Asterisk How to Dial to Originate a Call from Within the Dialplan?

Salvete! How can I dial a number and have Asterisk originate a call from extension sipX to sipY? Both sipX and sipY appear in extensions.conf of my dialplan. The trick is that I want to dial 337 on my phone, and then my phone goes out of the picture, then sipX calls sipY. Say I want to be able to push 337 on the phone, and have a sound played over the speakerphone of another phone, say, as an alarm. Another way to consider it is, how can I do this: Push 337 on my phone complete my call,

Oneway conference calling through asterisk

I am new to Asterisk and Voip. I wanted to accomplish a following small thing using Asterisk. Description Asterisk is used as server Several Voip clients. (Two types of clients possible. One which can start a conference call, other can't call but can only hear.) Only caller client can start/end this call. The call can't be longer then a particular time. Is it possible through Asterisk. How does asterisk help to implement this scenario. What does I need to learn? Any web links will be very

how to delay sip 183 in asterisk

my calls did not receive ringback tones but went to the IVR system in less than 4 seconds. When reviewed the SIP captures on my end and noticed that the SIP 180 Ringing message is followed instantly by SIP 183 Session Progress with SDP. The SIP 183 with SDP is indicating that my asterisk server is ready to send through audio and since there is no ringing within audio streams so, no ringback is observed. So, please tell me how to put delay in SIP 183. I am using asterisk 1.4 in centos 5

Asterisk record a .wav file and save a same file name of call file

I make a script that automatic create a .call file and auto-move to /var/spool/asterisk/outgoing/, im done with that but in recording i want to save a .wav file same as .call file FILE EXTENTIONS.CONF [outgoingcall] exten => s,1,Answer( ) exten=>s,2,Record(/var/spool/asterisk/tmp/${CALLFILENAME(name)}/${STRFTIME(${EPOCH},GMT-8,%m%d%y-%H:%M:%S)}.wav,0,0,qxk) OUTPUT: in /var/spool/asterisk/tmp/testcalls150(date-time).wav Thanks in Advance, OAcebes

Asterisk Need basic advice on creating a phone based search engine

I am thinking to build a Telephone based search engine. The concept is simple: User Dials the number. We record his input and convert the speech into text. Use Google API to search for the query. Fetch the top results and convert them into speech. Send output to the user. I'm comfortable in coding the mechanism. But i don't know how to implement this on a telephone line. I will need a IVR which will guide the user and a back end application for processing. I can code the backend application.

Asterisk what entries need to do in .conf files for call forwarding

I'm using Asterisk 10,centos 6. and I'V done changes in conf file as below for call forwarding "extension.conf" exten => 0010,1,Wait(0.05) exten => 0010,2,Queue(0010) exten => 0010,n,Dial(SIP/0011,15) exten => 0010,n,Dial(SIP/0012,15) exten => 0110,1,Dial(SIP/0110) exten => 0210,1,Dial(SIP/0210) "queues.conf" ;----------------------QUEUE TIMING OPTIONS------------------------------------ timeout = 15 retry = 5 ;timeoutpriority = app|conf timeoutpriority = conf [0010] member

Asterisk CDR duration difference on DB and h exten

I'm using an AGI script on 'h' extension to compute call cost and save it on CDR(userfield) variable. The problem is I'm getting duration differences between CDR(duration) variable and the value stored on MySQL. Mainly, duration stored on DB is 1 or 2 seconds greater than variable, like the inserting process delays and duration keeps counting. I need to know why I am having this duration difference and how should I avoid it to bill on the right way. Thank you very much.

Asterisk channel originate, how to do call from a local channel? (call intercom and send dtmf)

My goal is to : run a background task activated by dynamic feature while in active call, that will execute dial to another EXT and send DTMF. It means, when a user is active call with someone, when the user press 5555, the door will be opened. In order to open the door today, I have to manually call EXT 6(the door) and send DTMF digits: 00* All of this has to happen automatically when the user press 5555 without interfering the active call. I tried before to do all of this with dial, but dial bl

Asterisk Call Transfer to Playback then resume back call

Here is my scenario i want to achieve, I have do outbound call to a number, after conversation i need to transfer the call to play a message for callee. After the message end, the call will route back to caller continue conversation. How am i going to achieve this. using method MeetMe? or others?

