Opensips openser asteriskpráce

I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK format Full source code Simple manual for compiling and generating the application from source Features : -Route call from SIP to GSM -Convert audio from/to SIP and GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or ...

I'm looking for somebody that know asterisk I have a quick task that it's give me a head heck I pay $250.

Using asterisk 12.6.0 / FreePBX 12.0.1rc29 I need a method to add a leg to a call through the asterisk cli or other simple method. The intention is call the asterisk CLI from a simple php exec script. For example, 1231231212 from the outside calls inward, it rings through a ring group, and ext 500 picks up. I want to have a button on my crm call a php script to transfer the call that is ongoing between the outside number and ext 500, to another internal ext 100, or even external phone number. By the end of the function being completed, there will be a joined called from ext 500, the external number, and ext 100. The user at ext 500 will end the call by hanging up, while the call between the external number and the ext 100 continues. In some types of failure, a ...

Need to connect dialogflow with asterisk

Hello! Our company needs help with a full integration of our workflow with vtiger 6.5.0. Our system is running at the moment but we are trying to configure vtiger to our specific needs. We have a work flow ...full integration of our workflow with vtiger 6.5.0. Our system is running at the moment but we are trying to configure vtiger to our specific needs. We have a work flow diagram ready and we need someone to integrate that work flow with vtiger. There are also a few problems that we would like to get fixed. Mostly with pop ups/ history tabs/ pbx manager. We would like someone who is well versed in Asterisk, Freepbx and Vtiger 6.5.0. and has had prior work done similar to what we are asking. We will greatly appreciate any help! Thank you for your time and we hope to hear from som...

hello i have asterisk server i cant access throw asterisk -rvvvv always connect and disconnect suddenly

we need to use api with our device from asterisk or voipswitch or any way pleas check attach file

Using asterisk 12.6.0 / FreePBX 12.0.1rc29 I need help with how to trigger a query to an external program in a situation where i only want to trigger the query on an inbound call that the system forwards (or a user manually forwards) to an external phone number. Example #1: The main line is # 567567567. A call from # 123123123 rings in to the main line. The asterisk time condition/ring group/extensions is setup to forward the call to external external # 987987987. I want to trigger the query to the external program on the forward to the external # 987987987. I do not want trigger the query when the external # is not 987987987. Example #2: The main line is # 567567567. A call from # 123123123 rings in to the main line. The asterisk time condition/ring group/ext...

We run a FreePBX / Asterisk VOIP system. I would like to setup a call center in Salesforce. I need a customized Salesforce call center definition xml file reflecting our configuration. REQUIRED A: Salesforce and VOIP know how REQUIRED B: fluently in English REQUIRED C: Read the requirement and submit only a bid if you have done this before. THANK YOU!

Hi, I have a running astpp Billing server, I am l looking for someone who can help add the following to the application: 1. Striping 011, 00 from called numbers 2. Disconnect Charge 3. Disconnect Charge time 4. Block Charge 5. Block Charge time 6. Fine tune ASTPP Billing and Freeswitch. Please you must have experience in a2billing, ASTPP, VoIP, Freeswitch, asterisk, etc. We're only going to Pay for only 3 hours.

We have two immediate projects that need a developer. Once is a CDR report system, the other a PHP SMS system to integrate with our SMS server. CDR reporting requires the translation of CDR reports from Asterisk and other providers into a report to be added to customer invoices. PHP SMS system is to add SMS and MMS capability from groundwire into our existing SMS Diafaan server systems. Applicants will need to be familiar with PHP 5.6 and 7.

Are you an asterisk specialist? I pay you as a consultant per hour done to configure and form one of our IT member. Around 5 hours a week will be done. We will chat on skype during you performing your work

i want someone configure the asterisk with goip with my sim card so when i will be abroad i can receive or call to my friends and family. I have all the required stuff that is needed for project. i want to do in as quick as possible

I need someone to help me in setting up an asterisk server. The server is already installed, but configuration is needed. Required an experienced asterisk admin. English or French to make a Skype call.

