Re: [asterisk-users] Callback through extensions.conf?
Gilles, Nice! That was some good reading! On Tue, Feb 8, 2011 at 6:01 PM, Gilles wrote: > Interesting... > > http://en.wikipedia.org/wiki/Inotify > > http://blackfin.uclinux.org/gf/project/uclinux-dist/forum/?action=ForumBrowse&forum_id=39&_forum_action=ForumMessageBrowse&thread_id=33403 > > > -- > _ > -- 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
[asterisk-users] dial option 'g' not working
Hi, I'm trying to get my dialplan to continue executing in the current context after a third-party is called and hangs up. It seems like it should be straightforward but it's not working. Here's what I have in extensions.conf: exten => 333,1,Answer() exten => 333,n,Playback(hello) exten => 333,n,Dial(SIP/1999222@sipcarrier,,g) exten => 333,n,Playback(hello) exten => 333,n,Playback(hello) exten => 333,n,Playback(hello) exten => 333,n,Hangup() The 999222 number is dialed, but after that party hangs up, there's just dead air. No hello's are played and nothing seems to be happening. What am I doing wrong? Thanks, MS -- _ -- 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] Manual Call Transfer (Perl, Asterisk::AGI, MySQL)
Hello Everyone!
I've hit a bit of a roadblock and I am hoping that someone might point
me in the right direction.
I am using Asterisk 1.2.4 - I do not have the option of updating it,
please do not waste your time telling me to =)
I am using PERL AGI scripts to maintain an "active calls count" field
for each phone in a mysql database table, for example (not actual
code, just trying to illustrate)
$SIG{HUP} = 'IGNORE';
mysql_update_call_count($user_id, ($count +1) );
$dialret = $agi->exec('Dial', $dialstring);
mysql_update_call_count($user_id, ($count -1 ));
(ignore the count this, did that for clarity)
This works great, except when doing assisted transfers (or any
transfer for that matter).
We have Polycom IP550 Phones which can do the transfer with a button,
As an example of this process and the problem, and assuming these are
all internal phones dialing extensions...
phone A dials phone B
phone B presses transfer to transfer phone A to phone C
phone B hangs up
Because the Dial command in the AGI script executed when phone A
called phone B is still running the active call count remains at 1 for
phone B until the call between A and C ends (at which point they all
zero out).
I also tried using atxfer to resolve this problem and got a different behavior
phone A dials phone B
phone B presses *2 then phone C's extension to transfer phone A to phone C
phone B hangs up
an active call count remains at 1 for B and C but A drops to 0 count.
Might be worth mentioning the possibility that phone B is already on
the line when the call from phone A comes in.
I thought one possible solution might be creating an [applicationmap]
that essentially handles the assisted transfer manually. I've done a
great deal of reading on this matter and aside from the fact that I'm
still a bit fogy as to how i would even do that,.. it seems that there
is still no way for me to determine who is being transferred when the
second channel is opened (new uniqueid / agi script execution).
Is there perhaps something I am missing which would help resolve this?
I hope that I've explained my problem clearly. I have only been
tinkering with asterisk for about a week so I apologize if I'm not
using the appropriate vernacular.
Thank you!
-Ted
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Re: [asterisk-users] Callback through extensions.conf?
Interesting... http://en.wikipedia.org/wiki/Inotify http://blackfin.uclinux.org/gf/project/uclinux-dist/forum/?action=ForumBrowse&forum_id=39&_forum_action=ForumMessageBrowse&thread_id=33403 -- _ -- 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] Un message de Mickael t'attend...
Un message de Mickael t'attend... L'expéditeur et le contenu seront visibles seulement par toi et tu peux le supprimer à tout moment. Tu peux aussi y répondre directement au travers du messenger. Pour découvrir qui est à l'origine du message, suis simplement ce lien: http://eu1.badoo.com/0199422682/in/pe-wgHsDEkQ/?lang_id=6 D'autres personnes sont aussi présentes: Calu (Maputo, Mozambique) Nadia (Tunis, Tunisie) Juventino (Tunis, Tunisie) Pollox (Valencia, Espagne) Yawar (Linköping, Suède) ...Qui d'autre? http://eu1.badoo.com/0199422682/in/pe-wgHsDEkQ/?lang_id=6 Les liens ne fonctionnent pas dans ce message? Copie les dans la barre d'adresse de ton navigateur. Tu as reçu cet email suite à une requête de Mickael sur notre système. S'il s'agit d'une erreur, ignore simplement cet email. La requête sera alors effacée du système. Merci, L'équipe Badoo Courrier automatique de Badoo suite à l'envoi d'un message à ton attention sur Badoo. Les réponses ne sont ni stockées, ni traitées. Si tu ne veux plus recevoir de message de Badoo, fais-le nous savoir: http://eu1.badoo.com/impersonation.phtml?lang_id=6&mail_code=63&email=asterisk-users%40lists.digium.com&secret=&invite_id=87167&user_id=199422682-- _ -- 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] Microsoft Speech Server/UCMA Integration
Hello All, I was wondering if anyone's tried to use OR currently use the Microsoft Speech Server or their UCMA 3.x SDK etc. as their ASR/TTS backend/engines etc. If yes, then what's their experience? Please Note, this does NOT need to be integrated with Asterisk ala MRCP or some module/plugin etc. I just wanted to know if someone's used it and and what their experience has been in both, TTS and ASR. Thanks \RR -- _ -- 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] echo when calling to the pstn
Hello all. I have a asterisk implementation (asterisknow 1.7.1), with a card with 2 FXO interfaces. When I call (or receive a call) from the pstn, I ear echo. This happens if I use a softphone or IP phone, and does not happens if the call is internal. Can you help me with this issue? Best regards, Vitor Flausino -- _ -- 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] Manual Call Transfer // Perl // Asterisk::AGI // MySQL
Hello Everyone!
I've hit a bit of a roadblock and I am hoping that someone might point
me in the right direction.
I am using Asterisk 1.2.4 - I do not have the option of updating it,
please do not waste your time telling me to =)
I am using PERL AGI scripts to maintain an "active calls count" field
for each phone in a mysql database table, for example (not actual
code, just trying to illustrate)
$SIG{HUP} = 'IGNORE';
mysql_update_call_count($user_id, ($count +1) );
$dialret = $agi->exec('Dial', $dialstring);
mysql_update_call_count($user_id, ($count -1) );
This works great, except when doing assisted transfers (or any
transfer for that matter).