Can I set the asterisk_version string

I am trying various different options of building Asterisk 11 and these will be deployed on various servers. They are all built from the same sources and have what I presume to be some sort of checksum embedded in the version ID (26dd464). In order to distinguish the various versions of the executable I would like to add my own version number or string on similar. I note that /usr/src/asterisk/main/version.c specifies a const char [] variable asterisk_version, but if I manually edit this it get

Asterisk Regex check if number entered has 10 digits

I am using Asterisk 11, and I am new to it. I have added the following to my extensions.conf file: exten => 2, 1(start), Read(callBack,enter-phone-number10,10,,5) exten => 2, n, GotoIf($[${REGEX("^\d{10}$" ${callBack})} = 1]?confirm) -->not working, goes back to Start exten => 2, n, GotoIf($[${REGEX("^\d{10}$" ${callBack})} != 1]?start) --> works, goes back to Start exten => 2, n(confirm), Read(done,if-this-is-correct&press-1&otherwise&pr

Asterisk How to get the dialed number in Stasis app

I am trying to wrap my head around ARI and Asterisk, my goal is to dial from an extension to another. I dialed 5001 from extension 5002. Now in the stasisStart function, I want to create a new channel, and used the dialed number (5001) and pass 'PJSIP/5001' to the endpoint. How do I get the dialed number? Dialplan: exten => _500Z, 1, Stasis(test-app) test-app.js function stasisStart(event, channel) { // I want to dial 'PJSIP/5001' (the dialed number) client.channels.originate({

Asterisk Open Source Video Conferencing Server

How to create a Open Source Video Conferencing Server using Asterisk? Is it possible? I would like to create a n-way conference call but I cant using MeetMe and zaptel. I don't have Digium hardware. I am using asterisk 1.8 on ubuntu 10.10. Need some advices. How to do this? APpreciate!!

asterisk play music while waiting for DTMF

i am setting up a DISA in asterisk . what i would like to do is play a music to the caller while asterisk is receiving the DTMF. the DTMF would stop playing when it receives DTMF for "#" i have tried using background() and playback() command , but they do not serve the purpose! any ideas? khan

Not able to detect keypress in asterisk

I am using asterisk 10 with confbridge and have set a menu option in confbridge for key press 1. I am using voipvoip sip trunking services to call to phone numbers. The problem is sometimes when the user presses 1 or any key then it's not detected(The menu option defined for that dtmf is not performed). This is particularly happening on Indian phone numbers. Please this is only sometimes not everytime. Please help me, is this any configuration issue or something else.

Asterisk WaitExten during the call

Scenario: Asterisk receives incoming call to a DID number Asterisk forwards incoming call to a mobile number using PSTN termination. Call is answered on a mobile Question: Is there a possibility to transfer call to a different extension during the call ?

Asterisk FastAGI sending command at the same time

I was using Asterisk.Net for my AGI and my asterisk server can accept a command which convert the voice to text ( Now, I want to perform two commands at the same time which is the wait for digit and the voice to text so that it can accept key press and voice for authentication. How can I achieve this or it is possible to do?

Auto dial out issue in asterisk

I am applying an auto dial in asterisk using .call file My Channel: DAHDI/g0/09********* MaxRetries: 1 RetryTime: 600 WaitTime: 30 Context: outgoing Extension: 10 Priority: 1 My problem is that every time above number is called by same number means even if i change the dialled number(receiver number above) the caller number is same.How can i set the caller number in an outgoing call? Thanks in advance.

Unable to play music on hold in asterisk

In my dialplan music on hold was working earlier but its not working now.My musiconhold.conf ; ; Music on Hold -- Sample Configuration ; [old-default] mode=files directory=/var/lib/asterisk/mohmp3 random=yes [old-quiet] mode=files directory=/var/lib/asterisk/quiet-mp3 #include musiconhold-vicidial.conf ; valid mode options: ; quietmp3 -- default ; mp3 -- loud ; mp3nb -- unbuffered ; quietmp3nb -- quiet unbuffered

Asterisk Kick all user from confbridge when one user left

I have a problem,if a single user left the confbridge or disconnect his call... I want to hangup calls of all other users who are in that particular conference room...Any idea regarding this??? Basically I want to disconnect all channels if any of the channel hangup the call.Any guidance? Many thanks.

Receive Talk Detect events From Receiving channels in asterisk?

