on the call direction is as follows: SIP -> DAHDIISDN returns ALERTING, SIP uses 180 Ringing, all OK SIP -> IAX2 IAX2 returns PROGRESS, SIP uses 183 Progress, early audio works OK IAX2 -> DAHDI ISDN returns ALERTING, IAX2 uses RINGING, all OK IAX2 -> SIP SIP returns 180 ringing, IAX2 uses RINGING, all OK DAHDI -> SIPSIP returns 180 ringing, ISDN uses ALERTING, all OK DAHDI -> IAX2 IAX2 returns PROGRESS, ISDN uses PROGRESS(8), but the caller hears no ringing. I believe that my issue is that my UK ISDN provider does not accept early media, and will simply send silence instead of using the provided early audio stream. DAHDI is configured with: priindication=outofband The IAX2 trunk provider is using early-media to send the ringing tone, and as above, this mostly seems to work okay. The exception is when the call is bridged to ISDN, where I believe the ISDN provider does not pass on early media. I checked the IAX2 RFCs 5456/5457, but cannot find a definition of how RINGING/PROGRESS is meant to work. Is my IAX2 trunk provider doing something wring by not also sending RINGING? Is there a workaround that converts either IAX2 PROGRESS into RINGING, or allows DAHDI to send ALERTING if it receives an early media indication? I suspect the code to do the latter would be reasonably simple, but would appreciate pointers for any badness that it may cause. Thanks in advance for any suggestions. Regards, 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] What is the most stable version of asterisk?
1.2 is not active either. Both are solid. I am loving SNOM phones and OpenVPN software. Only port(s) open is what I assign to OpenVPN. CallWeaver was way ahead of asterisk at the time but you are right it died. Generally, the newer, the worse. 1.2 was very solid except for a few strange things that could be worked around. Newer versions have are like Fedora Core (or FC X) you are just testing beta software for Digium's commercial products. Thanks, Steve Totaro On Fri, Mar 25, 2011 at 10:25 AM, Douglas Mortensen wrote: > Based on the following URL, it seems that CallWeaver may not still be an > active project?? > > http://www.callweaver.org/blog/20 > > From a security standpoint, I would usually expect it is safer to be with > an active project, than a dead one. Otherwise who is going to patch > vulnerabilities? Not me. I'm not a software developer. > - > Doug Mortensen > Network Consultant > Impala Networks > P: 505.327.7300 > . > > From: Steve Totaro [mailto:stot...@totarotechnologies.com] > Sent: Thursday, March 24, 2011 11:11 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] What is the most stable version of asterisk? > > > On Wed, Mar 23, 2011 at 12:58 PM, Douglas Mortensen < > d...@impalanetworks.com> wrote: > 1.2? 1.4? 1.6? 1.8? > > Thanks, > - > Doug Mortensen > Network Consultant > Impala Networks Inc > CCNA, MCSA, Security+, A+ > Linux+, Network+, Server+ > . > www.impalanetworks.com > P: (505) 327-7300 > F: (505) 327-7545 > > > Callweaver? http://www.voip-info.org/wiki/view/CallWeaver. I believe > they forked somewhere in the 1.2 release. Many features ahead of Asterisk. > > Although I didn't see anything on FreeSwitch stating anything anything > about deadlocking, I know that was one of the main reasons for BKW, as > seasoned asterisk developer and folks to start from scratch. That and the > hybrid dual license in Asterisk. > > > http://www.sofaswitch.org/docs/How%20does%20FreeSWITCH%20compare%20to%20Asterisk.pdf > > Read the whole piece. I know it isn't Asterisk but BKW who contributed and > I believe is still helping Asterisk > > Besides, I feel that FreeSwitch is the most stable. > > I like 1.2 so I went with Callweaver for many installations. > > 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 > -- _ -- 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] checking dahdi channels
On Fri, 25 Mar 2011, Nathan Pryor wrote: Is there a command I could use directly in the dialplan or with the manager interface to get the number of used channels? Check out the GROUP() and GROUP_COUNT() functions. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Sox and bad quality when converting to 8 kHz
On Fri, 25 Mar 2011, Steve Underwood wrote: You really need to remove the bass end of the spectrum before downsampling to 8k/s. uLaw/ALaw sound pretty muddy and horrible if you don't do that, and the other common 8k/s codecs don't sound any better. Jean-Marc Valin wrote a little filtering utility for this purpose. A link? Casual googling didn't yield anything promising. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Sox and bad quality when converting to 8 kHz
On 03/25/2011 04:58 AM, Thomas Winter wrote: Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help best regards Thomas You really need to remove the bass end of the spectrum before downsampling to 8k/s. uLaw/ALaw sound pretty muddy and horrible if you don't do that, and the other common 8k/s codecs don't sound any better. Jean-Marc Valin wrote a little filtering utility for this purpose. 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] What is the most stable version of asterisk?
On Wed, Mar 23, 2011 at 12:58 PM, Douglas Mortensen wrote: > 1.2? 1.4? 1.6? 1.8? > > Thanks, > - > Doug Mortensen > Network Consultant > Impala Networks Inc > CCNA, MCSA, Security+, A+ > Linux+, Network+, Server+ > . > www.impalanetworks.com > P: (505) 327-7300 > F: (505) 327-7545 > > > Callweaver? http://www.voip-info.org/wiki/view/CallWeaver. I believe they forked somewhere in the 1.2 release. Many features ahead of Asterisk. Although I didn't see anything on FreeSwitch stating anything anything about deadlocking, I know that was one of the main reasons for BKW, as seasoned asterisk developer and folks to start from scratch. That and the hybrid dual license in Asterisk. http://www.sofaswitch.org/docs/How%20does%20FreeSWITCH%20compare%20to%20Asterisk.pdf Read the whole piece. I know it isn't Asterisk but BKW who contributed and I believe is still helping Asterisk Besides, I feel that FreeSwitch is the most stable. I like 1.2 so I went with Callweaver for many installations. 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] Sox and bad quality when converting to 8 kHz
On Thu, 24 Mar 2011, Thomas Winter wrote:
I have an 44100 Hz file with human voice, stereo with 16Bit.
When convertig this to 8 kHz, mono I loose a lot of quality and have
some ground noise. I tried several sox options but without success.
Can somebody help
What does your command line look like?
I use this with good results:
sox "${INPUT}" -c 1 -s -w -r 8000 "${OUTPUT}"
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] What is the most stable version of asterisk?
