[asterisk-users] A quick question in terms of DAHDI channel
Hello, I have an Asterisk 1.8.11 installation. When I built up this Asterisk, I didn't install DAHDI channel, if I issue command connect*CLI core show channeltypes I would have response like: connect*CLI core show channeltypes TypeDescription Devicestate Indications Transfer -- --- --- --- USTMUNISTIM Channel Driver no yes no Phone Standard Linux Telephony API Driver no yes no Console OSS Console Channel Driver no yes no Skinny Skinny Client Control Protocol (Skinny) yes yes no Local Local Proxy Channel Driver yes yes no SIP Session Initiation Protocol (SIP)yes yes yes Agent Call Agent Proxy Channel yes yes no MGCPMedia Gateway Control Protocol (MGCP)yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes MulticastR Multicast RTP Paging Channel Driver no no no Bridge Bridge Interaction Channel no no no -- 11 channel drivers registered. But right now, I am planing to connect a PRI trunk to this Asterisk. so I put in a PRI card install dahdi-linux-complete-2.6.2 and libpri-1.4.14. Afterward, dahdi_tool is able to find PRI board, and all channels. But my question is when I try to send call to DAHDI channel in the dial plan, CLI print out a warning saying [Jun 13 07:48:21] WARNING[2393]: channel.c:5606 ast_request: No channel type registered for 'DAHDI' According to my description above, it make sense, since my Asterisk does not install DAHDI channel before. Therefore my question is in my case, it is required to re-intall whole Asterisk, or there is some other way that I just could only install DAHDI channel. I did some google search. but I didn't find a proper answer. Thanks for your help. longst -- _ -- 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] VOIP PRI Gateways
If I understand you correctly, you test a service which converter SIP to ISDN PRI On Mar 8, 2013, at 2:16 AM, Daniel Harper dan...@harper.net.nz wrote: I was hoping someone might have some knowledge to impart regarding VOIP PRI Gateways or the psudo ISDN services being offered these days. The official line in Australia is that true ISDN services are on their way out. I am testing a service provided by one of the telcos I am told that it cannot provide ISDN cause codes for disconnected/invalid numbers and all we get is the audio. Also the service no longer sends the progress event RINGING to indicate the line is actual ringing (as apposed to the audio) Has anyone else ran into this problem and does anyone have any ideas how to address it? Can DAHDI detect ringing tones on PRI lines? -- Cheers, Daniel -- _ -- 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] ERROR: chan_dahdi.c: PRI Span: 3 PROBLEM: General: Badly Structured Component
Did you get it to work may I ask ? On Feb 20, 2013, at 3:49 PM, gincantalupo gincantal...@fgasoftware.com wrote: Hi all, has anybody ever encountered this ERROR before? It happens frequently on my debian6-based pbx. I'm using Asterisk 1.8.11 with dahdi-linux-2.4.1 and a quadBRI card. ERROR: chan_dahdi.c: PRI Span: 3PROBLEM: General: Badly Structured Component I tried to google but without success. Do you know what it means? Should I worry? Thank You Giorgio -- _ -- 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] About Asterisk with Digium TE405P PRI ISDN cad
Hello, I am trying to connect two asterisks with PRI connection. One asterisk has TE405P Quad PRI ports card, anther asterisk has TE110P 1 PRI port card. I am wondering if there would be some step by step guide that I could follow to to this kind of connection? Thanks -- from longst -- _ -- 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] Quick start configuration sample for chan_dahdi.conf