Let me explain my scenario first, what i am trying is to detect channel talking and silence events during call, and perform some task on event detection, i have successfully detect 'talk_detect' events on the channel who initiated the call but i am not able to detect the 'talk_detect' events on the channel who receives the call, here is a code sample: Dailing channel: exten =111,1,Answer() exten =111,n,Set(SPYGROUP=3300) exten =111,n,Set(DENOISE(rx)=on) exten =111,n,Set(TALK_DETECT(set)=1000)

Call hangup right after dial in asterisk

I am using asterisk 11 and my call hangup right after dial command and shows bellow error Retransmission timeout reached on transmission Mydial command is AGI Script Executing Application: (DIAL) Options: (SIP/112233@ Call works fine on default port (5060) in this case not work on given port 8060. Complete Debug: Everyone is busy/congested at this time (1:0/0/1) [Apr 23 17:27:42] WARNING[9213]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on t

Integration of AGI and its Working in Asterisk

I am very novice in asterisk telephony word and trying to learn how asterisk can be used in a professional way for delivering it to the customers.While searching it on google it got to know that Asterisk Gateway Interface is the way we can follow and proceed to write dialplans and do necessary changes.Also i got to know that Astive Toolkit(ATK) is a java based Tool by which i can step ahead.I have downloaded the ATK toolkit but i dont know how to install it and use it with the asterisk server.

Asterisk How to get domain of caller SIP account with PHPAGI?

I would like to write AGI script for Asterisk with PHPAGI which do following: + Check caller is SIP account or not. + If caller is SIP, hangup and redial to caller The problem I have is that I can only get caller SIP user by following code: $ret = $agi->parse_callerid(); $domain = $ret['host']; $user = $ret['username']; $user has correct value but $domain is always empty. How can I get domain of caller SIP account?

How to config instance messaging for asterisk 12

I am using the asterisk 12 with pjsip, I wonder how could I config the instance meesseging for pjsqip in asterisk 12 ? What is the default message context for pjssip ? I use the default extension.conf from the installation and I successfully could make the call over each but when I try to send message, it does not receive by the client.

Exporting asterisk cdr to phpmyadmin

I'm working on a PHP probject using Asterisk.I need to store Asterisk CDR in a database .I want to know how could I connect Asterisk to phpmyadmin.I installed Asterisk on centos 6( which is installed on virtual box) and phpmyadmin is installed on another system.

Asterisk How to put call on hold via cli or ami or agi

I tried already to use Park app in ami but the point that is always hangup the other channel for example I have this two channels PJSIP/600-00000076 PJSIP/300-00000075 and the already bridged and I want to put PJSIP/300-00000075 on hold and let the other channel do some stuff and then go back to PJSIP/300-00000075 I want to do that without any interaction from the phone just cli or ami not I tried this (AMI) Action: Park Channel: PJSIP/300-00000075 Timeout: 0 but the problem is that is ha

Asterisk cel_pgsql Inserting duplicate columns

The log shows there is duplicated columns, insert will fail. [Aug 29 08:14:42] DEBUG[8683] cel_pgsql.c: Inserting a CEL record: [INSERT INTO cel ("id","eventtype","eventtime","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","amaflags","accountcode","peeraccount","uniqueid","linkedid","userfield","peer","id","eventtype","eventtime","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","

Seek Help concerning IVR Menu in Asterisk

I am writing an IVR menu and I need to allow my users to press 0 anytime during the communication to exit. The following is how I do it: exten => 0,1,Playback(good-bye) exten => 0,2,Playback(beep) exten => 0,3,Hangup However, by doing so, when the user presses zero while some file is being played back or some other operation is taking place, he/she cannot exit, it is like if he/she didn't press zero. I hope I am clear enough and that you can help me out with this. cheers

Asterisk Cannot record call after fetching it from parking lot

I'm using asterisk/freepbx. Asterisk is not recording calls that are fetched from park. Here is my call flow. A calls to Asterisk Server(AS) Call is picked up by extension B B does an attended transfer by dialling *2200 (200 is my default parking lot) C dials 1 to fetch the parked call C dials *1 to record the call. Recording is not done.

Asterisk - Any way to end/cancel the queue wrap-up time?

Any way to end the queue wrap-up time? We define a wrap-up time of 60 seconds to allow the agents to finish their notes from a call. In some cases, the full 60 seconds is not needed and our agents would like to end their wrap-up time. Note:I dont want use pause/unpause commands, because I am tracking the pause/unpause events to track the breaks (metting, launch, break, etc) times.

Catch hangup while Asterisk AMD is checking

Im having this problem just when i answer the phone and then hangup, but asterisk does not detect the hangup while AMD is detecting ? Asterisk 11.11 -- Executing [09XXXXXXXX@appel-sortant:10] NoOp("Local/09XXXXXXXX@appel-sortant-40f9;2", "Next = 0") in new stack -- Executing [09XXXXXXXX@appel-sortant:11] Set("Local/09XXXXXXXX@appel-sortant-40f9;2", "GLOBAL(NEXT)=0") in new stack == Setting global variable 'NEXT' to '0' -- Executing [09XXXXXXXX@appel-sortant:12] Dial("Local/09XXXXXXXX@appel