On Wed, 23 Mar 2011, Douglas Mortensen wrote: 1.2? 1.4? 1.6? 1.8? On Thu, 24 Mar 2011, Gordon Henderson wrote: 1.2 has been the most stable version for me. Same setups with 1.4 +DAHDI has never been as stable with random crashes and re-starts - however they're not predictable and sometimes months apart. I had one instance of 1.2 run for over a year without a hiccup. I've not even thought about 1.8 yet. Well that's disconcerting. I've been feeling that I've been remiss by not moving my 1.2 clients into 1.6. I only have one 1.2 site that has issues. It processes about 15,000 calls a day and has a memory leak that affects voice quality after about 2,000,000 calls -- about every 4 months. This host runs a version of meetme I hacked up. The client needed custom 'whispers' played to their agent before a client joins the conference so the leak may be my fault. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 Tech Tips: Calling With Google Starts At Noon Central (30 minutes from now)
Greetings User's List, Just a last minute reminder that Malcolm Davenport will be headlining our first Asterisk Tech Tips webinar covering the cool ways you can integrate Asterisk with Google Chat and Google Voice. Register now and join us in 30 minutes. Here's the link: http://www.asterisk.org/techtips Thanks! -S Steve Sokol Asterisk Marketing Director 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] Reminder: Asterisk Tech Tips: Calling With Google on Thursday at 12PM CDT
Greeting Users List, Just a friendly reminder that the first Asterisk Tech Tips webinar will take place this coming Thursday at 12PM. The first half will be an in-depth tutorial on calling with Google presented by Malcolm Davenport, Senior Product Manager for Asterisk. Malcolm will teach users to connect Asterisk with both Google Chat (a.k.a. voice and video chat in Gmail) and Google Voice. Register to attend at: http://www.asterisk.org/techtips For more about the Tech Tips series, check out the blog post: http://blogs.digium.com/2011/03/21/asterisk-tech-tips-thursdays-at-noon/ Cheers, -S Steve Sokol Asterisk Marketing Director 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
Re: [asterisk-users] [1.4] Failed callfile doesn't jumpto"failed"extension
On Sat, 19 Mar 2011, Gilles wrote: Thanks but for some reason, after calling out through a call file, Asterisk doesn't jump to it although the callee hangs up while Asterisk is still playing: Somehow, I'm guessing that 'failed' means that something failed while processing the call file or that the call failed to answer, not that somebody terminated the call. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] [1.4] Failed callfile doesn't jumpto"failed"extension
On Behalf Of Gilles
Unfortunately, it can only jump to "h", and ${REASON} is empty.
On Fri, 18 Mar 2011, Danny Nicholas wrote:
I believe you will achieve the desired result by replacing ${REASON}
with ${HANGUP_CAUSE}.
REASON is documented as being valid in the 'failed' extension. If it is
not working as you expect it to, maybe you could read through the source
(/usr/src/asterisk-x.x.x.x/main/pbx.c) to understand why.
You could always submit a patch...
HANGUP_CAUSE should be HANGUPCAUSE.
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Executing shell commands via AMI
On Wed, 16 Mar 2011, Vinícius Fontes wrote: But I really don't see much of a threat on this. AGI does almost the same. I thought you didn't want to start a flamefest :) The security risk of AGI would be 'the same' if you provide a method for a miscreant to create a file on your Asterisk server, make it executable, modify your dialplan, reload the dialplan and execute that section of the dialplan. If all these conditions are already in place, your definition of 'secure' is different than mine. The ability to [remotely] execute a shell command via AMI does sound interesting. Can you describe where this would be needed and could not be accomplished with existing tools like ssh and sudo? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] New Webinar Series For Asterisk Users: Asterisk Tech-Tips
Greetings Users! On March 24th we're hosting the first of a new series of webinars entitled "Asterisk Tech-Tips". Our goal is to present a new "episode" or "issue" or "webisode" (or whatever you want to call it) every other week. Here's the idea: Asterisk Tech-Tips are all about helping people get the most from their Asterisk systems. Each webinar will start out with a tutorial covering some bit of Asterisk-fu in-depth. Our first topic is Calling with Google. Our first presenter is the internationally famous Asterisk guru Malcolm Davenport, Sr. Product Manger for Asterisk at Digium. Malcolm will take you through the process of setting up Asterisk to take advantage of Google Voice and Google Chat. When he's done, you'll know how to make and receive calls from Google users and how to make free US calls using Google Voice and Asterisk. Once we're done with the tutorial we'll open up the floor for general questions and answers. You're welcome to ask about the tutorial topic, but feel free to ask about any Asterisk-related subject. We hope to turn this into a dialog that helps people discover all the amazing things you can do with Asterisk. You can register now for the event: http://www.asterisk.org/techtips I hope to see you there! Cheers, -S Steve Sokol Asterisk Marketing Director Digium, Inc. PS. If you would like to suggest a Tech-Tips topic or would like to present a tutorial, please let me know. We're always looking for cool new things you can do with Asterisk. -- _ -- 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 -rx command not returning data - Version 1.4.33.1
On Mon, 14 Mar 2011, Paddy Grice wrote: I am having trouble running the command siptest:~# asterisk -rx 'dialplan reload' I assume the problem is timing but any ideas on how to fix it I'm just a 1.2 Luddite, but it's been my experience that issuing shell command lines and parsing the output is unreliable. Kind of hit or miss, sometimes you get more that you expect, sometimes less. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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 and PlayBack
On Sat, 12 Mar 2011, Thomas Winter wrote: when I audio studio should produce an sound file to play back with Asterisk. Whats the best format they should deliver the audio file? I like to receive audio at the highest quality the studio can provide and then transcode down to what Asterisk can handle. I can then encode with all the codecs I need. If I ever get to where I can use HD codecs, I still have the originals from the studio. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] G.711.0
Hi, Has anyone seen G.711.0 in real world use? The spec was published quite a while ago, but as far as I can tell there is no RFC defining the SDP and RTP details needed to deploy it, and nobody advertises that they support it in their products. 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 1.8 AGI error ast_carefulwrite: write() returned error
Un-top-posting... On Fri, 11 Mar 2011, satish patel wrote: We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ? [Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned error: Broken pipe On Mar 11, 2011, at 4:58 PM, Steve Edwards wrote: Without source code, I'd guess you are violation the AGI protocol. Can you reduce your source code to a simple application that reliably reproduces the error. Can you post the source to the simplified application? On Fri, 11 Mar 2011, Satish Patel wrote: I am not in office so i can't post script right now but will so once reach home. If you want to take a look at script I have following URL where someone already doing discusion. My script is pretty similer but I am grabbing all active extension via asterisk CLI commands not statically hardcoded. http://www.freepbx.org/forum/freepbx/tips-and-tricks/delayed-paging If you are referring to the allpage.agi script posted about 40% down the page... It is not an AGI. Note that it does not use any AGI library and that it does not read the AGI environment from STDIN -- which violates the AGI protocol. The allpage script connects to Asterisk via TCP using the AMI protocol. In your dialplan, if you change 'agi(allpage.agi)' to 'system(allpage.agi)' does it behave as you expect? Can you execute the script from a shell command line? -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 AGI error ast_carefulwrite: write() returned error
On Fri, 11 Mar 2011, satish patel wrote: We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ? [Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned error: Broken pipe Without source code, I'd guess you are violation the AGI protocol. What language are you using? which AGI library are you using? Can you reduce your source code to a simple application that reliably reproduces the error. Can you post the source to the simplified application? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] One Way Audio
On 10 March 2011 11:17, Ishfaq Malik wrote: > Just fixed our problem with > > directmedia=no > > but this only applies if your extensions are behind a nat > > Ish > There are several reasons why "directmedia=no" might be the correct configuration. 1) NAT - probably the most common reason 2) Routing - Sometimes devices cannot route to each other directly 3) ITSP calls. Many SIP providers will not accept a redirect and I am sure there are many more... Cheers, 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] OT: OpenSIPS vs Kamailio -- which do you use and why?