I am really a beginner of PRI ISDN board, I am wondering if there is a quick start chan_dahdi.conf configuration I could use. I tried to install two FreePBX boxes follow the instructions from http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html; connected them between PRIs, It worked. And now if I refer the FreePBX chan_dahdi.conf it looks like http://pastebin.com/kfWWL6dm; and it seems there is no specific configuration in FreePBX chan_dahdi.conf. And now I tried to add [global] [3:33pm] #include dahdi-channels.conf into chan_dahdi.conf. and do a static-host*CLI dahdi restart still seems no progress… longst -- _ -- 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] About Asterisk with Digium TE405P PRI ISDN cad
Thanks for your message. On Feb 11, 2013, at 2:34 PM, Yves A. yves...@gmx.de wrote: Hi, I think, you mean connecting the two boxes directly with a cable... not via PSTN, right? 1.) You need a special cross-over cable to connect one Port directly to another Port... (if you want to crimp it yourself, you can find the associated Pins via Google... ethernet crossover cables do not work as they have different links) Yes I made a special cross-over cable according to PRI cable pattern. 2.) configure one end as master (CPN) and the other asterisk as Network (CPN), otherwise you´ll get timing issues... I think I lack of some ISDN basic knowledge I am trying to follow the article from http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html to do this task. And I face some general question, for example : According to http://www.facebook.com/photo.php?fbid=10151387245397906set=a.10151387218792906.496748.736667905type=3theater It is a screen shot of one Span of a TE405P card from DAHDI tools. I am wondering if there are some document explain what different configurations means, for example, Current Alarms, Sync Source IRQ Misses, etc…. Thanks f thats all... regards, yves Am 11.02.2013 14:00, schrieb Shitian Long: Hello, I am trying to connect two asterisks with PRI connection. One asterisk has TE405P Quad PRI ports card, anther asterisk has TE110P 1 PRI port card. I am wondering if there would be some step by step guide that I could follow to to this kind of connection? Thanks -- from longst -- _ -- 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] About Asterisk with Digium TE405P PRI ISDN cad
On Feb 11, 2013, at 4:31 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Monday 11 February 2013, Shitian Long wrote: Hello, I am trying to connect two asterisks with PRI connection. One asterisk has TE405P Quad PRI ports card, anther asterisk has TE110P 1 PRI port card. I am wondering if there would be some step by step guide that I could follow to to this kind of connection? Thanks If you want to connect the two boxes together via the telephone network, then you will need appropriate NTEs (Network termination Equipment -- the boundary between where the telco's responsibility ends and yours begins) installed, and the telco should give you cables -- or at least advise on wiring. Connecting an Asterisk card to an NTE requires a straight-through cable. If you just want to connect the boxes directly (aot via the telephone network) then you will need to make up a special cable. Get CAT5 cable, plugs and crimping tool. (If you are especially lazy, you can even just cut the plug off one end of a pre-wired CAT5 cable, and crimp your own in place of where it used to be.) Now you need to swap over pin 1 (WHITE/orange) with pin 5 (WHITE/blue) and pin 2 (ORANGE/white) with pin 4 (BLUE/white). It won't do any harm leaving pins 3, 6, 7 and 8 connected, and it will make crimping up the plugs easier. One end: Standard wiring. 1: WHITE/orange 2: ORANGE/white 3: WHITE/green 4: BLUE/white 5: WHITE/blue 6: GREEN/white 7: WHITE/brown 8: BROWN/white Other end: Special wiring for ISDN crossover. 