Hi Sunil I saw that you have experience with SIP and VoIP, on android and iOS I need someone to build iOS and android App. If you are interested we can continue talking on chat See you
Hello There i need some help to make some config on my FreeBPX server like: 1-Remove and add new SIP line 2- there is problem to login to the control panel after i tried to make restore backup . so can you do it for me and home many time you need to finish it note: there is teamviwer access and then you can access the Unix server by web
Hello There i need some help to make some config on my FreeBPX server like: 1-Remove and add new SIP line 2-make my system cluod on internet so the agent can use there extension in anywhere 3- there is problem to login to the control panel after i tried to make restore backup . so can you do it for me and home many time you need to finish it
We have a few Skype Connect profiles with inbound numbers at...script should (persistent) connect to Skype servers, login and answer the calls by returning a recording (depending on the inbound number). That's it folks:-) It's all standard SIP, pretty straight forward. --> $100 FIXED (it's max 3 hours work, MAX, 30 minutes if you've done it before)
Hi, I look for Freelancer who can implement Asterisk with my MGCP provider. Not SIP. Can somebody do? Thanks
hi i want to register sip to sip and pass call g729 problem is i dont want to use big pc and lots of HW so i found a program its allow to do that i want [ログインしてURLを表示] i want to run this program on a router Router Link [ログインしてURLを表示] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580
Help programming sip endpoints, applications voice & sms in [ログインしてURLを表示] I need to connect my [ログインしてURLを表示] account to my PBX, be able to forward DID numbers to another number, and direct sms's messages. I would also like to complete the project described here [ログインしてURLを表示] Prior experience
In need of software or asterisk/freepbx hosted solution for "Voice Broadcast" or "Robo Dialer". - Connects to SIP Trunk - Alternate Caller ID; Area Code Caller ID campaigns with alternating numbers - Ability to play and add audio files messages that are randomized - Multichannel lines (100) to make hundred or thousands calls per minute - Choose number
I need help with setting u...setting up the Outbound Parameters of my 3CX-based SIP Trunk (in the cloud). I am able to make & receive calls using Nexmo. But with Anveo, I only succeeded one time to make a call by tweaking the Outbound Parameters in the 3CX SIP Trunk Setup and I was never able to reproduce that again. Knowledge of SIP trunking a MUST.
Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for this
i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole
I’m looking for an expert of VOIP, SIP, Hack for educational purpose
I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.
i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know
you have to install skype for business 2019 and set it up. we have multiple sip phone numbers which have to be also connected to skype for business 2019 for in and outgoing calls.
i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for
I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled
...com/click-to-call/ etc You don’t even need a fax machine…,Receive Fax from customer as PDF in your Email. The system will support setting up/administrate the call center SIP accounts (with hunt groups). Of course system needs configuring Asterisk for custom DID:s, call forwarding, internal calling, voicemail, ability to activate/deactivate service
Hi, I deliver VoIP services to my customers and i use 90% of the time the Yealink SIP phones. The problem in 50% of the cases is that when the client is turning on their voicemail, they can't see it anyway if it is turned on yes or no. I know it is possible to use something with XML and a webserver to create a connection with the phone which can enable
required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB
I need someone to setup my softphone with vpn and connect to an existing hosted pbx. my sip and voip are blocked by my isp
Cisco voice gateway connection to SIP server and PSTN on E1 PRI
I need someone who is expert in hacking, also expert in networks, cyber security, VOIP, Telecommunication servers.
...3.64, but i cannot setup outside domain, i want to use sip outside network, i know there are instructions but i cannot fix the problem. I know i just miss very little thing. i will give you teamviewer to setp up my pc. You need to do: 1)help me to set up sip system using sipxcom, i need the sip can be use in internal and external network. 2) tell me
I have just registered new firm in financial services. Stock broking, mutual fund, SIP investment advising, Various types loans & insurance services will be our products. I need logo & letterhead for this.
WE need to connect a cisco 7206VXR with VIC adapters for voice connection to the PSTN and to a Mizu SIP Server for users authentication and billing using sip/h323 protocol. We already done a major part of the setup and configuration of both the cisco and the softswitch but calls are still failing to authenticate on the cisco gateway to the pstn network
1 - We need to configure Asterisk FREP...documentation from the work with detailed description with step by step of how to configurate it. 3 - The developer will register some of our SIP channels into the freepbx. We currently have both sip clients: - SIP Phone - SIP on PC 4 - We need someone who is an expert on it, because we need it asap. Thanks.