Customize IVR for Asterisk: 1; Play announcements 2; Record 3; Transfer

Both Chrome extension and windows application will provide connection to Asterisk AMI for click2call dial and popup based the caller callID. The chrome extension will change all page phone numbers (based on a phone pattern set in option) to click to call link with a phone display. The extension should provide same capabilities as FOP2 The windows application will provide DLL for other application to get and set events.

i need With modules callcenter dialog CDR load balance NAT will Work With opensips-cp

Elastiks veya İssabel pbx santral üzerinde bir kaç ayar yapılması gerekiyor. Asteriks bilgisi olan uzman arkadaşa ihtiyacım olacak.

Need someone with experience in answering machine detection. All we need you to do is modify the and allow our agents to test that we are not getting voicemails. I understand that it is not possible to get 100% detection, however the amount of voicemails we are receiving is not what it should be.

je recherche à acheter, a louer ou à faire créer un predictive dialer pour les callcenter

We are looking for a virtual phone solution to be accessed with iPhone that can have an auto attendant (option 1: Sales and 2: Support). I nee...Sales and 2: Support). I need to build our own solution, especially we don't receive many calls. I expect that the solution would be in the cloud. The system will have a US number. The developer may utilize technologies and system components like soft PBX, SIP, Asterisk, DID, etc. I need a stable system that would have a cheap running cost. Please give me a plan of what you will do and a fixed rate. Example of optional technologies: Asterisk, FreeSwitch, Kazoo VoIP Cluster, Kamailio, callweaver (faxing), ASTPP, Auto Dialer, opensips, kamailio, Call Center, Elastix, Vicidial, VoIP, Ringless VoiceMail, 3CX, FusionPBX, Issa...

I have flexisip , Asterisk , Opensips working well . I want experienced people flexisip , Asterisk , Opensips Who can support configuration document. Client <--->Flexisip ( Frontend )<---> Asterisk ( Backend )<---->PSTN Please check this link and If You can configure and support contack to me.

I am running a FreePBX instance of asterisk and I was hacked, lots of outbound calls, I think I blacklisted the IP that hijacked an extension and made the calls. Now I see a lot of attempt in my CDR Report with "Congestion" under app and s[from-sip-external] under destination. System number keeps changing and the CallerIDs are all 4 digits. Need someone reliable that I can use for this PBX and other Asterisk enhancements and improvements.

i need With modules callcenter dialog CDR load balance NAT will Work With opensips-cp

I have flexisip and Asterisk , Opensips working well. I want experienced people Who can support configuration document. Flexisip () Frontend Asterisk , Opensips Backend

We use Asterisk PBX along with Free PBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX. Requirements: Inbound and Outbound Calling. Caller ID must be passed on incoming calls. Ability to have more than one Whats App number to work simultaneously

hi, i'm looking for a developer on asterisk, to create predictive dialer with open screen operator callcenter

How much money take you for install asterisk for link with app for retails calls

Hola quisiera saber cuanto me cobrarias por montar un asterisk para revender de llamadas y usarlo en una aplicacion

Hola Francisco M., vi tu perfil y me gustaría ofrecerte mi proyecto. Podemos conversar sobre los detalles por chat. Hola Quisiera saber cuanto me cobrarias por montar un sistema Asterisk para una app que utilizara llamadas

Are you an asterisk specialist? I pay you as a consultant per hour done to configure and form one of our IT member. Around 5 hours a week will be done. We will chat on skype during you performing your work

Goal ------------------------------------------------ Tha goal of this project is to enable our operators to issue an SMS message to a potential customer linking to Google maps with instructions on how to get to our location from their current whereabouts. Background ------------------------------------------------ we got a lot of callers looking for information about directions. The address alone isn't much useful getting them to the location, simply because we live in a very rural area. We've come up with the idea to be able to send them a link to Google maps directly to the phone from the phone call itself. This way we are hoping that the other was received the link click on it, Confirm they know where they are going, and we won't have to deal with repeat calls later o...