On Fri, 4 Mar 2011, Steve Edwards wrote: I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :) Which do you use and why? So I got 1 'vote' for each. Surely more than 2 users use OpenSIPS or Kamailio. I guess Friday afternoon is not the best time to post an open question :) -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Configuration for Multiple PRI cards
Un-top-posting... On Sat, Mar 05, 2011 at 08:14:37PM +0200, Elliot Murdock wrote: How does one go about creating a dahdi configuration file for multiple PRI cards? On Mon, Mar 7, 2011 at 10:00 PM, Tzafrir Cohen wrote: 1. vi 2. dahdi_genconf handles the common case quite well and will normally be a good start. On Mon, 7 Mar 2011, Gopalakrishnan A.N wrote: Basically each PRI card will be configured as g0, g1 and so on. Group is not bound by card or span. It is applied to a range of channels. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 anyone tell me how to set asterisk to record all phonecall
Un-top-posting... Please do not copy me privately. I obviously already read the mailing list -- that's how I found your first post. On Mon, 7 Mar 2011, John Wu wrote: Thanks for your reply, I did what u said, but still can not work. the /var/spool/asterisk/monitor do not have any file. Attachment is my extensions.conf On Mon, Mar 7, 2011 at 10:16 AM, Steve Edwards I don't think Jeremy intended for you to copy his example literally. Do you really have your endpoints pointed at '[outbound-or-wherever-you-dial]?' I suggest you take a step back and read 'Asterisk: The Future of Telephony' to get a bit more insight into how Asterisk works. You can buy a paper copy or google for the free PDF. On Mon, 7 Mar 2011, John Wu wrote: But I do not have many time to read asterisk usage. I will read it when I got free time. I find other ones use [macro-record-enable] to achieve auto record phone call. Can I directly use it? and how to debug the configure file? Please take the time before posting again. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 anyone tell me how to set asterisk to record all phonecall
On Mon, 7 Mar 2011, John Wu wrote: Thanks for your reply, I did what u said, but still can not work. the /var/spool/asterisk/monitor do not have any file. Attachment is my extensions.conf I don't think Jeremy intended for you to copy his example literally. Do you really have your endpoints pointed at '[outbound-or-wherever-you-dial]?' I suggest you take a step back and read 'Asterisk: The Future of Telephony' to get a bit more insight into how Asterisk works. You can buy a paper copy or google for the free PDF. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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, Sent accountcode between 2 asterisk
On Sat, 5 Mar 2011, brya...@zktech.com wrote: Send the account code as a custom header variable encode it on A and read it on B. You can send any variables you want using this method. I currently send about 10 variables on switch transfers. If you need an example ping me back and I will send one when I get in the office. Just noticed you are using IAX I don't think my method works with IAX. That is why I use SIP between systems. Someone correct me if there is a way to send custom variables with IAX. You can pass cruft between Asterisk servers via IAX using the caller ID name. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Help Asterisk / API / Perl
On Sat, 5 Mar 2011, Olivier CALVANO wrote:
i want use the API on my asterisk 1.6, but i have a small problems :
$typ don't have SIP or IAX, same test without succes:
$typ = $AGI->get_variable('type');
'agi_type' is part of the AGI environment, not a channel variable.
Read the documentation for your AGI library to see how to access the AGI
environment variables -- the cruft Asterisk writes to the STDIN of your
AGI before any of your requests.
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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[asterisk-users] OT: OpenSIPS vs Kamailio -- which do you use and why?
I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :) Which do you use and why? -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Provider Recommendation in US
On Thu, Mar 3, 2011 at 11:30 AM, Danny Nicholas wrote: > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen > Sent: Thursday, March 03, 2011 10:28 AM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] SIP Provider Recommendation in US > > On 11-03-03 11:22 AM, Brent A. Torrenga wrote: > > I am becoming frustrated with our current VOIP provider. Does anyone > have > > any suggestions for a provider that supports asterisk well and provides > > solid service? Voip-info.org has a husge list of providers, but it is > > impossible to tell the fly-by-night operations from the reputable > providers. > > I've had good luck with bandwidth.com for a couple of customers running > call > > centers. > > Leif. > > I'm happy so far with VoicePulse. Keep in mind, if you dig far enough, > you'll find good and bad comments about any provider. > > I managed a decent size system pushing $40k/mo through VoicePulse. Most of it going to Uganda, Peru, and Fiji. The voice quality was never an issue except for VSAT or whatever. My only complaint from years ago when I ditched them, through inheriting the new system is that they are constantly changing things that require user intervention. If you miss those emails, your phone system stops working. If I wanted that I would hard code their IPs rather than using DNS. Other than that, they had very good quality without any kind of direct link or real QoS. Obviously, pushing that amount of traffic their way also got me same day replies from the CEO and if I really complained, and threatened to move the business, he jumped on a plane to come visit but I was gone to Iraq already. Bandwidth.com is a bit pricey but great. I try to negotiate out old paradigm terms that are not applicable to VoIP so much, such as channels. Sure, you can limit me to whatever number we agree upon, but I am not paying an MRC for them unless they are going to give me a layer 2 handoff. Gafachi is good and pretty cheap too, I was bit surprised. 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] mySQL connection testing
On Thu, 3 Mar 2011, Andrew Thomas wrote: Gentlemen, can we please not turn this in to an Asterisk and DB commands bashing thread? I'm just suggesting that maybe you are 'swimming upstream' trying to use MySQL within the dialplan. Much the same as if you were proposing an office system using a 'tin cans and string' mesh with carrier pigeons for out of band call signaling and having a problem with poop buildup on the endpoints -- I might propose using Asterisk :) -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] mySQL connection testing
On Thu, 3 Mar 2011, Andrew Thomas wrote: Does anybody know of a way to test whether a mySQL connection invoked from the dialplan is current or not? I've never been a fan of using database commands in the dialplan. I prefer to wrap up all the database cruft into a nice little black box, an AGI, where I have full access to the database API and real debugging tools. I think database commands in the dialplan are just ugly. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Converting MP3 files to wav for Asterisk
On Thu, 3 Mar 2011, Timothy Smith wrote:
Do you guys know of a better way I can convert mp3 to wav and restore
quality?
You can't restore quality lost by converting to MP3. You shouldn't be
'losing' quality by converting from MP3 to WAV.