We have Polycom IP550 Phones which can do the transfer with a button,
As an example of this process and the problem, and assuming these are
all internal phones dialing extensions...
phone A dials phone B
phone B presses transfer to transfer phone A to phone C
phone B hangs up
Because the Dial command in the AGI script executed when phone A
called phone B is still running the active call count remains at 1 for
phone B until the call between A and C ends (at which point they all
zero out).
I also tried using atxfer to resolve this problem and got a different behavior
phone A dials phone B
phone B presses *2 then phone C's extension to transfer phone A to phone C
phone B hangs up
an active call count remains at 1 for B and C but A drops to 0 count.
Might be worth mentioning the possibility that phone B is already on
the line when the call from phone A comes in.
I thought one possible solution might be creating an [applicationmap]
that essentially handles the assisted transfer manually. I've done a
great deal of reading on this matter and aside from the fact that I'm
still a bit fogy as to how i would even do that,.. it seems that there
is still no way for me to determine who is being transferred when the
second channel is opened (new uniqueid / agi script execution).
Is there perhaps something I am missing which would help resolve this?
I hope that I've explained my problem clearly. I have only been
tinkering with asterisk for about a week so I apologize if I'm not
using the appropriate vernacular.
Thank you!
-Ted
--
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[asterisk-users] Scheduled Maintenance: wiki.asterisk.org and code.asterisk.org
On Thursday, February 10, 2011 at 8:00AM CST (GMT-5), two servers that provide community services will be upgraded with new software releases: * wiki.asterisk.org will be upgraded to Confluence 3.4.8. This upgrade should take less than 20 minutes. * code.asterisk.org will be upgraded to Crucible+Fisheye 2.5.0. The actual upgrade will take less than 20 minutes, but the entire set of repositories serviced by Fisheye will need to be re-indexed, which could take anywhere from 2 to 8 hours. During this time, portions of code.asterisk.org may display incomplete contents. -- _ -- 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] Polycom Blf / Directed Pickup
Hi, For future reference, it might be useful to notice (from SIP 3.1 Admin Manual): " attributes are only available to SoundPoint 320/330, 430, 550, 560, 600, 601, 650 and 670 phones only". For a 3.1.3-enabled 501, has someone been able monitor a third status beyond Idle, OnCall ones ? I can successfully see that an extension is idle but as soon as it receives an incoming call, it status is immediately changed to OnCall : I can still dial a *8 sequence to pickup the call but I can't do anything more. The strange thing is I can add a line like call.directedCallPickupString="*8" is config file but I can't see how I can use it (with a 501). Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for actual user opinions on Telephony card
Hello all, Just hoping to get some opinions from folks that have actually used the Rhino R4FXO-EC. Looking for user experiences, good or bad. This looks like a nice unit and I have a need for exactly this config, 4FXO and EC TIA, JohnM -- _ -- 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] Set variable on Call Answer
Thanks, I will check our that. It seems M macro would work. -dani On Tue, Feb 8, 2011 at 7:02 AM, Sherwood McGowan wrote: > the M option in your Dial command will execute a macro upon connection, > there's also an option to perform a Gosub... > > http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial > > ;-) > > *keeps his "mailing-list police" badge in it's box in his office* > (that wasn't directed at you Dan...there was a little flamewar that I > stirred up the other day..that was my troll bit for the day) > > Check out that link, or run > core show application dial > from the Asterisk console..look at the options list and find the Macro > reference and the Gosub reference...they should light a candle for ya :D > > On Tue, Feb 8, 2011 at 8:36 AM, Dan Dan wrote: > >> Hi All, >> >> First post here. I am dialing out via call file to remote number, when >> call is connected a local number is dialed. And on success both calls get >> bridged and works fine. >> >> This is a parallel auto dialout application. I want to set a variable as >> soon as the local number answers the call, so that system won't try to >> dialout that local number again and stops further dialing. What should be >> the best way to deal this situation ? >> >> Any help would be appreciated. >> >> Thanks >> -dani >> >> >> -- >> _ >> -- 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 > -- _ -- 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] Call Recording audio file quality query
On Tuesday 08 February 2011 06:34:56 Sherwood McGowan wrote: > On Tue, Feb 8, 2011 at 6:01 AM, wrote: > > But if you are getting calls all the way on VoIP then you can have > > calls in HD audio using HD audio codec on all locations (Server and > > Client). In that case you either need use some available 3rd party > > solution which uses packet capturing to trace the calls and record > > call using packet capture and assembling regardless of server as > > asterisk still will not be able to record call in HD but some other > > switches like FreeSWITCH can do it or you need to write your own app > > like it. > > It's not difficult at all to perform what you're referring to..If you > have the hardware... > > A simple way is to have a port on your main network switch/router that > will "firehose" the traffic the device interacts with In case someone > reading this doesn't know, I'm talking about having a port that just > makes a copy of EVERY PACKET that the device "sees" and sends those > copies out over the port that you've set up for the purpose..It just > GUSHES data over that port...like a firehose just gushes out all the > water it possibly can... LOL > > Anyway, once your data is being mirrored over that firehose, send it to > a dedicated "recording" server...all it has to do is find the signaling > packets for each call and then just dump the "payload" from the RTP. > It'll come out exactly as it was transported within RTP...in the codec > the call set up > > I may be wrong, but I'm fairly sure that Asterisk can write a filetype > for almost any of it's codecs...I know it can READ audio files that are > encoded in GSM, uLaw (ul), aLaw (al), G726 and G729 formats (.g729, > g.726)...etc... > > If the "DECoding" portion is there, there's almost GOT to be the > "enCOding" functionality... Actually, the writing of encoded voice has nothing to do with codecs. The format modules simply expect a particular type of packet to be fed in, and they simply reformat the audio (without transcoding) to be stored on disk. One caveat is that the format in which they are stored on disk is not guaranteed to be a standard format that is at all useful to outside utilities; just that Asterisk can read it off disk and reassemble the packets. -- Tilghman -- _ -- 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] terrible MeetMe sound with 1.6.2.9
On Tue, Feb 08, 2011 at 11:09:19AM -0600, Warren Selby wrote: > On Tue, Feb 8, 2011 at 11:04 AM, Louis-David Mitterrand < > vindex+lists-asterisk-us...@apartia.org> wrote: > > > Forgot to add that our MOH sounds fine when listened to (on the same > > extension as MeetMe) with MusicOnHold(default). So it's not a MOH > > problem as speakers in the MeetMe conference are affected too. > > > Do you have DAHDI installed and running? Yes, all our calls come through a dahdi device. The calls sound fine. Only MeetMe is affected it seems. > Show us the output of dahdi_test from the command line. Opened pseudo dahdi interface, measuring accuracy... 99.999% 99.998% 99.995% 99.995% 99.999% 99.992% 99.998% 100.000% 100.000% 99.996% ^C --- Results after 10 passes --- Best: 100.000 -- Worst: 99.992 -- Average: 99.997198, Difference: 99.998508 -- _ -- 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] terrible MeetMe sound with 1.6.2.9
On Tue, Feb 8, 2011 at 11:04 AM, Louis-David Mitterrand < vindex+lists-asterisk-us...@apartia.org> wrote: > Forgot to add that our MOH sounds fine when listened to (on the same > extension as MeetMe) with MusicOnHold(default). So it's not a MOH > problem as speakers in the MeetMe conference are affected too. > > Do you have DAHDI installed and running? Show us the output of dahdi_test from the command line. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound SIP calls work, just not when making calls between extensions.