1: WHITE/blue 2: BLUE/white 3: WHITE/green 4: ORANGE/white 5: WHITE/orange 6: GREEN/white 7: WHITE/brown 8: BROWN/white Thanks for your message, at moment, I have an Asterisk with a TE405P Quad ports PRI ISDN card, Span 1 connect to a NT(network terminal) equipment, which is a GSM gateway with straight cable Span 2 connect to a TE(telecom equipment), which is an another asterisk installation with TE110P, with cross PRI rewiring cable. At moment, I think the cable are properly connected, since I check out TE405P card, actually two ports indicate green. TE110P indicate Green. And GSM Gateway LAY1 indicate Green, but LAY2 indicate blinking green. I am tying to process the following work according to http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html And may I have some general ISDN questions: according to dahdi_tools, I would be able to check configuration from TE405P card. It has following configurations : Current Alarms: No alarms. Sync Source:T4XXP (PCI) Card 0 Span 1 IRQ Misses: 0 Bipolar Viol: 0 Tx/Rx Levels: 0/ 0 Total/Conf/Act: 31/ 31/ 0 112233 1234567890123456789012345678901 my question is what is meaning of each of configuration mentioned above? Thanks Don't forget, one of the machines has to be told (in chan-dahdi.conf) to pretend it is an NTE rather than subscriber's equipment! -- AJS Answers come *after* questions. -- _ -- 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] Question about directmedia or canreinvite in sip.conf
Hello, I have a question about directmedia or canreinvite, I have experience that whatever I set directmedia=yes or no. After I run sip show settings. all settings looks the same. My question is how I could make sure from sip show settings that my directmedia configuration is applied. 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] Questions about extension.conf
Hello
I have been reading the sample extension.conf
;###
[outbound-freenum2]
; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
;
exten = _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same = n,Set(SUFFIX=${CUT(EXTEN,*,2-)}); make
sure the suffix is all digits as well
same = n,GotoIf($[${FILTER(0-9,${SUFFIX})} != ${SUFFIX}]?fn-CONGESTION,1)
;
filter out bad characters per the README-SERIOUSLY.best-practices.txt document
same = n,Set(TIMEOUT(absolute)=10800)
same = n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ;
perform our lookup with freenum.org
same = n,GotoIf($[${isnresult} != ]?from)
same = n,Set(DIALSTATUS=CONGESTION)
same = n,Goto(fn-CONGESTION,1)
same = n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} = ]?dial) ; check
if we set the FREENUMDOMAIN global variable in [global]
same = n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ;if
we did set it, then we'll use it for our outbound dialing domain
same = n(dial),Dial(SIP/${isnresult},40)
same = n,Goto(fn-${DIALSTATUS},1)
exten = fn-BUSY,1,Busy()
exten = _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same = n,Congestion()
;##
According to
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf;
Syntax for defining a context: keywords exten, include, ignorepat and switch.
same is not mentioned in this wiki.
There is a part of dial plan from sample extension.conf above. My Question is
how same = key word works .
Thanks
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[asterisk-users] If would possible use a custom function in Asterisk Dialplan
Hello,
If would be possible to use a function concept in side of Asterisk DialPlan
For example:
I have following logic in my dial plan remove country code a add an 0 before
the rest of the numbers
exten = _X.,1, NoOp( call ID ${CALLERID(num)} exten: ${EXTEN}))
; remove my country code
exten = _X.,n,
GotoIf($[${CALLERID(num):0:4}=${country-code}]?international-format:national-format)
exten = _X.,n(international-format), Set(CALLERID(num)=0${CALLERID(num):4})
exten = _X.,n(national-format), NoOp(call ID: ${CALLERID(num)} exten:
${EXTEN}))
Do you think if would be possible that I could write a function something like
REMOVEMYCOUNTRYCODE(${NUM}) with a return value of a number with out country
code and with an 0 add in front of the rest of the numbers.
like
exten = _X.,1, NoOp( call ID ${CALLERID(num)} exten: ${EXTEN}))
; remove my county code
exten = _X.n, Set(CALLERID(num=REMOVEMYCOUNTRYCODE(${CALLERID(num)} ));
then I have to define this function in someway ……
I am trying to googling for a while but I did not find any idea to achieve this
task.
I would appreciate if someone have an idea…
Thanks for your time in advance.