...discutere i dettagli tramite chat. A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [ログインしてURLを表示] secret GZD397XU Host [ログインしてURLを表示] port 5060 Register String 390415541985@[ログインしてURLを表示]:GZD397XU:390415541985@[ログインしてURLを表示] B) TRUNKS MESSAGENET Protocollo SIP username 5406073184 From Domain [ログインしてURLを表示] secret ajFFzpNj
...to amend for us. Do you have IAX SIP VOIP Softphone system and apps? We would like to do similar system like www . zoiper . com Zoiper is a phone solution perfectly fit for end users, service providers, call centers or any business willing to benefit from VoIP communications. Zoiper softphone is a standalone client-side software VOIP phone application
...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are looking for someone who can optimize our cutting routines and prepare new
Lightweight SIP Client with Dynamic CRM which would display the customer information. CRM Field and information would display dynamically and action should collect and send to server Database. Required Features: SIP Call with transfer/conference/hold/Mute facility SIP messenger. Dynamic CRM-Dynamic field header and value display for each call according
I need some one who is expert in cyber security & hacking and specially VOIP, Sip, you should be expert in this field.
I need professional help with setting up duplicate ...• 18 Avaya Phones – Model 1416 • 23-24 extensions Needed • 4 Trunk Cards That I see • 5 Analog Phone Numbers Following Licenses currently on system: • SIP Trunk Channels • SIP Trunk Channels • SIP Trunk Channels • Essential Edition • Essential Edition Additional Voicemail ...
as the title indicates, need sip installed on cisco 7960 to work with astersik. total 10 phones
Hi. We need to configure asterisk on our ubuntu machine. We need someone high skilled to do the job. We will also need to make it visible over internet with public IP. Also we need to configure some IP phones on it. After all, we need a manual with step by step about how to configure it.
I have lan calls working fine, but require someone with Cisco/Asterisk SIP knowledge to help resolve this issue with NAT, no audio not working.
I require a voicemail d...csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [ログインしてURLを表示] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.
we need a Predictive Dialler with freepbx to link to Qs and have a wall-board with reports. reports need to link to recordings. this all has to work on freepbx and sip and softphones. The PBX will make the call and the agent when signed into the Q will receive a call. the agent phone rings first then the PBX makes a call according to a predefined list
Hi I need an application that can receive a call through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. The app should be able to run on cheap devices (+-100$). I was thinking on a way to emulate a headset and set it as default
...to (IAX2/SIP) and vise versa. Requirements: 1- Work on any android mobile platform especially low price mobiles. 2- Register by SIP or IAX2 to any required server. 3- if the mobile have two SIM cards it should work for both and each by separate registration. 4- send calls from sip/iax2 to gsm and receive calls from gsm and send it to sip/iax2 5-
...provisioning by email address login. Users should instead enter their mobile phone number to register. After receiving an SMS with a verification code the softphone should download the SIP provisioning data from the server. - Create IOS version and deploy to appstore - Create Android version and deploy to playstore Server side works - Install vialer-middleware
...1main server other all asterisk server work like VM host1 IP [ログインしてURLを表示] SIP 5060 DB NAME datastore1 host2 IP [ログインしてURLを表示] SIP 5060 DB NAME datastore2 host3 IP [ログインしてURLを表示] SIP 5060 DB NAME datastore3 host4 IP [ログインしてURLを表示] SIP 5070 DB NAME datastore4 host5 IP [ログインしてURLを表示] SIP 6060 DB NAME datastore5 i hope you understand if you can make it please
Hello All, We need to integrate Vonage and Callcentric SIP with our Bitrix24 installation for calling to USA. This will be used with a 10 PC user system for a call center. Other requirements with same: - SIP set up in Bitrix24 - Setting user right for 10 users with extendable option to more users. - Setting up rules for call forwarding, call picking
We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Proficient in Lantek program (lazer tube cutting) is a must. We have ongoing work for the right programmer on a long term basis.
...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Artificial Intelligence: It involves Studying Human thought processes representing thought process on machines. Knowledge encompasses the implicit and explicit restriction
...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are looking for someone who can optimize our cutting routines and prepare new
URL Free to Call System from the website, to use my sip trunk for clients to call me free.
...DTMF button on screen so user can press options from IVR and user use inbuilt microphone then connect call to SIP address on Incredible PBX OR When number clicked on "Free Call" button, viewer types in his/her phone number and submits to SIP address which setup on Incredible PBX Example [ログインしてURLを表示] The right candidate will be also contracted
ccnp voice training and configuration for voice gateway mgcl sip and h323