Looking for an expert to integrate Vtiger with Asterisk FreePBX What Exactly I need is: 1.Click-to-call right from a lead or contact record to save time. 2. See the callers' contact information on the screen whenever they call 3. Quickly create opportunities and contacts right from the incoming call popup 4. All calls are logged so that you can refer back to call histories later 5. Automatically record calls and link them to contact records in case you missed a detail

Hello, I need some help in Asterisk IAX2 trunking and callflows over it. Experience in the fork Elastix or Issabel is a great plus. Will be a half day job or a bit longer. Kind regards, Danny

I want to develop a application that can do real time call transcribe in a call center environment, you must provide your description of solution in detail on how you can push the audio stream from a traditional call center architecture (IVR/ACD) or in open source environment like Asterisk. how to classify which audio stream belongs to which agent. Currently we already developed a demo code which can receive audio stream from Twilio through ngrok but we understand that in real call center environment it is more complicated which involves multiple agents working at the same time. we study the IBM solution in voice gateway and understand that SBC (session border contoller) probably needed to fork the call and audio stream to voice gateway and further to Rest Server. we need an e...

Need a caller Id spoofed prefrably done on asterisk it any platform you see fit

I’m need a caller Id spoofing system done on asterisk it any platform you see fit with easy interface prefably connected via mobile

I need a system on asterisk or best platform to make spoofed calls need simple interface and preferably to connect via mobile device

I need a system developed on asterisk for caller I’d spoofing where I can input my own number si caller sees this I also have another ivr project needed after depending on success of this project

I want to be able to respond with PRACK with an opensips proxy.

Hi Alex, I am looking for an Asterisk expert who can help me a bit out with IAX2 trunking and callflows over it. I need a half day of education. Kind regards, Danny

I have an issue with Asterisk FreePBX configuration. More details in chat.

Necesito crear un softphone web que se conecte con Asterisk 13.18.2.

Need to add a Number and Voice mail to and exisiting Asterisk FreePBX server.

1) Install vTiger CRM via Installatron (cPanel - GoDaddy) and manage inbound / outbound calls. 2) No sync between IPPBX (Asterisk) and Inbound calls (Exotel). Currently customers dial a vanity number and connect to respective agent groups via Exotel. Agents see only a PRI number whenever a call lands on their mobile phones. Outgoing to the same customer is made via GSM gateway connected to a IPPBX. In order to allow my agents to handle calls / follow-up with calls, we would like to receive calls on our own PBX using the GSM gateway and manage the calls via vTiger CRM integrated with ERP for database.

I need a java application that can upload a FreeSwitch or Asterisk CDR to a MySQL database. Identifying every PBX with an unique ID. I need get the code and not only the .jar.

I need you to build it. i need asterisk voip server setup done on cloud server

...include 2 load balancing only asterisk instances. 1/3 of agents register softphones and use this as a webserver) 2 - Primary server A load balancing dialer instances ( agent webphones are registered on calling server A. These two instances do not have registered SIP softphones for agents.) 1- Secondary calling server B (which does not part of the load balancing. 1/3 of agents register soft phones and use as web server) 1 - Third calling server C (which does not part of the load balancing. 1/3 of agents register soft phones and use as web server) all using: VERSION: 2.14-715a BUILD: 190705-1012 © 2019 ViciDial Group To describe the issue as best as possible: inbound/outbound blended calling is seamless with 40 users registered and taking calls while the asterisk is ra...

...include 2 load balancing only asterisk instances. 1/3 of agents register softphones and use this as a webserver) 2 - Primary server A load balancing dialer instances ( agent webphones are registered on calling server A. These two instances do not have registered SIP softphones for agents.) 1- Secondary calling server B (which does not part of the load balancing. 1/3 of agents register soft phones and use as web server) 1 - Third calling server C (which does not part of the load balancing. 1/3 of agents register soft phones and use as web server) all using: VERSION: 2.14-715a BUILD: 190705-1012 © 2019 ViciDial Group To describe the issue as best as possible: inbound/outbound blended calling is seamless with 40 users registered and taking calls while the asterisk...

I need to place bulk calls using asterisk-java. I am using asterisk and AMI as of now. Please contact.