On Tue, Feb 8, 2011 at 11:02 AM, Ernie Dunbar wrote:
> Internal calls:
>
> exten => _312,1,Set(CALLERID(name)="Internal call")
> exten => _312,n,SIPAddHeader(Alert-Info: info=)
> exten => _312,n,Dial(SIP/username2,20)
> exten => _312,n,Voicemail(312,u)
> exten => _312,n,Macro(handle-hangup)
>
>
Try taking the quotes ("") out of the line that says "Internal call". So it
should be:
exten => _312,1,Set(CALLERID(name)=Internal call)
...and see if that helps.
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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Re: [asterisk-users] terrible MeetMe sound with 1.6.2.9
On Tue, Feb 08, 2011 at 10:59:47AM -0600, Danny Nicholas wrote: > Any idea? > > I use mpg123 to play my MOH so I can control the volume (my users complain > that standard MOH is a bit loud). Forgot to add that our MOH sounds fine when listened to (on the same extension as MeetMe) with MusicOnHold(default). So it's not a MOH problem as speakers in the MeetMe conference are affected too. -- _ -- 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] Inbound SIP calls work, just not when making calls between extensions.
This is a problem that is completely stumping me, and my understanding of
Asterisk dialplans tells me this should never be a problem. Moreover, this
scenario works on Asterisk 1.4 but not 1.6.
We have a customer with several Aastra 6731 phones. They want incoming
calls from the PSTN to work and they also want to be able to call each
other "internally" on a special non-DID number (like extensions 311, 312,
313, etc).
In the dialplan, both the extensions for their DID and their internal
extensions use the same Dial() command. The only difference that I can see
is that we make changes to the CallerID Name field and do a little dance
with SIPAddHeader() to make the Aastra phones ring differently. This
doesn't appear to have any effect on Asterisk, but when the call is made,
the phone responds back with "SIP response 400 "Bad Request"".
Here's the two dialplans (private details redacted):
Internal calls:
exten => _312,1,Set(CALLERID(name)="Internal call")
exten => _312,n,SIPAddHeader(Alert-Info: info=)
exten => _312,n,Dial(SIP/username2,20)
exten => _312,n,Voicemail(312,u)
exten => _312,n,Macro(handle-hangup)
Calls from the PSTN:
[Somecompany-IVR-day]
exten => s,1,Dial(SIP/username1&SIP/username2&SIP/username3,20)
exten => s,n,Goto(Somecompany-IVR-night,s,1)
The errors from Asterisk when internal calls are made:
-- Executing [311@somecompany:1] Set("SIP/username3-01b0",
"CALLERID(name)="Internal call"") in new stack
-- Executing [311@somecompany2] SIPAddHeader("SIP/username3-01b0",
"Alert-Info: info=") in new stack
-- Executing [311@somecompany3] Dial("SIP/username3-01b0",
"SIP/username1,20") in new stack
== Using SIP RTP CoS mark 5
-- Called username1
-- Got SIP response 400 "Bad Request" back from XX.XXX.XXX.X
-- SIP/username1-01b1 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [311@somecompany4] VoiceMail("SIP/username3-01b0",
"311,u") in new stack
--
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[asterisk-users] Asterisk CallCompletion dialplan
Hi Users,
I'm planing to implement call completion feature in asterisk 1.8 but having
some issue. I am following this document
https://wiki.asterisk.org/wiki/display/AST/Generic+Call+Completion+Example
I am getting error non-zero error on console. I am using softphone x-lite
root@tux:/etc/asterisk# asterisk -r
Verbosity is at least 3
== Using SIP RTP CoS mark 5
-- Executing [30@from-sip:1] CallCompletionRequest("SIP/7623-0013", "")
in new stack
== Spawn extension (from-sip, 30, 1) exited non-zero on 'SIP/7623-0013'
sip.conf
[Mark]context=phone_callscc_agent_policy=genericcc_monitor_policy=generic ;We
will accept defaults for the rest of the cc parameters;We also are not
concerned with other SIP details for this;example
[Richard]context=phone_callscc_agent_policy=genericcc_monitor_policy=generic
extensions.conf
[phone_calls]exten => 1000,1,Dial(SIP/Mark,20)exten => 1000,n,Hangupexten =>
2000,1,Dial(SIP/Richard,20)exten => 2000,n,Hangupexten =>
30,1,CallCompletionRequestexten => 30,n,Hangupexten =>
31,1,CallCompletionCancelexten => 31,n,Hangup
--
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Re: [asterisk-users] terrible MeetMe sound with 1.6.2.9
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis-David
Mitterrand
Sent: Tuesday, February 08, 2011 10:50 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] terrible MeetMe sound with 1.6.2.9
Hi,
Using MeetMee(1234,M) on asterisk 1.6.2.9 (debian/testing) we have
terrible sound: the MOH is unrecognizable and speakers can't be
understood; it sounds "ghostly". However the prompts ("your are the only
one in this conference, etc.") sound fine.
Our server has a Digium T410P card with two E1 lines going in and the
wct4xxp dahdi module.
Any idea?
I use mpg123 to play my MOH so I can control the volume (my users complain
that standard MOH is a bit loud).
