FAQ Database Discussion Community


How does a SIP based app scheme work?

android,asterisk,sip,voip
This is an abstract question on how the SIP protocol works. Let us say I have a SIP server (Asterisk/Yate). And I have two Android devices that wish to connect to each other to have an audio call. ( I am looking for a purely VoIP call, no need for...

Asterisk ARI call from external to external

asterisk,sip,pbx
I want to make a call using Asterisk 13 ARI from my mobile number to another mobile number. I have tried different calls but all return "Allocation failed" response: endpoint:my_mobile_number,extension:other_mobile_number,context:from-trunk endpoint:SIP/my_mobile_number,extension:other_mobile_number,context:from-trunk endpoint:SIP/my_mobile_number,extension:other_mobile_number,context:from-external endpoint:my_mobile_number,extension:other_mobile_number,context:from-external etc. How can I initiate a call from my mobile phone to another...

Troubles with calls by simple PJSIP softphone via Asterisk

c,asterisk,sip,pjsip
I need to make a simple softphone based on the PJSIP Library to make calls via Asterisk server. I used a simple example from PJSIP official site written on C ( http://www.pjsip.org/pjsip/docs/html/page_pjsip_sample_simple_pjsuaua_c.htm ). This softphone can register on the Asterisk server (to make it work I replaced in the line...

Escaped asterisk (*) in list of characters outputs file in folder

bash,shell,escaping,asterisk
At the very beginning of my Bash script I'm passing a set of characters like this: #!/bin/bash set0="0 1 2 3 4 5 6 7 8 9" set1="° § + \" ç % & / ( ) = ? ' ^ ! £ $ - _ ; , : ....

Unable to hear ringing signal when calling out on a SIP trunk

asterisk,sip,voip
I run an Asterisk server with 10 IAX2 extensions (located in different countries). I am able to call make calls between my extensions without any problems. My asterisk server is behind a NAT router. I have the appropriate firewall/port forwarding setup so that my clients can connect to my box...

Asterisk: how test dial plan?

asterisk
I have the follow extension: _[*#+0-9]./_014.,1,NoOp(_[*#+0-9]./_014. matches Rule rtg-Rotaoi-1) It's generated by OpenVox Gateway, and he allways put _[*#+0-9] on begin of expression. Then I tried to dial many numbers, what number, for ie, matches with this extension? ...

asterisk get credit card info

asterisk,telephony
I`m trying to build a script that will capture the credit card info like card number,cvc and expiration date using asterisk 11.x and asterisk-java library for AMI/AGI integration. Right now I am able to build a script that will acquire that info if it is called via dialplan but i...

Configure linux asterisk for inbound calls

linux,centos,asterisk
Would anyone please help me figuring this out: I have configured Asterisk in Linux CentOS 5. I can make outbound calls from my sip phone in windows machine using asterisk server. But I need to receive incoming calls. My ISP provided me 4 information. Username, Password, domain name & DID...

Asterisk-java originated call's billsec and other data

java,asterisk
Can I take billsec or duration of originated call from response event? I need originated calls final data. OriginateAction originateAction = new OriginateAction(); originateAction.setChannel("SIP/xxxxxx"); originateAction.setContext("xxxxx"); originateAction.setExten("xxxxx"); originateAction.setCallerId("xxxxx"); originateAction.setAsync(Boolean.TRUE); originateAction.setPriority(1); managerConnection.sendAction(new StatusAction()); ManagerResponse mr = managerConnection.sendAction(originateAction); //ResponseEvents mr =...

Trouble with Cross Compiling Asterisk

cross-compiling,asterisk
I've trying to Cross Compile Asterisk for Armhf platform. I was using g++-arm-linux-gnueabihf && gcc-arm-linux-gnueabihf packages for cross-compiling Asterisk with 2 following prerequisite : SQlite3 : ./configure --prefix=/usr/arm-linux-gnueabihf --host=arm-linux-gnueabihf CC=arm-linux-gnueabihf-gcc make && make install Ncurses : ./configure --host=arm-linux-gnueabihf --prefix=/usr/arm-linux-gnueabihf CXX=arm-linux-gnueabihf-g++ make && make install When I cross compile Asterisk...

Asterisk in Nested loops, Java

java,eclipse,asterisk,nested-loops
I am trying to make my code print out the Asterisk in the image, you see below. The Asterisk are align to the right and they have blank spaces under them. I can't figure out, how to make it go to the right. Here is my code: public class Assn4...

Asterisk Integration with Symfony2 application

symfony2,asterisk,voip,telephony,asteriskami
I'm new about Asterisk, it's already installed and I have all host details, what I need is how to use Asterisk in my symfony2 web application; Someone have an idea or he worked on this before? EDIT Here is the list what I should do in my web application: Create...