longst
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Re: [asterisk-users] Web based Click to Call Application
more specific, Asterisk Manager Interface , originate action will help you to do so. On Nov 9, 2012, at 5:34 PM, Danny Nicholas da...@debsinc.com wrote: If you want to have a good level of control, AMI is the way to go. If you just want simple and quick, .call files is going to do it. I posted some PERL code a few months back that uses the Asterisk::Manager module to make and monitor calls. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of akhilesh chand Sent: Friday, November 09, 2012 4:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Web based Click to Call Application On Fri, Nov 9, 2012 at 4:11 PM, OCEANET - Cédric BASSAGET ced...@oceanet.com wrote: Or use a php socket and the AMI. Cédric Le 09/11/2012 11:39, A J Stiles a écrit : On Friday 09 November 2012, akhilesh chand wrote: Dear All, I want to develop click to call(C2C) web based application.Is there any study material. I will really appreciate your help, thank you. Look into call files. Basically, you inject a file into the folder /var/spool/asterisk/outgoing/ and this sets up a call for you. And search the archives; because I remember posting a simple click-to-call example script on this list, sometime back this Summer just gone. -- OCEANET --- [AGENCE DU MANS] 7, rue des Frênes ZAC de la Pointe 72190 SARGE LES LE MANS [t] +33 (0)2.43.50.26.50 [f] +33 (0)2.43.72.21.14 [AGENCE D'ANGERS] 5, rue Fleming Angers Technopole 49066 ANGERS [t] +33 (0)2.41.19.28.65 [f] +33 (0)2.52.19.22.00 http://www.oceanet.com http://www.oceanet-telecom.com I'm basically use Asterisk::Manager package ,php and perl. Regards Akhilesh -- _ -- 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] Good way to query data from asterisk realtime with Asterisk Manager API
Thanks for everyone's message, Finally, I figure out how to use AMI fetch data from realtime. After log in AMI. First step, a setVar action should be sent (for example, in this case, I am interested in the content of realtime family = test_family, key = mykey,) setVar action should look like Action: Setvar Variable: REALTIME(test_family, key, mykey) After that, I should have a success response like Response: Success Message: Variable Set Afterward. I could use getVar action to fetch the result Action: Getvar Variable: REALTIME(test_family, key, mykey) Finally, query result will be in the response. From this practice, it seems, AMI getVar and setVar action have more function than I through. Thanks for every one's help. On Sep 1, 2012, at 10:14 AM, Olle E. Johansson wrote: 31 aug 2012 kl. 16:58 skrev Shitian Long longst...@gmail.com: Do you think it is a good way to use Manager API command action to implement this feature? No. The command action should be avoided since the output from the CLI commands is not made for parsing by applications and may change too. Sometimes we cut of informaiton to fit into a terminal window. If you use manager actions instead, you will always get the full data in a format you can parse. If you have to use the command action you have found a place where a manager action is missing and we developers would like to know that and fix it :-) For realtime, there's a dialplan function REALTIME() that you can use with the manager actions that change or read channel variables. That's the best way, since we lack manager realtime commands. One reason for not going directly to the database API is that when building 3rd party apps, we don't know what database you are using and can benefit from the ARA interface to databases, exactly like Asterisk. It's not as effecient as going directly when you can, but sometimes you just don't know what's behind ARA and thanks to ARA you don't have to. :-) /Olle * The new Edvina SIP Masterclass - Stockholm, Sweden Oct and Miami, FL, Dec 2012 http://edvina.net/training/new-sip-masterclass/-- _ -- 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] any good way to reload realtime configuration
Hello, properate After I alter configurations of asterisk realtime in extconfig.conf If there is a good way to apply new configuration? At moment, I do a core restart now or core reload. Probably either of them is not an appropriate reload realtime. But in Asterisk CLI realtime, there is not a reload like sip reload or dialplan reload I am wondering if you guys have suggestions for this realtime reload action? 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] Good way to query data from asterisk realtime with Asterisk Manager API
Hello.
I am trying to use Asterisk Manager API query data from realtime. From Asterisk
CLI, we could use
realtime load realtime-family key matched key value
query realtime
it would have response like
Column Name Column Value
id 1
mykey content
myvalue value
I am wondering how I could make this type of query from Manager API.
Thanks for your time in advance.
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Re: [asterisk-users] Question about cli
As far as I know there is no function add SIP extension directly from CLI, and actually, it is not convenience that add an extension from CLI as well. On Aug 31, 2012, at 4:18 PM, Giuseppe Longo giuseppe...@gmail.com wrote: Hello guys, i would like to ask a question about cli. Today, while i was using the cli, i thinked that there could be more features. IMHO, might be interesting, for example, to add a sip extensions from cli, or other similar functions, without having to modify the configuration files. Or not? What do you think? 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 -- _ -- 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] Good way to query data from asterisk realtime with Asterisk Manager API
Do you think it is a good way to use Manager API command action to implement
this feature?