#!/bin/bash
for i in `ls $1/*mp3`
Using "for i in $1/*mp3" will save creating a process.
do
lame -a $i $i.wav
Lame is an MP3 encoder. Specifying '.wav' doesn't create a WAV encoded
file, just a 'single channel' (because of '-a') MP3 encoded file with
'.wav' tacked on the end of the file name.
mplayer -quiet -vo null -vc dummy -ao pcm:waveheader:file="$i.h.wav" $i.wav
sox $i.h.wav -t raw -r 8000 -s -2 -c 1 `echo $i|sed "s/.mp3/.sln/"`
Using "sox $i.h.wav -t raw -r 8000 -s -2 -c 1 ${i%.mp3}.sln" will save
creating a process.
done
-
I tried your commands converting an old Pink Floyd track and it sounded
about as good as I would expect.
Try something 'simpler'
mpg123 -q -w "${TEMP}" "${INPUT}"
sox "${TEMP}" -c 1 -s -w -r 8000 "${OUTPUT}"
and see if that helps. Otherwise, how do the 'intermediate' files in your
process sound? Can you hear when things fall apart?
If you post a link to a sample input file and a 'degraded' output file,
this may provide more clues.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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Re: [asterisk-users] how to use qualify times to route calls
On Wed, 2 Mar 2011, sean darcy wrote: That would be a great idea, but would stretch my limits. Isn't that what makes it fun? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] records inbound and outbound calls
Un-top-posting...
2011/3/2 Danny Nicholas
How I did it
exten => 3009,1,Answer()
exten => 3009,2,MixMonitor(test.wav|av(0)V(0))
exten => 3009,3,Dial(SIP/144)
exten => 3009,4,Hangup()
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
salaheddine elharit
thank you i have one question waht is 3009 is the called
On Wed, 2 Mar 2011, Danny Nicholas wrote:
I made a sub-context 3009 in default to let me call from my phone
“sipphone” to my phone “144” and record the conversation.
3009 is an extension, not a [sub]context.
I'd add a suggestion to use the 'n' priority to make maintenance easier.
I use
'mixmonitor(/tmp/${EXTEN}-${STRFTIME(${EPOCH},,%Y-%m-%d-%H-%M-%S)}.wav)'
on my dev box so the file name has the number I dialed as well as the
timestamp.
Also, recording calls without warning is illegal in [many|most|all]
countries.
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
_
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Re: [asterisk-users] wav files are not playing asterisk
On Tue, 1 Mar 2011, Nikhil wrote: I try to play a wav file in asterisk ,but its accepting only gsm files.Do u know where I need to change to make it works. Asterisk chooses a file encoding based on the channel encoding. If your channel is encoded as GSM, Asterisk will not look for a .wav of the same name if a .gsm is available. If the .gsm is not available, Asterisk will use the .wav with the additional 'overhead' of transcoding the data to GSM. Without any console log, these are just guesses: 1) Don't specify the file type in your dialplan. Asterisk chooses a file type for you based on channel encoding and formatting modules loaded. Is format_wav.so loaded? 2) Your WAV file is not encoded correctly. The 'file' command should show something like 'RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz' 3) You have permission issues. From a shell, as the user your instance of Asterisk runs as, can you access the file? If this doesn't help, please repost including the relevant dialplan context (as displayed by 'dialplan show ) and a snippet of the console log of a call playing the WAV file. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Using voice modem as poor man's FXO in Asterisk 1.8
On 02/28/2011 10:12 AM, Stuart Longland wrote: Hi all, I've tried researching this, and so far, have struggled to find any contemporary information on the issue, so I do apologise if asking this irritates people who have answered this before. I have managed to set up Asterisk 1.8 on the web server here. I have two softphones (Ekiga) able to communicate with it. So far so good. I'm now curious to see if I can link it with the PSTN phone line here. The web server in question is an Intel Atom system with a Mini-ITX motherboard. Its one and only PCI slot is occupied by a PCI ethernet card. So FXO card is not an option even if it were within budget. My options therefore look to be an external FXO device of some description (Ethernet or USB), or to use a voice modem. I fear external FXOs are going to be even more expensive than internal FXO cards. Now, I have here an old Maestro JetStream 56k modem here that does amongst other things, voice comms, and I have used it in the past as a telephone by plugging a headset into the front of it (and it was full duplex too if I recall correctly). I have also used it as an answering machine, with the audio being transmitted digitally over the RS232 link. So that to me suggests it is possible to get audio in to and out of the modem, either via a sound card or using the serial port. The web server has a sound card too (hard not to buy a motherboard with one these days). Apart from the lack of any hardware signal processing, it seems all the components are there. The server isn't particularly heavily loaded, and thus I see no reason why the machine wouldn't theoretically be able to handle the DSP in software … I've seen lesser hardware do quite sophisticated DSP in real-time. Now, I've hunted high and low for where this is configured. Some mailing list threads point me to the nonexistant /etc/asterisk/modems.conf. One points me to /etc/asterisk/phone.conf, but nothing there jumps out at me as being an obvious means for configuring a modem — nor can I find where it's documented on the Asterisk wiki. Where abouts should I look for documentation on configuring these modules? Regards, There is no requirement for DSP. There is a requirement for getting duplex audio in and out of the PC. *Very* few full blown external modems will do that. The very simple USB winmodems will, but nobody has produced drivers to make it work for any of the common chips used in those devices. Its not hard to do, though. Source code exists which is not a million miles from that required to hook a USB winmodem into DAHDI. 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] Using voice modem as poor man's FXO in Asterisk 1.8
On Mon, 28 Feb 2011, Stuart Longland wrote: Now, I have here an old Maestro JetStream 56k modem here that does An external modem is a non-starter. If you have infinite time and your time is worth US$0 and you're doing it just for the thrill of it -- maybe. Ward Mundy & crew seem to think this kit (http://nerdvittles.com/?p=720) is pretty hot stuff. Sangoma makes a 2 FXO port USB thingy that looks interesting. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] missing argument on AGI
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron
actually the ${OUTBOUND} is generated by the php... but based on what
will be received on ${ARG1}. unfortunately i am not getting the value
from the argument. not sure why. thanks again.
On Fri, 25 Feb 2011, Danny Nicholas wrote:
This probably seems too elementary, but a "standard practice" I use with
my PERL AGI scripts is to run each of them from bash before I put them
into the dialplan so I have an idea how they will react.
In your particular case, you should be able to do this from bash:
$ php getchannel.php 100 (change 100 to what you are sending in the
dialplan)
The program should print back out
"Set OUTBOUND="
Until you get this output, you are wasting everyone's time.