--
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Re: [asterisk-users] About maxlen parameter in queues
On Mon, 2011-02-07 at 10:44 -0500, Daniel - Asterisk wrote: > Hi Danny, > > > Could you please let me know what function do I use to get if the > queue is full? > > > Elder > > On Mon, Feb 7, 2011 at 10:42 AM, Danny Nicholas > wrote: > > __ > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > Daniel - Asterisk > Sent: Monday, February 07, 2011 9:38 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] About maxlen parameter in queues > > > > > > Dear list, > > > > > I want to avoid sending calls to a queue when it is full. From > the fact that 'maxlen' must be at least 1 (I wish it could be > zero but it isn't) I'd like to know if there's a way to do it. > Setting the Queue() timeout to a little value is not the most > suitable option. > > > I'm using asterisk 1.4.21 but I don't know if there are some > options available on release 1.8 > > > > > > Thanks, > > > > > > > > > Elder Arohuanca Lagos > > > t. 992728100 > > > > This is a bit “hackish”, but why don’t you just make a context > that uses AGI to query the queue and only let the call proceed > if not full? > > > Maybe it would be easier to use the GROUP and GROUP_COUNT functions to see how many users are in the queue and decide on that. Although this really defeats the purpose of having a Queue. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] terrible MeetMe sound with 1.6.2.9
Hi,
Using MeetMee(1234,M) on asterisk 1.6.2.9 (debian/testing) we have
terrible sound: the MOH is unrecognizable and speakers can't be
understood; it sounds "ghostly". However the prompts ("your are the only
one in this conference, etc.") sound fine.
Our server has a Digium T410P card with two E1 lines going in and the
wct4xxp dahdi module.
Any idea?
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Re: [asterisk-users] fail-over server
On Tue, Feb 8, 2011 at 8:07 AM, Vieri wrote: > Suppose you have 2 identical Asterisk servers and 1 alias IP address that you > assign to either one, according to system failures, etc. > Also suppose that all SIP clients register requests go to the alias IP > address. This is a typical setup for two node HA. Just be careful when clustering only two servers. > Imagine server1 fails and server2 gets the alias IP address. > Correct me if I'm wrong but I would have to wait at least 60 seconds before > most SIP clients re-register to server2 and that server2 knows that they are > actually "on-line" so calls can be routed to them. It depends on your configuration. If you use Asterisk Realtime to store SIP registrations, then the database will contain information on how to contact the device (fullcontact, ipaddr, and port fields). Then on a failover, Asterisk will do a lookup for the peer in the database, find the needed information and dial the device. Of course any registrations that happen before being written right before the server fails may not work. Also make sure to use the latest version of Asterisk as there was a bug where fullcontact wasn't saved correctly. > How can I minimize this time lapse? Can Asterisk "notify" all SIP > clients in its sip.conf that they need to acknowledge being on-line > or not (thus forcing re-registration in my scenario)? In the above scenario, I can kill Asterisk, start it again, and place a call from two devices that have not registered again. So, the best timeout is your dead time detection and failover startup time. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail-over server
Hi, Thats very simple. Use sip realtime registration with mysql and heartbit to control switiching. Regards, Carlos M Cruz Em 2011/02/08 16:07, "Vieri" escreveu: Hi, Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to server2 and that server2 knows that they are actually "on-line" so calls can be routed to them. How can I minimize this time lapse? Can Asterisk "notify" all SIP clients in its sip.conf that they need to acknowledge being on-line or not (thus forcing re-registration in my scenario)? Thanks, Vieri -- _ -- 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] fail-over server
Take a look at High Availability ASTerisk (HAAST) from www.generationd.com Their software sits between the OS and asterisk, and can failover servers, switch IP addresses, control external interfaces, etc. It can run on different hardware (make a cluster from different/cheap boxes), it allows long distance seperation of cluster members, etc. Also, it's easy to install. Michelle (I'm affiliated with generationd so I may be biased, but I think the product is awesome) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Gergo Csibra [csi...@gmail.com] Sent: Tuesday, February 08, 2011 11:17 AM To: Asterisk Users List Subject: Re: [asterisk-users] fail-over server Tuesday, February 8, 2011, 5:07:29 PM, Vieri wrote: > How can I minimize this time lapse? Can Asterisk "notify" all SIP > clients in its sip.conf that they need to acknowledge being on-line > or not (thus forcing re-registration in my scenario)? If you have two identical servers online, it is better to make a HA sollution. Sorry, I haven't made HA Asterisk yet, I can not help more. -- Best regards, Gergomailto:csi...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] fail-over server
Tuesday, February 8, 2011, 5:07:29 PM, Vieri wrote: > How can I minimize this time lapse? Can Asterisk "notify" all SIP > clients in its sip.conf that they need to acknowledge being on-line > or not (thus forcing re-registration in my scenario)? If you have two identical servers online, it is better to make a HA sollution. Sorry, I haven't made HA Asterisk yet, I can not help more. -- Best regards, Gergomailto:csi...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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
[asterisk-users] fail-over server
Hi, Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to server2 and that server2 knows that they are actually "on-line" so calls can be routed to them. How can I minimize this time lapse? Can Asterisk "notify" all SIP clients in its sip.conf that they need to acknowledge being on-line or not (thus forcing re-registration in my scenario)? Thanks, Vieri -- _ -- 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 registration
Hi, Are sip.conf's defaultexpiry and maxexpiry global? Or can they be used on a per-extension basis? I'd like to "force" some extensions to re-register more frequently than others (server-side). Thanks, Vieri -- _ -- 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?
Thanks Faisal. That is it. I was confused by the fact that there is also the Context, Extension, and Priority in the .call file that should be filled along with the Channle: local. I found out that the call file first calls the local channel context and once that is connected then it moves onto the second context that is defined in subsequent the variables. Indeed this was what was throwing me off. -Bruce On Tue, Feb 8, 2011 at 1:57 AM, wrote: > Hi, > > If you need full control on both legs of call you can redirect Leg-1 to > your dialplan as Channel: > Local/your-extension@your-context/n and > from there you control the Leg-1 using dial-plan or AGI as you like while > Leg is normally comes to dialplan and totally in controll. > > Regards, > > Faisal > > > scussion" > Subject: Re: [asterisk-users] Can a duration limit be specified in spool > call file? > > > Bruce, > > All in all, I don't think it's that hostile, it just goes through > cycles...maybe a good number of us may indeed have estrogen issues and it's > the moon, who knows ;-) LOL > > Cheers (and I always mean it, seriously :D ) > > Sherwood McGowan > Yes, THAT Mick > > -- > _ > -- 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] Set variable on Call Answer
the M option in your Dial command will execute a macro upon connection, there's also an option to perform a Gosub... http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial ;-) *keeps his "mailing-list police" badge in it's box in his office* (that wasn't directed at you Dan...there was a little flamewar that I stirred up the other day..that was my troll bit for the day) Check out that link, or run core show application dial from the Asterisk console..look at the options list and find the Macro reference and the Gosub reference...they should light a candle for ya :D On Tue, Feb 8, 2011 at 8:36 AM, Dan Dan wrote: > Hi All, > > First post here. I am dialing out via call file to remote number, when call > is connected a local number is dialed. And on success both calls get bridged > and works fine. > > This is a parallel auto dialout application. I want to set a variable as > soon as the local number answers the call, so that system won't try to > dialout that local number again and stops further dialing. What should be > the best way to deal this situation ? > > Any help would be appreciated. > > Thanks > -dani > > > -- > _ > -- 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
[asterisk-users] forward calls by the ports
hi i searched a lot but i couldn't find the answer i have two openvox(fxo/fxs) card so I have 24 ports! 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? i should make an AGI? or can i make it with extentions.conf? how can i get the caller's port number? thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set variable on Call Answer
Hi All, First post here. I am dialing out via call file to remote number, when call is connected a local number is dialed. And on success both calls get bridged and works fine. This is a parallel auto dialout application. I want to set a variable as soon as the local number answers the call, so that system won't try to dialout that local number again and stops further dialing. What should be the best way to deal this situation ? Any help would be appreciated. Thanks -dani -- _ -- 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] Call files error
Just verified I faced the same issue once and got it reolved by adding /n like
Channel: [mailto:Local/0036701234567@CustomCallOut-1/n]
Local/0036701234567@CustomCallOut-1/n in you case.