On Aug 31, 2012, at 4:42 PM, Danny Nicholas da...@debsinc.com wrote:
There might be a specific command to do it, but you can do almost any CLI
command using command function.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shitian Long
Sent: Friday, August 31, 2012 9:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Good way to query data from asterisk realtime with
Asterisk Manager API
Hello.
I am trying to use Asterisk Manager API query data from realtime. From
Asterisk CLI, we could use realtime load realtime-family key matched
key value query realtime it would have response like
Column Name Column Value
id 1
mykey content
myvalue value
I am wondering how I could make this type of query from Manager API.
Thanks for your time in advance.
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Re: [asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API
For some reason, we don't want directly access database. more over, it avoid to
develop different database connection code, and leave the database connection
part with Asterisk
On Aug 31, 2012, at 5:00 PM, Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From: Shitian Long [mailto:longst...@gmail.com]
Sent: Friday, August 31, 2012 9:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
da...@debsinc.com
Subject: Re: [asterisk-users] Good way to query data from asterisk realtime
with Asterisk Manager API
Do you think it is a good way to use Manager API command action to
implement this feature?
On Aug 31, 2012, at 4:42 PM, Danny Nicholas da...@debsinc.com wrote:
There might be a specific command to do it, but you can do almost any
CLI command using command function.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shitian
Long
Sent: Friday, August 31, 2012 9:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Good way to query data from asterisk
realtime with Asterisk Manager API
Hello.
I am trying to use Asterisk Manager API query data from realtime. From
Asterisk CLI, we could use realtime load realtime-family key
matched key value query realtime it would have response like
Column Name Column Value
id 1
mykey content
myvalue value
I am wondering how I could make this type of query from Manager API.
Thanks for your time in advance.
I agree with Warren. Why bother having realtime if you're going to add a
layer you don't need? Just query the database.
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Re: [asterisk-users] Install AsteriskNow
Thanks for your message, If I understand correctly. I have to copy the Asterisk installation code to AsteriskNow installation. And setup complier in AsteriskNow and do the make sample? On Aug 30, 2012, at 12:37 AM, Richard Mudgett rmudg...@digium.com wrote: I am trying to install an AsteriskNow. When system boot up, there are two options, To install with Asterisk 1.8 and FreePBX type 1 ENTER To install with Asterisk 1.8 only type 2 ENTER If I want to install Asterisk 1.8 only for example. After asterisk is install, I found the /etc/asterisk/ is empty. I am wondering if any good way that I could have some sample configurations. Run make samples That will copy all of the sample config files from ./configs into /etc/asterisk with appropriate removal of the .sample from the filenames. Richard -- _ -- 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] Install AsteriskNow
Hello, I am trying to install an AsteriskNow. When system boot up, there are two options, To install with Asterisk 1.8 and FreePBX type 1 ENTER To install with Asterisk 1.8 only type 2 ENTER If I want to install Asterisk 1.8 only for example. After asterisk is install, I found the /etc/asterisk/ is empty. I am wondering if any good way that I could have some sample configurations. Thanks for your time Best Regards longst -- _ -- 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] TE110P Wildcard does not work with Ubuntu 12.04 server
I am trying to setup TE110P wildcard on a PBX running ubuntu 12.04 server edition. I followed the procedure from http://docs.digium.com/misc/ADL_quickstart.pdf step by step. During the process of installing dahdi-linux-complete I got following warnings: root@ubuntu:/usr/local/src/dahdi-linux-complete-2.6.1+2.6.1# make perl: warning: Setting locale failed. perl: warning: Please check that your locale settings: LANGUAGE = en_US:en, LC_ALL = (unset), LC_CTYPE = UTF-8, LANG = en_US.UTF-8 are supported and installed on your system. perl: warning: Falling back to the standard locale (C). Frist of, I am wondering if this error matters? Second question, after installation process complete, and reboot the machine I got the following error, when machine boot up: Loading DAHDI hardware modules: wcte11xp: error I think the TE110P card is no properly loaded. I try to confirm my thought by using root@ubuntu:~# dahdi_tool There is no interface listed on the table. I am wondering if anyone got idea about this issue. Thanks. longst -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How I could Insert a record into the table with REALTIME