SIP/2.0 603 Declined, Not able to acheive normal call clearing

asterisk,sip
I am working on an application in which I have to disconnect the call after 2-3 rings,The dialplan of this application is mentioned below [public] exten => BB12345,1,Goto(vivek_star,BB12345,1) [vivek_star] exten => BB12345,1,Ringing() exten => BB12345,n,Wait(7) exten => BB12345,n,Hangup() exten => h,1,NoOp("call dropped") Now the issue which I am facing is,...

The voip service which don't care about ACD/ASR?

asterisk,voip,freeswitch
I need to know which voip termination service(A-Z International Termination) doesn't care about ACD / ASR . ACD (Average Call Duration) Means the average duration of the calls routed bya a VoIP provider. It is a quality parameter given by the VoIP providers.. ASR (Answer-Seizure Ratio) The ratio of successfully...

Create own IVR that will access Database

python,database,asterisk,twilio,ivr
I'm currently interning with a company and have been tasked with researching some methods of using telephony. The goal is to provide our clients with the ability to call in and through an IVR-prompted questions, get information back. The information will be from our database. I have successfully done this...

Firefox crashes when a websocket call answered

firefox,websocket,asterisk,sipml
I'm experimenting with asterisk 13 and sipml5 on a cantos virtual machine, Everything is configured properly. I use Firefox 38.0.1 to make calls from an extension to the other, and till last night everything worked just fine. Now today when I start to call an extension and want to...

Build DAHDI for Beablebone Black

linux,linux-kernel,cross-compiling,asterisk,beagleboneblack
I'm planning to build DAHDI for Beagleboneblack Firstly, I built the kernel for BBB completely by following this tutorial http://elinux.org/Building_BBB_Kernel, make ARCH=arm CROSS_COMPILE=arm-linux-gnueabihf- beaglebone_defconfig make ARCH=arm CROSS_COMPILE=arm-linux-gnueabihf- uImage dtbs make ARCH=arm CROSS_COMPILE=arm-linux-gnu- uImage-dtb.am335x-boneblack modules I used linaro toolchain for armhf (CROSS_COMPILE=arm-linux-gnueabihf-) instead of the instructed one. And then I start...

How do I use REPLACE in Asterisk to escape input

asterisk
I want to escape \ to \\ and " to \" in Asterisk. I've tried to use REPLACE but I can't get it to work. My current approach is as follows exten => sms,1,Set(UNSAFESMSTXT=${REPLACE(SMSTXT,\\,\\\\)}) exten => sms,2,Set(SAFESMSTXT=${REPLACE(UNSAFESMSTXT,",\\")}) ; Echo escaped input to terminal safely exten => sms,n,System(echo "${SAFESMSTXT}") When I...

How to troubleshoot socket connection from Asterisk

c,asterisk
I wrote a module for asterisk that needs to communicate to a service request information an return it, but for some reason my socket does not connect at all. When I telnet to the service it works fine, but I can not figure out why the it returns a -1...

Asterisk 11 queue log to mysql

asterisk
How can i change default storage of queue log from /var/log/asterisk/queue_log file to asteriskcdrdb.queue_log table in MySQL in Asterisk 11?

What does asterisk before brackets on object creation mean in C++?

c++,asterisk,indirection
I was reading an example of a hash table implementation in C++ from a website and saw this. private: HashEntry **table; public: HashMap() { table = new HashEntry*[TABLE_SIZE]; for (int i = 0; i < TABLE_SIZE; i++) table[i] = NULL; } The line with the syntax I don't understand is:...

Laravel DB Join with asterisk and alias select

sql,laravel,join,alias,asterisk
I am trying to get following query on Laravel : select TABLE1.* , TABLE1.COLB AS AAA from TABLE1 left join TABLE2 on TABLE2.COLA = TABLE1.COLB This is the Laravel DB Query code : DB::table('TABLE1') ->leftJoin('TABLE2','TABLE2.COLA', '=', 'TABLE1.COLB') ->select('TABLE1.* , TABLE1.COLB AS AAA') ->get(); But it doesn't work and I get...

Asterisk recording all sound in a call

asterisk,record,monitor
I have an Asterisk, I want to record all sound(tone, ring, pressing number...) in a call not just the conversation between two endpoints. Is there any solution for this requirement? Thanks a lot for helping me!...

How to write Dahdi extra module from beginning?

linux,hardware,asterisk,beagleboneblack
I'm trying to port Asterisk into an armed linux operating system (particularly beagleboneblack). I'm partly done, but when I tried to cross compile DAHDI (or direct compile with build-essential installed on BBB's ) to communicate my BBB with FXO Card, I figured out that there are some modules (wctdm,wcfxo, ...)...

Asterisk & Freepbx : Multiple Day Night Toggles?

asterisk,pbx
I am in a situation where I often need to change the active call flow control (which can only be one..right?). I find that netsting call flow controls is very unhandy if you have 4 different scenarios. So what I would like to do is to create an 4 different...