A 'better' way to 'desk check' an AGI would be to feed it a somewhat
static AGI environment using a script like:
agi-environment.sh
==
# output a fake AGI environment
echo "agi_accountcode: "
echo "agi_callerid: 1234567890"
echo "agi_calleridname: example"
echo "agi_callingani2: 0"
echo "agi_callingpres: 0"
echo "agi_callingtns: 0"
echo "agi_callington: 0"
echo "agi_channel: SIP/example"
echo "agi_context: example"
echo "agi_dnid: *"
echo "agi_enhanced: 0.0"
echo "agi_extension: *"
echo "agi_language: en"
echo "agi_priority: 1"
echo "agi_rdnis: unknown"
echo "agi_request: example"
echo "agi_type: SIP"
echo "agi_uniqueid: 1234567890.12"
echo ""
# respond to requests
echo "200 result=0"
echo "200 result=0"
echo "200 result=0"
echo "200 result=0"
# (end of agi-environment.sh)
And then:
./agi-environment.sh | /var/lib/asterisk/agi-bin/getchannel.php 100
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] [1.4.39.2] Simple AGI doesn't reply
On Thu, 24 Feb 2011, Gilles wrote: = /var/tmp/basic.agi #!/bin/bash #Ripped from #http://lists.digium.com/pipermail/asterisk-users/2003-July/008554.html while read -e ARG && [ "$ARG" ] ; do done echo "NOOP Here" while read line do On Thu, 24 Feb 2011, Gilles wrote: Turns out Bash doesn't allow empty loops. Bash has a thing about syntax too. Note you're not 'done' with your second loop. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] [1.4] Still can't get it to call back
On Thu, 24 Feb 2011, Gilles wrote:
No matter what I try, Asterisk still fails dialing back through a
callfile built through an AGI script.
I don't think it has anything to do with the method used to create the
call file.
AGI script
#!/var/tmp/lua
--Must first empty stdin
while true do
local line = io.read()
if line == "" then break end
-- Without line below, script never ends
io.write("NOOP ",line,"\n")
end
This script violates the AGI protocol.
In addition to suggesting to use an established library, I'd suggest
picking a language and sticking with it. Personally, I use C because it's
the sharpest tool in my toolbox.
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
On Wed, 23 Feb 2011, vip killa wrote: I've exhausted every option without paying someone to fix this, so asterisk might as well be commercial software. You 'effing' kill me :) You have to be a troll. You can't be this stupid. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Un-top-posting... On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas wrote: My bad – “natively” means using the Queue command from the dialplan. Since the “powers that be” are aware of this problem, I suppose it will get fixed when somebody either has some spare time or a sufficient bounty is offered. On Wed, 23 Feb 2011, vip killa wrote: Yes, they want money, they've told me that several times...it's unfortunate that asterisk's dev community is not in it to make a good product but a profit And what are you 'in it' for? The developer community is populated by many kinds of people. Some do it because it's their job, some do it for the challenge, some do it 'for the greater good' and some do it as 'a gun for hire.' Whatever their motivation, are you receiving more than you give? My guess is 'yes' which makes it 'fortunate' for you and me. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] calls between iax and sip
On Wed, 23 Feb 2011, salaheddine elharit wrote: == Agent '1018' logged in (format ulaw/slin) An agent is not the same as an extension. but when i call from sip extension 106 to iax extension (1018) i got the message below [Feb 23 09:55:49] NOTICE[25420]: chan_sip.c:13952 handle_request_invite: Call from '106' to extension '1018' rejected because extension not found. This NOTICE is from the SIP channel driver, not the IAX channel driver. What does the dial statement that generates the above NOTICE look like? What does 'iax2 show peer 1018' display? -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] [1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe
On Tue, 22 Feb 2011 14:54:38 -0600, Warren Selby You're not properly reading in the response after each NoOp you send out. Each time you send something to asterisk in AGI, you must read the response in your script. On Wed, 23 Feb 2011, Gilles wrote: Thanks for the tip. It's working now. While the documentation on the protocol is clear, nobody gets it right the first time -- which is why I always suggest using an established library for the language of your choice. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] calls between iax and sip
On Tue, 22 Feb 2011, salaheddine elharit wrote: i have asterisk installed and i have configured a client iax and sip without any issue, when i call a internal extension sip from iax there is no problem but when i call a iax extension from sip extension the result is KO(wrong number) any help please No details, no help. Crank up verbosity on the CLI and see if the messages yield a clue. If not, please post the console messages. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] IAX channel name incorrect - Found in 1.2 still happens in 1.6
*Bump* No takers? Perhaps no-one else thinks this is a bug? Regards, Steve On 7 February 2011 16:45, Steve Davies wrote: > Hi, > > The following IAX config (slightly edited) causes an issue for me in > version 1.6.2.16.1, where my CDR data is unreliable. > > [user1] > type=friend > auth=md5 > accountcode=user1 > notransfer=yes > context=context1 > host=10.0.0.250 > username=user1 > secret=secret1 > disallow=all > allow=alaw > > [user2] > type=friend > auth=md5 > accountcode=user2 > notransfer=yes > context=context2 > host=dynamic > deny=0.0.0.0/0.0.0.0 > permit=10.0.0.0/24 > username=user2 > secret= > disallow=all > allow=alaw > > If a call comes in from 10.0.0.250, using username "user2" and with no > password, then it is correctly authenticated against the [user2] > section. > Accountcode is set to user2 > Context is set to context2 > and the call mostly proceeds correctly, BUT the source channel name is > set to IAX2/user1-, which is then seen both in the dialplan debug > output, and in the CDR. I would expect the channel name to reflect the > section name that was used to authenticate the call ie. > IAX2/user2-; I specifically put a password onto [user1] so there > is no possibility that the call is authenticating there. > > Am I missing something? Or is this a bug? > > Thanks, > 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] [1.4] "show channels" in extensions.conf?
On Sat, 19 Feb 2011, Gilles wrote: I was wondering if there were a way to use NoOp/Verbose to display the output of the CLI's "show channels" from within extensions.conf? Ugly: system(asterisk -r -x 'show channels') Elegant: AGI using AMI. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 script dies after receivefax
On Fri, 18 Feb 2011, Mike Diehl wrote:
I've got a perl agi script that exec()'s the FFA version of receivefax to...
receive a fax.
However, after the fax is received, the script seems to die.
This is what I have:
$main::agi->exec("receivefax","/tmp/${$}.tiff|fs");
$main::agi->verbose("FAX COMPLETE",1);
I never see the "FAX COMPLETE" message on the console, I've set verbose to 25.
Any ideas? I'd like to take the next few instructions to log
success/failure.
What version of Asterisk?
Would enabling AGI debugging on the console shed any light?
Any chance receivefax is generating a signal you're not trapping?
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
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Re: [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?
On Fri, 18 Feb 2011 09:19:01 -0800 (PST), Steve Edwards The action and username lines were followed by pressing . The secret line was followed by pressing . On Fri, 18 Feb 2011, Gilles wrote: Thanks for the tip. I figured this out after a while ;-) I can now successfully log on, although I have to type the whole set twice: Odd. I don't even using telnet which is not a 'transparent' connection. Netcat (nc) would probably make a better took for testing your understanding of the protocol. I'd resolve the 'twice' bit before going further. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Newbie´s question about Asterisk...