Why is my asterisk PBX not registering an extension but registers the sip lines from

When making a call from a cellphone to the SIP external number 416XXXXXXX the call is going straight to the Ext voicemail, then to email. Call log shows the following: [2015-06-02 13:52:49] WARNING[13331][C-00000003]: func_channel.c:538 func_channel_read: Unknown or unavailable item requested: 'reversecharge' [2015-06-02 13:52:49] WARNING[13331][C-00000003]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) Looks like my extension to regis

Path Translation issue in Asterisk PBX DialPlan

Asterisk Pbx cannot translate path for Calling from Cell to DID. when i call from Cell to My DID Number then Warning in Pbx occuring No path to translate from SIP/1005-000002cc to SIP/ForNishantPBX-000002cb Spawn extension (from_NishantPSTN, 14692498805, 1) exited non-zero on 'SIP/ForNishantPBX-000002cb' following is my extensions.conf configuration for DialPlan: [from_NishantPSTN] exten => _14692498805,1,Dial(SIP/1005,20) here i am calling from my cell number to DID number: 14692498805

In Asterisk, should I worry about reloading configuration files during calls?

I have a small application that reads the extensions, peers voicemail info etc from an Excel file and generates what Asterisk understands: sip.conf, extensions.conf, voicemail.conf, etc... Once its done generating the configuration files it reloads them by executing: sip reload dialplan reload voicemail reload moh reload // etc... Is it safe to reload all this while I am using the system? For example if someone is in a call and I reload all those configuration files can a problem occur? Shou

When using asterisk ari, am i meant to develop my own softphone ?

i'm a quite a rookie at asterisk development, i understand some basic fundamental concepts of channels and bridges . i understand that channels are created by the channel driver written in c when using a dialplan configuration.i have been able to configure sip phones to make inbound calls and outbound and also been able to configure a local sip provider to make trunk calls. The issue i have is with ari, what i dont understand is if i am meant to develop a soft sip phone (on a browser) from where

Asterisk audio capture on another machine on same LAN

Using Asterisk 13.12.1, which working fine. Also setup an AGI (AsterNet) on remote windows 10 machine which working fine too. Able to route calls to AGI on remote windows 10 machine fine, using like - exten => 1001,1,agi(agi:// Problem - I need to capture audio too on remote machine. There are different ways like EAGI, JACK_HOOK etc. But not able to find an starting point to use those. 1 JACK_HOOK - configured jack hook like - exten => 1003,1,Set(JACK_HOOK(manipulate,s

Asterisk, FreePBX and siprec

I recently setup a FreePBX (version with asterisk (version 17.9.3). I now have the idea to connect this FreePBx to a SRS to be able to record my calls via that external server that is compatible with the SIPREC protocol. I'm trying to figure out a way, but I can't any instructions anywhere. All I can find, is instructions to record calls locally on the server, but I need them to be recorded via this SRS using the SIPREC protocol. Does anybody know where should I start looking for inf

Asterisk Unable to create channel of type 'DAHDI' (cause 17 - User busy)

I have 6 Red FXO with TDM2400p in my PC. I have install asterisk and dahdi driver. Scenario is jitsi-----> asterisk server-----> analog PBX ----> landline phone I configured this scenario as follow in chan_dahdi.conf file ; General options [channels] usecallerid=yes hidecallerid=no callwaiting=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;FXO Modules group=2 echocancel=yes signalling=fxs_ks context=Incoming channel=1-20 After loadin

How can I increase the default ringing time on a phone using asterisk

I have an asterisk pbx setup on my server. We call to phone numbers using our pbx asterisk 11. The default ringing time on the phone numbers seems to be very low around 20 seconds. Is there any way in asterisk to increase the ringing time, if yes then please let me know how can I do this. Regards

Asterisk disable announce_only_user in confbridge

When first user entered in conference using confbridge, prompt played to user that he is the only person in thee conference room and its due to the default setting in confbridge.conf, I want to disable this prompt.Anybody can please give me how to disable this thing.

Asterisk: Where asterisk stores it's data

I'm finding the place that Asterisk stores it's data. For example: Command "Show sip peers" will return all sip peers with their IP address and status. I wonder where they were stored in the drive. I didn't install MySQL or any DB

Asterisk Maximum waittime in call file

I create a callfile as below: channel: SIP/To_Avaya_PEER/20022 callerid: 1788888888 waittime: 300 context: from-primas extension: 100 priority: 1 account: primas archive: no alwaysdelete: yes I want to wait for answer longer than 3 minutes or forever. I want to keep the call in queue without using MaxRetries. How can i do? I set waittime longer but it doesnot work. Please help.

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