Asterisk 13 ARI not firing “ChannelTalkingStart” events

asterisk
I have Asterisk 13 configured and debugging all received events, but I can't get it to fire a ChannelTalkingStart event. If I press tones on my phone it does fire ChannelDtmfReceived, so I know it can sorta hear me. Is there something special I have to do to receive talking...

Asterisk create_stasis_message Invalid magic number

c,asterisk
I stuck to send a stasis_message for a self made module to the ARI. I try to use the code example from the documentation : https://wiki.asterisk.org/wiki/display/AST/Stasis+Message+Bus I use asterisk 13 instead example (who use the 12), and some signature are changed. Here is the initialisation : struct stasis_topic *foo_topic; static...

How to cross-compile asterisk with dahdi already cross-compiled for arm?

linux-kernel,arm,cross-compiling,asterisk
I'm trying to cross-compile asterisk for ARM. Everything's fine when I cross compile mandatory modules like sqlite3, ncurses and openssl and then including them when cross-compiling asterisk with their respective option --with-sqlite3, --with-ncurses --with-crypto and --with-ssl. But when I tried to include dahdi, nothing came true. This happened when --with-dahdi=$(DAHDI_DIR)/linux:...

send action to Asterisk Call Manager

php,curl,http-post,asterisk,telnet
I'm trying to make a checker script to check if the IP asterisk have Asterisk Call Manager. I did it by make php script and using curl - the result and the response was Asterisk Call Manager/1.3 Response: Error Message: Missing action in request It's good for right now, it's...

Use of CHANNEL function instead of CDR function in asterisk 13

asterisk,channel,ivr
I am using Asterisk 13 and facing this warning : WARNING[10303] func_cdr.c: Using the CDR function to set 'accountcode' is deprecated. Please use the CHANNEL function instead. Anyone please guide how can I use CHANNEL function to set 'accountcode'? Many thanks....

Asterisk Instant Messaging Errors

asterisk,sip,voip,rtp,instant-messaging
I was trying to setup instant messaging in asterisk server. For the client i'm using Blink Softphone. I did add to my sip.conf [general] accept_outofcall_message=yes outofcall_message_context=dialplan_name auth_message_requests=yes and to my extensions.conf [dialplan_name] exten => _XXX,1,MessageSend(sip:${EXTEN},"${CALLERID(name)}"${MESSAGE(from)}) So this is a simple extension for testing. But when i try and send the...

Passing PHP value to AGI

php,html,asterisk
I'm still new to PHP and asterisk. I am trying to pass the value from an HTML text input to a php page which communicates with asterisk to send an sms text via GSM Modem. So far this is the code that I have experimented on. $num = $_POST['cNum']; $msg...

How to enable Agent Channel Type on Asterisk 13

asterisk
I'm using Asterisk 13 and building a PBX application controller to Call Centers. I'm facing a issue when handling agents, for some reason, Asterisk 13 doesnt have the channel type Agent enabled by default, so I don't know how to do to add an Agent on a queue member. [May...

telnet command not listening asterisk manager events

asterisk,asteriskami
I am trying to catch some log of events in asterisk manager (AMI) with command telnet 0.0.0.0 5038, like in the image below, but nothing appears. [[email protected] vagrant]# telnet 0.0.0.0 5038 Trying 0.0.0.0... Connected to 0.0.0.0. Escape character is '^]'. Asterisk Call Manager/1.3 In Asterisk CLI the connection of manager...

IPSec linux doesn't route SIP connection to trunk

linux,asterisk,vpn,ipsec
Im trying to connect to SIP trunk with Asterisk through IPSec Tunnel and it seems that it doesn't route ok... As I'm coming from OpenVPN I was thinking that IPSec enables some interface and puts traffic through. I will list here my IP-s as X,Y,Z... My configuration for IPSec is:...

Asterisk CLI - reset scrollback

command-line-interface,asterisk,elastix
in an Asterisk CLI, is it possible when scrolling back to have the cursor not to jump back when new output hit the CLI and only revert after a key is pushed or after scrolling back to the bottom? Alternatively, is it possible to limit the CLI output to a...

sip authorization header cannot be added

java,asterisk,jain-sip
I have a question about authentication with jain sip library. I am using jain sip for registering sip accounts to Asterisk server. As soon as I try to add AuthorizationHeader the following error message appears: The method makeAuthHeader(HeaderFactory, Response, Request, String, String) is undefined for the type Utils Here is...

Asterisk / Freepbx / Call doesn't disconnects after hangup

asterisk,voip,pbx
When i call to trunk -> internal number and hangup from SIP client it doesn't disconnect the line. It seems that BYE is sent to wrong trunk or it is authorized with wrong username. [2015-02-16 15:51:01] NOTICE[3053]: chan_sip.c:22109 handle_response: Failed to authenticate on BYE to ';tag=SDdu8i501-snl_004195XX18_NSN_CLIENT' On first trunk it...