(Please don't top-post and please trim posts that are no longer relevant.) On Fri, 18 Feb 2011, Francisco Javier Cintrón Olguín wrote: I think I have 3 PSTN lines because I can connect a normal telephone to them all and make calls between each of them. We have 5 normal telephones and 1 panasonic. From what I got I need a PC and a of PCI card to interface to my 3 external lines and my 6 internal lines. For the PC I was planning to use the smallest PC posible like a HP Proliant Microserver but it doesn´t have space for this PCI card. Is there another way to interface to 3 external and 6 internal lines?? There re USB and Ethernet devices (Xorcom, Sangoma, Sipura/Linksys/Cisco, and others) that can interface analog phones to your Asterisk server. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] [1.4/AGI] CHANNEL STATUS never "down & available"?
On Fri, 18 Feb 2011, Gilles wrote: I'm not having much luck with AMI: After typing the right commands, it just stays there, not replying to the Login action: = Telnet to TCP5038 Action: Login Username: admin Secret: secret It's waiting for another . This is from a 1.2 box, but it should look like this: -t2::sedwards:~$ telnet localhost 5038 Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Asterisk Call Manager/1.0 action: login username: example secret: example Response: Success Message: Authentication accepted The action and username lines were followed by pressing . The secret line was followed by pressing . -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] [1.4/AGI] CHANNEL STATUS never "down & available"?
On Fri, 18 Feb 2011, Gilles wrote:
I'm using an AGI script in Lua to make a callback through Zaptel.
=== AGI script
#!/var/tmp/lua
for i=1,10 do
io.write("CHANNEL STATUS\n")
response=io.read()
_, _, key, value = string.find(response, "(%a+)=(%d+)")
--Channel never "down and available"!
if value=="0" then
io.write("NOOP Channel idle\n")
response=io.read()
else
io.write("NOOP Channel N.A.\n")
response=io.read()
end
os.execute("/bin/sleep 2")
end
I'm just a 1.2 Luddite...
I've never written an AGI in lua, but don't you have to read the AGI
environment (from STDIN) before issuing requests?
Also, you execute your AGI in the 'h' extension. I think once a channel is
hung up, it's state will not change until you reach the end of your
dialplan execution and the channel is destroyed.
I'm guessing you would have better luck kicking off an external process
that checks the channel status via AMI.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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Re: [asterisk-users] Barge in.
On 16 February 2011 10:13, Peter den Hartog wrote: > I'm running Asterisk 1.6 and was wondering if anybody have a workig "barge > in" solution running. > I was thinking of using chanspy, but i would like that the original call > would be dropped, and the new call would be the only one there. What you are describing looks to me like a third party controlled transfer, and not a barge-in at all. I suspect that the Asterisk Manager API action "Redirect" will be your friend. Regards, 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] Aastra phones cannot transfer calls?
On 16 February 2011 00:22, Ernie Dunbar wrote: >> At 12:12 PM 2/15/2011, you wrote: >>>I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk >>>v1.6.2.1. They can call each other's extensions (and make and receive >>>calls otherwise), but they cannot transfer calls, not even to outside >> >> I'm running 1.6.2.16.1 and have three Aastra 480i phones and have had >> no problem at all with transfers. Have you considered trying a newer >> version? >> > > Nope. Upgraded to 1.6.2.16.1, and I still see the same effect. > > It may be a setting on the phone or a SIP setting. I'll investigate this > elsewhere but report back about the solution. > > I also tried this with a 6757i and a 6753i with no problems (blind and attended) on Asterisk 1.6.2.16.1. Have you updated the handset firmware to 2.6.0.2010? Cheers, 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] Hide the plain text password
On 15 Feb 2011, at 13:17, Richard Kenner wrote:
> Of course not! It would be useless if that were the case: the whole
> point here would be that you need the master encryption key.
>
> Here's a possible design:
>
> - There's optionally a file in the config
> directory called "master_key". It contains just a string.
>
> - A CLI command "core encrypt " is added to Asterisk. It takes the
> provided string, encrypts it using the string in master_key, and outputs
> a string of the form "{enc:
> - The config file reader looks for strings of the form "{enc:}:
> and replaces them, before otherwise parsing the line, with the decrypted
> version of the string using the key in the "master_key" file.
Let us know when you've made the patch..
S
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Re: [asterisk-users] Fax Woes
On Mon, Feb 14, 2011 at 11:00 PM, Mike Diehl wrote: > > Hi all, > > I'm trying to get outgoing fax working from an SPA2102 to a PSTN fax machine > via a T.38 enabled trunk. I've got > t38pt_udptl = yes > faxdetect=no > > in my sip.conf file. The ATA has all of the T.38 options turned on, echo > cancellation is off, as well as silence suppression off. The only > configured codec is u711. > > When the user tries to send a fax, it gets to the point where it issues a > reInvite to start the T.38, then the called side receives a SIP 488 (Not > Acceptable Here) > > Where should I start? Any pointers would be most welcome. > > Take care and have fun, > Mike Diehl. > -- > Did you turn on sip debugging? I bet you are sending a DID that your provider is not the RespOrg. Some will not take outbound calls with a tollfree number. I screws up the billing when one toll free calls another, who pays for the minutes? So they block it out and if memory serves me correctly, you would get the same message in Asterisk and the call would not go through. It is a shame because if you have toll free DIDs, you probably want them to show up rather than a toll call. I have seen this with XO, you have to plead your case and push alot of minutes for them to break their rules. If your issue is the same as what I had, you have to set the callerid to a number that has been ported over to that provider. It is too bad because there are many legit reasons to send a number that does not belong to you.. Call forwarding to your cell phone will not work, you will just see the office calling, not the actual caller. I have used it as a GUID, starting at 01 and increment by one, this was in a call center environment that had many people working in Baltimore, MD, Pakistan. Bogotá, and the Philippines. So a call would come in on a Spanish speaking DID, it would be sent to Bogotá with the callerid of whatever I sent. At the end of the month, when it was time to settle with the remote call centers, we had a way to really audit the bill rather than just paying it. The destination, GUID, CDRs all stored in a database. We also recorded using Orecx and and tied that into the same database and had full integration with the home brewed CRM. 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] uptime
On 15 Feb 2011, at 03:39, Jeff LaCoursiere wrote: > minipbx*CLI> show uptime > System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds > Last reload: 8 hours, 3 minutes, 51 seconds What's the highest current 'genuine' one on-list?.. klein*CLI> core show uptime System uptime: 2 years, 1 week, 4 days, 21 hours, 52 minutes Last reload: 41 weeks, 6 days, 16 hours, 6 minutes, 39 seconds That's the best I can come up with.. 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] Hide the plain text password
On 14 Feb 2011, at 22:30, Jian Gao wrote: > I am thinking using MySQL DB to save the user account information. And let > mysql encrypt the password, (MD5 maybe?). I remember I've done SIP realtime > registration. Can I also use this way on the Google Voice account? If you hash the password, you cant 'un-hash' it to send to Google. It'd have to be two-way encryption.. Which is pointless because as Kevin says, someone will just reverse it. 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] IP ban list by country
On Mon, 14 Feb 2011, Bruce B wrote: What sources do you use to limit SIP connecting customers to specific countries by IP (e.g. allowing USA and not China). It would help me a lot of you can note the sources you trust that are complete and up to date. I compiled this list a few (6?) months ago by typing class A address blocks into Arin.net's 'whois' web page and noting which Regional Internet Registry it was allocated to. http://www.voip-info.org/wiki/view/allocated-class-a-ip-address-blocks After plonking this into a couple of production hosts, attacks of all ports dropped dramatically. I note there have been changes since then (128.0.0.0 was assigned to RIPE back in November), so if anybody wants to 'refresh' and post changes, please do. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] On-Hold Music
Un-top-posting, but continuing to abuse the hijacked thread... > On Sat, 12 Feb 2011, ayodele abejide wrote: > > > I am having problems playing files with the playback command, also with > > the Dial (A()) option this is the output from console: > > > > This is the dialplan: > > > > exten => 1003,n,Playback(home/abejide/Desktop/a.wav) On Fri, 11 Feb 2011, Steve Edwards wrote: > Don't specify the file type. On Sat, 12 Feb 2011, ayodele abejide wrote: I tried what you suggested and this is the console output: [Feb 12 03:18:41] WARNING[2774]: file.c:650 ast_openstream_full: File /var/lib/asterisk/sounds/home/abejide/Desktop/a does not exist in any format Does the following shell snippet yield any clues? for F in /var/lib/asterisk/sounds/home/abejide/Desktop/a* do file $F done -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] On-Hold Music
On Sat, 12 Feb 2011, ayodele abejide wrote: I am having problems playing files with the playback command... And don't hijack other people's threads :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] On-Hold Music
On Sat, 12 Feb 2011, ayodele abejide wrote:
I am having problems playing files with the playback command, also with
the Dial (A()) option this is the output from console:
This is the dialplan:
exten => 1003,n,Playback(home/abejide/Desktop/a.wav)
Don't specify the file type. Asterisk will try to find a file encoded and
formatted to match the encoding of the channel. Failing a match, Asterisk
will try to find a file it can transcode to match the encoding of the
channel.