-Original Message-
From: "Tamás Dajka"
Sent: Tuesday, February 8, 2011 8:49am
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] Call files error
How can I do that, and do it with LCR?
2011/2/8 <[mailto:fai...@vopium.com] fai...@vopium.com>
Why don't you use single callfile and set CLI and other perameters in dial-plan
as unique as you need?
-Original Message-
From: "Tamás Dajka" <[mailto:tda...@gmail.com] tda...@gmail.com>
Sent: Tuesday, February 8, 2011 7:45am
To: [mailto:asterisk-users@lists.digium.com] asterisk-users@lists.digium.com
Subject: [asterisk-users] Call files error
Hi All,
I'm having some troubles with using call files.
I'm trying to establish the following:
- want to use call files to connect two (outside) extensions
- want to use the outbound routes set in FreePBX
- want to set the outgoing callerid for both calls
- want to set a custom CDR field in MySQL ( field name 'azonosito' )
Asterisk is version 1.8.2.3 with freepbx 2.8.1.
What I've tried is to create two custom context and place the call through
them.
The call file:
; First CID
SetVar: callid1=0036
SetVar: azon1=elso hivas azonosito { Viperke }
; Frist phone num
Channel: Local/0036701234567@CustomCallOut-1
WaitTime: 45
MaxRetries: 0
RetryTime: 0
; 2nd CID
SetVar: callid2=0036204313763
SetVar: azon2=masodik hivas azonosito { V1pr: ehehhe }
Context: CustomCallOut-2
; 2nd phone num
Extension: 003617654321
The contexts:
[CustomCallOut-1]
; set custom CDR
exten => _0X.,1,Set(CDR(azonosito)=${azon1})
exten => _0X.,n,Set(CALLERPRES()=allowed)
exten => _0X.,n,Set(CALLERID(number)=<${callid1}>)
exten => _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include => from-internal
[CustomCallOut-2]
exten => _0X.,1,Wait(1)
; set custom CDR
exten => _0X.,2,Set(CDR(azonosito)=${azon2})
exten => _0X.,3,Playtones(ring)
exten => _0X.,n,Set(CALLERPRES()=allowed)
exten => _0X.,n,Set(CALLERID(number)=<${callid2}>)
exten => _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include => from-internal
However the two calls are placed, the CDRs and the callerids are set correctly,
we can't hear each other. As I saw in the logs, the problem is that the calls
are placed in the same context, and not being connected ( like one call, but
with the variable EXTEN changed ).
I'm really confused about doing this, so can you please advise?
Thanks,
Tamas
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Re: [asterisk-users] Call files error
How can I do that, and do it with LCR?
2011/2/8
> Why don't you use single callfile and set CLI and other perameters in
> dial-plan as unique as you need?
>
>
>
>
> -Original Message-
> From: "Tamás Dajka"
> Sent: Tuesday, February 8, 2011 7:45am
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Call files error
>
> Hi All,
>
> I'm having some troubles with using call files.
>
> I'm trying to establish the following:
> - want to use call files to connect two (outside) extensions
> - want to use the outbound routes set in FreePBX
> - want to set the outgoing callerid for both calls
> - want to set a custom CDR field in MySQL ( field name 'azonosito' )
>
> Asterisk is version 1.8.2.3 with freepbx 2.8.1.
>
> What I've tried is to create two custom context and place the call through
> them.
>
> The call file:
>
> ; First CID
> SetVar: callid1=0036
> SetVar: azon1=elso hivas azonosito { Viperke }
> ; Frist phone num
> Channel: Local/0036701234567@CustomCallOut-1
>
> WaitTime: 45
> MaxRetries: 0
> RetryTime: 0
> ; 2nd CID
> SetVar: callid2=0036204313763
> SetVar: azon2=masodik hivas azonosito { V1pr: ehehhe }
> Context: CustomCallOut-2
> ; 2nd phone num
> Extension: 003617654321
>
>
>
> The contexts:
>
> [CustomCallOut-1]
> ; set custom CDR
> exten => _0X.,1,Set(CDR(azonosito)=${azon1})
> exten => _0X.,n,Set(CALLERPRES()=allowed)
> exten => _0X.,n,Set(CALLERID(number)=<${callid1}>)
>
> exten => _0X.,n,Set(KEEPCID=TRUE)
> ; pass the call to internal routing
> include => from-internal
>
> [CustomCallOut-2]
> exten => _0X.,1,Wait(1)
> ; set custom CDR
> exten => _0X.,2,Set(CDR(azonosito)=${azon2})
>
> exten => _0X.,3,Playtones(ring)
> exten => _0X.,n,Set(CALLERPRES()=allowed)
> exten => _0X.,n,Set(CALLERID(number)=<${callid2}>)
> exten => _0X.,n,Set(KEEPCID=TRUE)
> ; pass the call to internal routing
>
> include => from-internal
>
>
>
> However the two calls are placed, the CDRs and the callerids are set
> correctly, we can't hear each other. As I saw in the logs, the problem is
> that the calls are placed in the same context, and not being connected (
> like one call, but with the variable EXTEN changed ).
>
> I'm really confused about doing this, so can you please advise?
