VOIP on WebRTC with ASTPP
if you know how to build it please contact me
Read moreif you know how to build it please contact me
Read more+ Ideally Asterisk Coding experience is desirable / MUST + You must have very good experience with Customising Asterisk / Vicidial according to requirements + S...
Read moreAvaya Sip Trunk Configuration (Inbound/Outbound Route)
Read moreDevelop a new pbx system voip with modern interface (need to have images to see) where we use didww for the numbers which are ready and setup call routing, musi...
Read moreI need a network expert who can - Resolve the problem of not provisioning Yealink phones (Use endpoint manager module) from the location that uses the Ubiquiti...
Read morei would like a asterisk and sip software install on my laravel webapp so Each operator is loged into it with an extension is assigned an extension and loged in...
Read moreI need help about A2billing. I have a2billing customize switch server one freelancer person customize this and now he is not responding . i have some issues or...
Read moreDevelop a sip Voip test application: a command line client in golang to connect, receive and place voice calls using sip services like sipgate or linphone. Must...
Read moreNeed to shift Issabel (Asterisk) live recording data store to AWS S3 Storage
Read morei want to build a mobile dialer + VOIP billing with prepay account
Read moreHello I am looking for a few writers who can write quality SEO content for the US, and Australian. You must be good at doing research and being creative with yo...
Read moreHey, Looking for someone to create Call History List from FREEPBX (Based on Asterisk) by their API using PHP. Instant Work.
Read moreQUICK TAKS before monday SIP is working, PJSIP is not working (offline) WARNING[1368] chan_sip.c: Invalid localnet value: 192.168.88.0/88 ERROR[1368] chan_sip.c...
Read moreWe want to provide our cloud switchboard solution via freeswitch. For this, I would like to discuss the project with people who have experience in Freeswitch an...
Read moreI am looking for someone who can do the following : We are receiving phonenumber from API. After receiving we should instantly (really fast (maximum) automatica...
Read moreGoogle Dialogflow kaynağını kullanarak Asterisk veya Freeswitch ile çalışaçak Etkileşimli Sesli yanıt IVR oluşturma.
Read moreI'm using FreePBX 14 and asterisk 18. I want config FreePBX to work with sipml5 or jssip. I'm trying to config webrtc, websocket to work with sipml5 but I'm not...
Read moreI am looking for Asterisk Developer, I want to install Go AutoDial or Vici dial installation on our server.
Read moreWe have the XactDialer module in our FreePBX. We are going to implement the API CRM in our custom CRM. And we would like to instead of importing CVS campaign to...
Read morewhat i would like is an asterisk and sip so Each operator is assigned an extension to use a SIP software in order to answer customer calls. What I want is a con...
Read moreHello , We are looking for the expert who can develop the Sip Mobile dialer for Android and Iphone Application and Sip Voip Tunnel , Where sip mobile dialer and...
Read moreWE ARE LOOKING FOR ENTHUSIASTIC INDIVIDUALS MALE OR FEMALE WHO ARE INTERESTED TO WORK FROM HOME ON COMMISSION BASIS FOR SALES PROCESS OF VITAMINS AND SUPPLEMENT...
Read moreAsterisk full setup in our enviroment ( 10 user , 6 cisco phones).
Read moreWe currently have an asterix pbx without any userinterface. We want to have vicidial added with a fresh new custom theme that you have to provide. The call flow...
Read moreHello, We need someone who can build a fax-to-email module for Asterisk. DM for details.
Read moreWant to provide toll-free service to my client for usa UK Canada numbers
Read moreI wana provide tollfree services to clients for usa canada and uk
Read moreI am looking to get Asterisk events via AMI like incoming calls ,queues,ringing , answered , duration,hangup,recordings. Wewould like to show up pop up on agent...
Read moreI want a telecaller who can convince clients.
Read moreI need Asterisks IVR update there is a some issue with IVR
Read moreI need to record ivr welcome message for my company with the text below first record you have reached Consult navigation agency second record for the account de...
Read moreHELP ME TEST RUN MY TWILIO SMS SENDER SCRIPT CODE. I Have twilio sender script and twilio registered account, I want you to help me testrun and fix it
Read moreHi I use a webphone (SipML5) with Asterisk 16 and chan_sip with the opus codec and tls certs. I want to use Homer to monitor the voice quality but I'm unable to...
Read morea test panel for asterisk to make calls and calculate for users , only experts in this
Read moreWe require assistance to configure a Grandstream UCM6202 and a handset. Two lines including Google Voice.
Read moreneed fix this issue , >> 502 Bad Gateway Bitrix Environment (Budget: $15 - $25 USD, Jobs: Asterisk PBX, Linux, PHP, Software Architecture, VoIP)
Read moreneed fix this issue , >> 502 Bad Gateway Bitrix Environment
Read moreWe want to use Telegram as trunk SIP in Asterisk. System needs to route the incoming SIP calls, through Telegram accounts and to manage all hangup causes.
Read moreI NEED TWILIO AND NEXIMO SMS SENDER SCRIPT CODE. I HAVE TWILIO SMS ACCOUNTS, BUT I NEED GOOD SENDER SCRIPT. AM READY TO PAY
Read moreHallo werte Jitsi Experten seit einem Update von Jitsi funktioniert leider die FreePBX anbindung nicht mehr. Es konnten Sich bisher die Teilnehmer von extern pe...
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