Since you specified a relative path ('does not start with a slash'),
Asterisk will prefix (by default) '/var/lib/asterisk/sounds/' to your
path yielding:
/var/lib/asterisk/sounds/home/abejide/Desktop/a.*
is this what you want?
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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Re: [asterisk-users] On-Hold Music
On 11 Feb 2011, at 22:37, Danny Nicholas wrote: > In 500 words or less (if possible), please explain what is a legal > music-on-hold file? Depends on the country, and what licence you posses. Googling ' hold music regulations' may help. 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] dialplan announcements
On Fri, 11 Feb 2011, ERIC HERRON wrote: I want to have an option off the IVR that plays back the announcement for the day. At the end of the message, I want the caller to get kicked back to the previous menu. The conditions are that I want the recorder to dial a feature code that prompts him to record the message. He then presses 1 to accept. This gets saved as announcement.wav. The record() and background()/playback() applications should be appropriate based on the level of detail you have supplied. Show us what you have so far. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] meetme conference & playback of random sound file
On Thu, 10 Feb 2011, John Jolly wrote: i am trying to configure the meetme conference (asterisk 1.8) to play a random sound file from a specific directory prior to it dropping the caller into the conference itself. i am able to successfully get it to play a specific file prior to entering the conference unsure how to implement this sort of randomization. Who is the sound file played to? The caller or the conference? Please show what you are using now. Would an AGI that selected a random file from the directory and set the path as a channel variable work? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Defining what an extension should do after the Dial() command returns busy.
Un-top-posting...
On Wed, 9 Feb 2011, Ernie Dunbar wrote:
We have a customer who wants to forward an extension to their cell
phone, if and only if that extension is "unavailable", or when the
Dial() command times out. However, should the Dial() command return
"busy" it should go to voicemail instead.
On Wed, 9 Feb 2011, Danny Nicholas wrote:
Perhaps your "googling" skills need some management - look for S-BUSY,
S-NOANSWER.
Here's a snippet that might do what they want
- exten => s,1,Dial(DAHDI/1/5551212,30)
- exten => s,n-BUSY,voicemail(blah)
- exten => s,n-UNAVAILABLE,Dial(DAHDI/1/5552323,30)
- exten => t,1,Dial(DAHDI/1/5552323,30) Cell
On Wed, 9 Feb 2011, Ernie Dunbar wrote:
It's nice to know that you've tried this and are presenting me with a
proven solution.
FYI, this doesn't work. Neither do any of the following variations:
Off the top of his head, Danny put you into the 'ballpark,' a little bit
more googling on your part would have brought you home.
Off the top of my head, the missing step is using the DIALSTATUS returned
by the dial() application as the target of a goto. Like:
exten = s,n,goto(s-${DIALSTATUS},1)
and then your dialplan should include extensions like:
exten = s-BUSY,1, verbose(1,[${EXTEN}@${CONTEXT}])
exten = s-BUSY,n, ...
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] queue called by agi doesn't re-enter the script
On Wed, 9 Feb 2011, gincantalupo wrote: I tried this piece of extensions on my Asterisk 1.8: exten => 679,1,NoOp(start) exten => 679,2,AGI(/var/lib/asterisk/bin/test.py) exten => 679,3,NoOp(--- end ---) exten => 679,n,Hangup where test.py executes a queue command. The strange thing is my CLI never shows the '--- end ---' string. It seems that with queues, the normal script flow is not going on to the next step...just like the queue forces an exit from extension.conf. What queue command does test.py execute? Did you use a tested 'py' AGI library or write your own? (Nobody gets it right the first time.) If you crank up verbosity and debug do you get any clues? The CLI command 'agi set debug on' may also yield some clues. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] forward calls by the ports
On 8 Feb 2011, at 14:52, mehran khajavi wrote: > i searched a lot but i couldn't find the answer . > i have two openvox(fxo/fxs) card so I have 24 ports! Ok! > on first card i have 12 fxs and on the second i have 12 fxo > i want to then one person calling from dahdi/13 forward it to dahdi/1 > when a person calling from dahdi/14 forward it to dahdi/2 > when a person calling from dahdi/15 forward it to dahdi/3 > > how can i do this? You dont need a PBX for that... Just plug the phones into the line?.. > i should make an AGI? or can i make it with extentions.conf? how can i get > the caller's port number? You could do either. extensions.conf is more sensible. Put ports in different contexts / use channel variables. How to do this is probably in the extensive documentation you've been studying. 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] ${HANGUPCAUSE} in CDR
On 8 Feb 2011, at 13:30, Shariq Khan wrote:
> Can i add ${HANGUPCAUSE} in CDR after the Dial command using h extension? I
> want to add the Hangup reason of call in userfield of CDR.
http://www.google.com/search?q=asterisk+hangupcause+cdr
Top result... Should do it
Steve
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[asterisk-users] IAX channel name incorrect - Found in 1.2 still happens in 1.6
Hi, The following IAX config (slightly edited) causes an issue for me in version 1.6.2.16.1, where my CDR data is unreliable. [user1] type=friend auth=md5 accountcode=user1 notransfer=yes context=context1 host=10.0.0.250 username=user1 secret=secret1 disallow=all allow=alaw [user2] type=friend auth=md5 accountcode=user2 notransfer=yes context=context2 host=dynamic deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/24 username=user2 secret= disallow=all allow=alaw If a call comes in from 10.0.0.250, using username "user2" and with no password, then it is correctly authenticated against the [user2] section. Accountcode is set to user2 Context is set to context2 and the call mostly proceeds correctly, BUT the source channel name is set to IAX2/user1-, which is then seen both in the dialplan debug output, and in the CDR. I would expect the channel name to reflect the section name that was used to authenticate the call ie. IAX2/user2-; I specifically put a password onto [user1] so there is no possibility that the call is authenticating there. Am I missing something? Or is this a bug? Thanks, 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] Callback through extensions.conf?