>
> Thanks,
>
> Tamas
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Call files error
This is obvious for the first Channel ( Channel: Local/0036701234567@CustomCallOut-1/n ), but how to set on the other party? I tried with Context: CustomCallOut-2/n but didn't worked. 2011/2/8 Sherwood McGowan > > > > >> However the two calls are placed, the CDRs and the callerids are set >> correctly, we can't hear each other. As I saw in the logs, the problem is >> that the calls are placed in the same context, and not being connected ( >> like one call, but with the variable EXTEN changed ). >> >> I'm really confused about doing this, so can you please advise? >> >> Thanks, >> >> Tamas >> > > > Tamas, > Try appending /n to both of your Local channel definitions... literally a > forward slash and a lowercase n...not newline :D > > > > -- > _ > -- 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] Call files error
> However the two calls are placed, the CDRs and the callerids are set > correctly, we can't hear each other. As I saw in the logs, the problem is > that the calls are placed in the same context, and not being connected ( > like one call, but with the variable EXTEN changed ). > > I'm really confused about doing this, so can you please advise? > > Thanks, > > Tamas > Tamas, Try appending /n to both of your Local channel definitions... literally a forward slash and a lowercase n...not newline :D -- _ -- 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
Steve Howes
SMTP to Google proxy Inc
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Re: [asterisk-users] ${HANGUPCAUSE} in CDR
${HANGUPCAUSE} value is available on h extension.
-Original Message-
From: "Shariq Khan"
Sent: Tuesday, February 8, 2011 8:30am
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: [asterisk-users] ${HANGUPCAUSE} in CDR
Hello Gurus,
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.
Regards,
Shariq Khan
0333-3501125
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Re: [asterisk-users] Call Recording audio file quality query
yep..that would be what i said, using the nifty slang my "peeps" use in the datacenters I just wanted to be "cool" like them...*hangs head*... great...now I gotta transfer to another school... LOL, have a good one mate! On Tue, Feb 8, 2011 at 7:23 AM, wrote: > Yes. The technology need to be used on LAN switches is "port mirroring" or > "line tapping" > > > > > -Original Message- > From: "Sherwood McGowan" > Sent: Tuesday, February 8, 2011 7:34am > To: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users@lists.digium.com> > Subject: Re: [asterisk-users] Call Recording audio file quality query > > On Tue, Feb 8, 2011 at 6:01 AM, wrote: > >> But if you are getting calls all the way on VoIP then you can have calls >> in HD audio using HD audio codec on all locations (Server and Client). In >> that case you either need use some available 3rd party solution which uses >> packet capturing to trace the calls and record call using packet capture and >> assembling regardless of server as asterisk still will not be able to record >> call in HD but some other switches like FreeSWITCH can do it or you need to >> write your own app like it. >> >> > > It's not difficult at all to perform what you're referring to..If you have > the hardware... > > A simple way is to have a port on your main network switch/router that will > "firehose" the traffic the device interacts with In case someone reading > this doesn't know, I'm talking about having a port that just makes a copy of > EVERY PACKET that the device "sees" and sends those copies out over the port > that you've set up for the purpose..It just GUSHES data over that > port...like a firehose just gushes out all the water it possibly can... LOL > > Anyway, once your data is being mirrored over that firehose, send it to a > dedicated "recording" server...all it has to do is find the signaling > packets for each call and then just dump the "payload" from the RTP. It'll > come out exactly as it was transported within RTP...in the codec the call > set up > > I may be wrong, but I'm fairly sure that Asterisk can write a filetype for > almost any of it's codecs...I know it can READ audio files that are encoded > in GSM, uLaw (ul), aLaw (al), G726 and G729 formats (.g729, g.726)...etc... > > If the "DECoding" portion is there, there's almost GOT to be the "enCOding" > functionality... > > > > -- > _ > -- 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
[asterisk-users] ${HANGUPCAUSE} in CDR
Hello Gurus,
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.
Regards,
Shariq Khan
0333-3501125
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Re: [asterisk-users] Callback through extensions.conf?
On Tue, 08 Feb 2011 14:23:12 +0100, Gilles wrote: >However, by chance, I happened on a pattern: The callfile is handled >only if I... >1. Stop Asterisk through its init.d script >2. Mv the callfile >3. Start Asterisk through its init.d script It also works if I launch Asterisk manually with eg. "asterisk -ddc". -- _ -- 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] Call files error
Why don't you use single callfile and set CLI and other perameters in dial-plan
as unique as you need?
-Original Message-
From: "Tamás Dajka"
Sent: Tuesday, February 8, 2011 7:45am
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call files error
Hi All,
I'm having some troubles with using call files.
I'm trying to establish the following:
- want to use call files to connect two (outside) extensions
- want to use the outbound routes set in FreePBX
- want to set the outgoing callerid for both calls
- want to set a custom CDR field in MySQL ( field name 'azonosito' )
Asterisk is version 1.8.2.3 with freepbx 2.8.1.
What I've tried is to create two custom context and place the call through
them.
The call file:
; First CID
SetVar: callid1=0036
SetVar: azon1=elso hivas azonosito { Viperke }
; Frist phone num
Channel: Local/0036701234567@CustomCallOut-1
WaitTime: 45
MaxRetries: 0
RetryTime: 0
; 2nd CID
SetVar: callid2=0036204313763
SetVar: azon2=masodik hivas azonosito { V1pr: ehehhe }
Context: CustomCallOut-2
; 2nd phone num
Extension: 003617654321
The contexts:
[CustomCallOut-1]
; set custom CDR
exten => _0X.,1,Set(CDR(azonosito)=${azon1})
exten => _0X.,n,Set(CALLERPRES()=allowed)
exten => _0X.,n,Set(CALLERID(number)=<${callid1}>)
exten => _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include => from-internal
[CustomCallOut-2]
exten => _0X.,1,Wait(1)
; set custom CDR
exten => _0X.,2,Set(CDR(azonosito)=${azon2})
exten => _0X.,3,Playtones(ring)
exten => _0X.,n,Set(CALLERPRES()=allowed)
exten => _0X.,n,Set(CALLERID(number)=<${callid2}>)
exten => _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include => from-internal
However the two calls are placed, the CDRs and the callerids are set correctly,
we can't hear each other. As I saw in the logs, the problem is that the calls
are placed in the same context, and not being connected ( like one call, but
with the variable EXTEN changed ).
I'm really confused about doing this, so can you please advise?