Hello, I am trying to working with Asterisk RealTime. According to http://www.voip-info.org/wiki/view/Asterisk+func+realtime Example writing to database using realtime Add avp into extconfig.conf avp = pgsql,hostedpbx,thiak_avp and add new extension into extension.conf exten = s,n,Set(REALTIME(avp,attribute,USED_CHANS/BOB,value)=1) This will execute the following SQL: UPDATE avp SET value = '1' WHERE attribute = 'USED_CHANS/BOB' My Question is if I would like to execute INSERT SQL statement like INSERT INTO avp (value, attribute) VALUES(value1, value2) How I am able to make it work with REALTIME. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Integration with building intercom systems
I think you are supposed to have a IP based terminal in order to carry both video and audio. If budget is accept, it is possible to setup several iOS devices as dedicated SIP terminals On Jul 6, 2012, at 8:49 AM, Olivier wrote: 2012/7/5, giovanni.v i...@keybits.org: The matter becomes more difficult approaching a building install as there are no devices to handle properly that. I think the snom PA-1 may be a good candidate to play with because of a versatile I/O that could be interfaced to a custom door-phone bridge to IP. Handling both audio and video seems difficult. -- _ -- 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] sip and extensions
Hello, If you would like to make out bound call (from Asterisk to SIP provider), it is fine. But if you want have inbound call (from SIP provider to Asterisk). I think you are supposed to have something like this sip.conf register = 5552530146:your_password@sip3.voipvoip.com/5552530146 [5552530146] ... context=incoming extensions.conf [incoming] ;first creating extensions for your local users exten = 5552530146,1,Goto(5552530146_incomming,s,1) [5552530146_incomming] ;more logic wish it would help. On Jul 5, 2012, at 11:44 PM, Thomas Perron wrote: I am new. Here is the code that I am playing with on CentOS 6.x When I dial the number that corresponds w/ my SIP account I get a recording: reached a non-working number I built Asterisk a few times last year and am now back working on a similar project. In my view, there is something wrong in sip.conf I don't remember using a file that long to get a basic call set up. The format was provided to me by voipvoip.com (the SIP provider). Does anyone have any comments please? I just want a very simple config to get my machine to recognize a call to the SIP provider. Here is results of sip show registry: Hostdnsmgr Username Refresh State Reg.Time sip3.voipvoip.com:5060 N 5552530146 285 Registered Thu, 05 Jul 2012 21:39:56 1 SIP registrations. Here is sip and extensions.conf sip.conf [general] register = 5552530146:funnytiger...@sip3.voipvoip.com ; [sip3.voipvoip.com] [outgoing] username=5552530146 type=peer qualify=yes secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromuser=5552530146 fromdomain=69.90.209.57 dtmfmode=rfc2833 allow=g729 allow=ilbc allow=ulaw allow=alaw disallow=all srvlookup=no [incoming] username=5552530146 type=user secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromdomain=69.90.209.57 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc disallow=all srvlookup=no extensions.conf [general] ; ; [incoming] ;first creating extensions for your local users exten= s,1,Dial(SIP/1703717) exten= s,2,Hangup() -- _ -- 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] after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port
if you check out your sip.conf. On Jun 29, 2012, at 5:54 PM, gincantalupo wrote: Hi all, after upgrading my Asterisk 1.4.26.2 to 1.8.11.0 I cannot register to my VoIP provider because it says I'm trying to connect to port 55150 (that's what the call center guy told me)...but I'm not. In my sip I've set port=5060, not 55150. The strange thing is that the rport inside SIP packets (sip set debug) coming back from my provider is set to 55150.seen on both Asterisk 1.4 and 1.8 Does anybody have any idea? Thank you. Giorgio -- _ -- 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