On Mon, 7 Feb 2011, Steve Edwards wrote: sudo /usr/sbin/asterisk -d -d -d -n -v -v -v Oops. A '-c' should be in there :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Callback through extensions.conf?
On Mon, 7 Feb 2011 04:06:52 -0600, Sherwood McGowan wrote: ok, first of all, it can take a little while for those spooled callfiles to be executed in Asterisk... On Mon, 7 Feb 2011, Gilles wrote: Thanks for your help. The same callfile works fine in Ubuntu, but not at that appliance. Since I can dial through the FXO, it doesn't seem to be a Zaptel issue either. I'll investigate further, and find a work-around if the appliance just doesn't support this feature for some reason. Like maybe pbx_spool.so not being loaded? Bump up the logging in logger.conf, verbose and debugging in the CLI and see if you can get any clues. Another useful exercise is to start Asterisk like: script startup-log sudo /usr/sbin/asterisk -d -d -d -n -v -v -v exit and then read every line of startup-log. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Any voice changer applications for Asterisk?
On 02/06/2011 05:05 PM, Sherwood McGowan wrote: AAhem. https://wiki.asterisk.org/wiki/display/AST/Function_PITCH_SHIFT Granted, it's in 1.8, but it's in the documentation ;-) Cheers That seems to do exactly what the Lobstertech code does. What do people use this for? The Lobstertech one was a fun toy, but seems to be of no practical use. Changing female to male, child to adult, etc. seems pretty useful, but these modules make no attempt to perform a meaningful voice change. They would need to control the formants independent of the pitch to produce anything like a plausible voice adjustment. On Sat, Feb 5, 2011 at 9:44 PM, Steve Underwood <mailto:ste...@coppice.org>> wrote: On 02/06/2011 05:39 AM, Bruce B wrote: Hello, Are there any other other voice changer applications to Asterisk other than the one from Lobstertech? (http://lobstertech.com/voice_changer.html) Specifically interested in open-source but can have a look at economical commercial alternatives as well. It might help if you explained the kind of change you would like to make, which the lobstertech module doesn't offer. Steve 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] Can a duration limit be specified in spool call file?
On Sun, 6 Feb 2011, Bruce B wrote: Can you you please explain the Local Channel concept. I am not sure what should be the Channel line: Channel: xxx/yyy/ Gosh. This was the first result returned by googling 'asterisk local channel.' http://www.voip-info.org/wiki/view/Asterisk+local+channels While there is a lot of out of date crap out there, www.voip-info.org is still a valuable resource. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Any voice changer applications for Asterisk?
On 02/06/2011 05:39 AM, Bruce B wrote: Hello, Are there any other other voice changer applications to Asterisk other than the one from Lobstertech? (http://lobstertech.com/voice_changer.html) Specifically interested in open-source but can have a look at economical commercial alternatives as well. It might help if you explained the kind of change you would like to make, which the lobstertech module doesn't offer. 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] MP3 Crashing Asterisk
On Fri, 4 Feb 2011, Timothy Smith wrote: I have a problem with some of my mp3 files. they crash the system (Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to play them. Unfortunately the logs do not give me a clear fault or cause of crash but i can clearly see that ts because of the MP3 files. Read up on how to create a crash dump and submit a bug report. A 'bad' file shouldn't crash Asterisk. I also tried converting the files to wav or sln but there is severe music quality loss. By converting to MP3, some would say the 'music quality' has already been lost :) I convert MP3s with the following: mpg123 -q -w example.mp3.wav example.mp3 sox example.mp3.wav -c 1 -s -w -r 8000 example.wav normalize example.wav If this doesn't help, can you post links to the MP3 and the WAV? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Monitor() in Python AGI
On Tue, 1 Feb 2011, Felix Dong wrote:
How can I use the application Monitor() in the Python AGI skripts?
Use the exec AGI command.
I use C so it looks something like this:
exec_agi("exec MONITOR wav|%s/%02d-prompt|m"
, recording_path
, idx
);
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Newbie Question...
On Mon, 31 Jan 2011, Piotr Górski wrote: I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of free calls from each of 4 pstn lines... Can I configure Asterisk to call thru pstn line that has free minutes? For example Outgoing calls are going through PSTN 1 for 60 minutes. When I use all of these free minutes - outgoing calls go thru PSTN 2. When I use all free minutes from PSTN 2 outgoing calls go via PSTN3. You will need to keep track of the call duration for each channel in a persistent store -- something like MySQL. You may also want to read up on setting the absolute timeout on a channel so a caller won't consume all of your 'prepaid' (nothing is free) minutes and drive you into unexpected charges. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Calling Directory app from AGI
On Mon, 31 Jan 2011, Mike Diehl wrote:
I've got an agi script that calls the directory function, which seems to
work to a point. However, once the caller has selected an entry, I need
my agi script to find out which extension was selected. I've RTFM'd and
don't see that the extension is returned. Nor is a variable set, as far
as I can see.
Is there a way to get this information from the directory application?
No channel variable is set, but it would be a simple modification to the
source code to return the extension instead of dialing it.
Another course of action would be to break your AGI into 2 parts and then
'catch' the exten in the 'dial-context.' For example, assuming you are
using the 'default' voicemail context and the 'directory-test'
dial-context, you would execute the directory application like (1.2):
exec_agi("exec directory default|directory-test");
and then in your dialplan you would have something like:
[directory-test]
exten = _!.,1, verbose(1,[${EXTEN}@${CONTEXT}])
exten = _!.,n, agi(part-two,--extension=${EXTEN})
exten = _!.,n, hangup()
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
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Re: [asterisk-users] faxter
On 30 Jan 2011, at 09:21, Pezhman Lali wrote: > Faxter is an opensource email to fax gateway, > please check it, let me know if any bug. Only bug i can see is the attitude of the developer... As for the bugs, having the config variables liberally scattered throughout the script makes it's use (and then subsequent update) near impossible. There are even context names towards the end of the file. Ideally you'd want a separate config.php which you then include from your main script. A readme would then document what you'd put in here (and their default values if you dont). The tabbing is pretty random, and the commented out test data is pretty decorative. chmod($save_dir.$filename,0777); Is a slightly interesting idea. Not actually run it to see if it works, wouldn't know how.. 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