Thanks,
Tamas
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Re: [asterisk-users] Call Recording audio file quality query
Yes. The technology need to be used on LAN switches is "port mirroring" or "line tapping" -Original Message- From: "Sherwood McGowan" Sent: Tuesday, February 8, 2011 7:34am To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Call Recording audio file quality query On Tue, Feb 8, 2011 at 6:01 AM, <[mailto:fai...@vopium.com] fai...@vopium.com> wrote: But if you are getting calls all the way on VoIP then you can have calls in HD audio using HD audio codec on all locations (Server and Client). In that case you either need use some available 3rd party solution which uses packet capturing to trace the calls and record call using packet capture and assembling regardless of server as asterisk still will not be able to record call in HD but some other switches like FreeSWITCH can do it or you need to write your own app like it. It's not difficult at all to perform what you're referring to..If you have the hardware... A simple way is to have a port on your main network switch/router that will "firehose" the traffic the device interacts with In case someone reading this doesn't know, I'm talking about having a port that just makes a copy of EVERY PACKET that the device "sees" and sends those copies out over the port that you've set up for the purpose..It just GUSHES data over that port...like a firehose just gushes out all the water it possibly can... LOL Anyway, once your data is being mirrored over that firehose, send it to a dedicated "recording" server...all it has to do is find the signaling packets for each call and then just dump the "payload" from the RTP. It'll come out exactly as it was transported within RTP...in the codec the call set up I may be wrong, but I'm fairly sure that Asterisk can write a filetype for almost any of it's codecs...I know it can READ audio files that are encoded in GSM, uLaw (ul), aLaw (al), G726 and G729 formats (.g729, g.726)...etc... If the "DECoding" portion is there, there's almost GOT to be the "enCOding" functionality... -- _ -- 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?
Thanks much everyone for the great help. I did go through the last suggestions about the callfile (no CRLF issue, permissions are 644 and file owned by root, starting asterisk through strace, etc.), but none helped. However, by chance, I happened on a pattern: The callfile is handled only if I... 1. Stop Asterisk through its init.d script 2. Mv the callfile 3. Start Asterisk through its init.d script Here are the commands I run, the little script I use to move the callfile, and what it contains: === /var/tmp> /etc/init.d/asterisk start /var/tmp> ./mvSIP.bash /var/tmp> /etc/init.d/asterisk stop /var/tmp> /etc/init.d/asterisk start === /var/tmp> cat mvSIP.bash #!/bin/sh cp callfileSIP.call.backup callfileSIP.call mv callfileSIP.call /var/spool/asterisk/outgoing === Channel: SIP/xlite Context: callback-dialtone-auth Extension: s Priority: 1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Archive: yes === Once the callfile has been handled, it is moved from /var/spool/asterisk/outgoing to ./outgoing_done and has a couple of lines appended: === ... StartRetry: 2306 1 (1297171283) Status: Completed === I don't know if it means anything, but here's the output of "mount" on this appliance (the root filesystem uses yaffs for persistence): === /var/tmp> mount rootfs on / type rootfs (rw) /dev/root on / type yaffs (rw) proc on /proc type proc (rw) ramfs on /var/tmp type ramfs (rw) sysfs on /sys type sysfs (rw) devpts on /dev/pts type devpts (rw) usbfs on /proc/bus/usb type usbfs (rw) securityfs on /sys/kernel/security type securityfs (rw) === Thank you. -- _ -- 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] Call files error
Hi All,
I'm having some troubles with using call files.
I'm trying to establish the following:
- want to use call files to connect two (outside) extensions
- want to use the outbound routes set in FreePBX
- want to set the outgoing callerid for both calls
- want to set a custom CDR field in MySQL ( field name 'azonosito' )
Asterisk is version 1.8.2.3 with freepbx 2.8.1.
What I've tried is to create two custom context and place the call through
them.
The call file:
; First CID
SetVar: callid1=0036
SetVar: azon1=elso hivas azonosito { Viperke }
; Frist phone num
Channel: Local/0036701234567@CustomCallOut-1
WaitTime: 45
MaxRetries: 0
RetryTime: 0
; 2nd CID
SetVar: callid2=0036204313763
SetVar: azon2=masodik hivas azonosito { V1pr: ehehhe }
Context: CustomCallOut-2
; 2nd phone num
Extension: 003617654321
The contexts:
[CustomCallOut-1]
; set custom CDR
exten => _0X.,1,Set(CDR(azonosito)=${azon1})
exten => _0X.,n,Set(CALLERPRES()=allowed)
exten => _0X.,n,Set(CALLERID(number)=<${callid1}>)
exten => _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include => from-internal
[CustomCallOut-2]
exten => _0X.,1,Wait(1)
; set custom CDR
exten => _0X.,2,Set(CDR(azonosito)=${azon2})
exten => _0X.,3,Playtones(ring)
exten => _0X.,n,Set(CALLERPRES()=allowed)
exten => _0X.,n,Set(CALLERID(number)=<${callid2}>)
exten => _0X.,n,Set(KEEPCID=TRUE)
; pass the call to internal routing
include => from-internal
However the two calls are placed, the CDRs and the callerids are set
correctly, we can't hear each other. As I saw in the logs, the problem is
that the calls are placed in the same context, and not being connected (
like one call, but with the variable EXTEN changed ).
I'm really confused about doing this, so can you please advise?
Thanks,
Tamas
--
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Error loading module 'cdr_radius.so'
Hello, you have to install radiusclient-ng http://developer.berlios.de/projects/radiusclient-ng/ Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording audio file quality query
On Tue, Feb 8, 2011 at 6:01 AM, wrote: > But if you are getting calls all the way on VoIP then you can have calls in > HD audio using HD audio codec on all locations (Server and Client). In that > case you either need use some available 3rd party solution which uses packet > capturing to trace the calls and record call using packet capture and > assembling regardless of server as asterisk still will not be able to record > call in HD but some other switches like FreeSWITCH can do it or you need to > write your own app like it. > > It's not difficult at all to perform what you're referring to..If you have the hardware... A simple way is to have a port on your main network switch/router that will "firehose" the traffic the device interacts with In case someone reading this doesn't know, I'm talking about having a port that just makes a copy of EVERY PACKET that the device "sees" and sends those copies out over the port that you've set up for the purpose..It just GUSHES data over that port...like a firehose just gushes out all the water it possibly can... LOL Anyway, once your data is being mirrored over that firehose, send it to a dedicated "recording" server...all it has to do is find the signaling packets for each call and then just dump the "payload" from the RTP. It'll come out exactly as it was transported within RTP...in the codec the call set up I may be wrong, but I'm fairly sure that Asterisk can write a filetype for almost any of it's codecs...I know it can READ audio files that are encoded in GSM, uLaw (ul), aLaw (al), G726 and G729 formats (.g729, g.726)...etc... If the "DECoding" portion is there, there's almost GOT to be the "enCOding" functionality... -- _ -- 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] Call Recording audio file quality query
> > > That answer was pretty much what I was expecting. Just wanted to make > sure. > Glad to be of service :D -- _ -- 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] Call Recording audio file quality query
But if you are getting calls all the way on VoIP then you can have calls in HD audio using HD audio codec on all locations (Server and Client). In that case you either need use some available 3rd party solution which uses packet capturing to trace the calls and record call using packet capture and assembling regardless of server as asterisk still will not be able to record call in HD but some other switches like FreeSWITCH can do it or you need to write your own app like it. -Original Message- From: "Ishfaq Malik" Sent: Tuesday, February 8, 2011 6:47am To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Call Recording audio file quality query On Tue, 2011-02-08 at 05:40 -0600, Sherwood McGowan wrote: > On Tue, Feb 8, 2011 at 5:09 AM, Ishfaq Malik > wrote: > Hi > > We're getting requests coming in for higher quality audio in > our call > recordings. We currently use MixMonitor and everything is > being saved in > it's native 8000Hz, 16 bit wav format. > > I have seen information on using Monitor and specifying a > conversion to > mp3 when the call ends and the 2 channels get mixed but surely > the 2 > channels are already saved as 16bit 8000Hz wav files so the > quality is > lost already? > > Is there any way of making high quality recordings of call > content? > > > Have you ever heard of the saying "You can't polish a turd" ? > > It doesn't matter if you have an app capable of recording 196Khz 24bit > recordings (or capable of upsampling to that sample rate)...if the > call itself is native at 8Khz 16bit, you'd just be making a bigger > recording file with no literal improvement in quality. > > You can't create more samples of audio from nothing. it's like taking > a new box of, say, 50 paperclips... Now, go get an empty box that says > it contained 250 paperclips when it was purchased... Now, throw all 50 > paperclips from the little box into the big box marked 250..now, > imagine REALLY REALLY hard that you think you can perceive about 5 > more paperclips somewhere all mixed up in the > jumble...(Extrapolation) > > that, my friend, is an over simplified metaphor, but in essence it's > close enough to get the point across.. > > Sorry bud :( If you don't believe me, I can refer you to my old audio > production school ;-D ) > > Slainte! > the Mick > That answer was pretty much what I was expecting. Just wanted to make sure. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Call Recording audio file quality query
> -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Ishfaq Malik > Sent: Tuesday, February 08, 2011 6:10 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Call Recording audio file quality query > > Hi > > We're getting requests coming in for higher quality audio in our call > recordings. We currently use MixMonitor and everything is being saved > in > it's native 8000Hz, 16 bit wav format. > > I have seen information on using Monitor and specifying a conversion to > mp3 when the call ends and the 2 channels get mixed but surely the 2 > channels are already saved as 16bit 8000Hz wav files so the quality is > lost already? > > Is there any way of making high quality recordings of call content? > > We're currently using asterisk 1.4 and soon upgrading to 1.8 > > Thanks in Advance > > Switch everything to ulaw/alaw codecs, and stop using highly compressed codecs As for 16bit, 8khz, that is as high as your going to get in the telephone world. -- _ -- 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] Call Recording audio file quality query
On Tue, 2011-02-08 at 05:40 -0600, Sherwood McGowan wrote: > On Tue, Feb 8, 2011 at 5:09 AM, Ishfaq Malik > wrote: > Hi > > We're getting requests coming in for higher quality audio in > our call > recordings. We currently use MixMonitor and everything is > being saved in > it's native 8000Hz, 16 bit wav format. > > I have seen information on using Monitor and specifying a > conversion to > mp3 when the call ends and the 2 channels get mixed but surely > the 2 > channels are already saved as 16bit 8000Hz wav files so the > quality is > lost already? > > Is there any way of making high quality recordings of call > content? > > > Have you ever heard of the saying "You can't polish a turd" ? > > It doesn't matter if you have an app capable of recording 196Khz 24bit > recordings (or capable of upsampling to that sample rate)...if the > call itself is native at 8Khz 16bit, you'd just be making a bigger > recording file with no literal improvement in quality. > > You can't create more samples of audio from nothing. it's like taking > a new box of, say, 50 paperclips... Now, go get an empty box that says > it contained 250 paperclips when it was purchased... Now, throw all 50 > paperclips from the little box into the big box marked 250..now, > imagine REALLY REALLY hard that you think you can perceive about 5 > more paperclips somewhere all mixed up in the > jumble...(Extrapolation) > > that, my friend, is an over simplified metaphor, but in essence it's > close enough to get the point across.. > > Sorry bud :( If you don't believe me, I can refer you to my old audio > production school ;-D ) > > Slainte! > the Mick > That answer was pretty much what I was expecting. Just wanted to make sure. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Call Recording audio file quality query
On Tue, Feb 8, 2011 at 5:09 AM, Ishfaq Malik wrote: > Hi > > We're getting requests coming in for higher quality audio in our call > recordings. We currently use MixMonitor and everything is being saved in > it's native 8000Hz, 16 bit wav format. > > I have seen information on using Monitor and specifying a conversion to > mp3 when the call ends and the 2 channels get mixed but surely the 2 > channels are already saved as 16bit 8000Hz wav files so the quality is > lost already? > > Is there any way of making high quality recordings of call content? > > Have you ever heard of the saying "You can't polish a turd" ? It doesn't matter if you have an app capable of recording 196Khz 24bit recordings (or capable of upsampling to that sample rate)...if the call itself is native at 8Khz 16bit, you'd just be making a bigger recording file with no literal improvement in quality. You can't create more samples of audio from nothing. it's like taking a new box of, say, 50 paperclips... Now, go get an empty box that says it contained 250 paperclips when it was purchased... Now, throw all 50 paperclips from the little box into the big box marked 250..now, imagine REALLY REALLY hard that you think you can perceive about 5 more paperclips somewhere all mixed up in the jumble...(Extrapolation) that, my friend, is an over simplified metaphor, but in essence it's close enough to get the point across.. Sorry bud :( If you don't believe me, I can refer you to my old audio production school ;-D ) Slainte! the Mick -- _ -- 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] Call Recording audio file quality query
Hi We're getting requests coming in for higher quality audio in our call recordings. We currently use MixMonitor and everything is being saved in it's native 8000Hz, 16 bit wav format. I have seen information on using Monitor and specifying a conversion to mp3 when the call ends and the 2 channels get mixed but surely the 2 channels are already saved as 16bit 8000Hz wav files so the quality is lost already? Is there any way of making high quality recordings of call content? We're currently using asterisk 1.4 and soon upgrading to 1.8 Thanks in Advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Error loading module 'cdr_radius.so'
I have this Error Please Help me loader.c: Error loading module 'cdr_radius.so': libradiusclient-ng.so.2: cannot open shared object file: No such file or directory -- _ -- 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
