I'm looking for a tech who already has completed a Ringless Voicemail drop system. We are US based company and will target users in US only so will calling 10-digit US phone numbers. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. If you Already have completed a ringless voicemail drop system please bid
We have a hmailserver and require the config to be tweaked, and TLS or SSL to be installed. Our server has been running for a few years bit is rejecting mail from some ISPs due to misconfiguration. Job scope:Install self signed server certificate. (freelancer will have to generate certificate)Make mods to hmailserver to ensure mail is not rejected
...VoIP provider. I gave the green color for your easier to understand. -- Executing [s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in new stack -- Executing [s@macro-user-callerid:38] Set("SIP/8902050098-00000024", "CALLERID(name)=+918820094576") in new stack [login to view URL]: Caller ID name is...
...[s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in new stack -- Executing [s@macro-user-callerid:38] Set("SIP/8902050098-00000024", "CALLERID(name)=+918820094576") in new stack [login to view URL]: Caller ID name is '+918820094576' number is '8902050098' This should be a ver...
...checkout page the fields, Pais, Endreço. Cidade,Estado, CEP, they are [login to view URL] I do not know why the red asterisk does not appear. I installed the plugin WooCommerce Checkout Field Editor, although I enter the fields as mandatory, the red asterisk does not appear , and it is not possible cancel the writing (OPTIONAL). mysite: [login to view URL]
Looking for an experienced contractor to update configuration on our VOIP environment (a dozen or so phones). Currently using PiaF / FreePBX but happy to change to another distro if needed. Most phone are Mitel/Aastra 6739i and plus a couple cheaper aastra and a polycom conf phone. Key Issues / Targets: 1. Hot-desking - Configuration to allow for user speeddials to move with users as they log i...
Asterisk PBX free software programmed for six internet phones which we own. Three 800 numbers which we now have to be transferred to this system. Standard small business features: Transfer calls, Voicemail including after hours message, desktop window showing who is on a phone call, etc.
...retire our incoming ISDN lines and are setting up to test sip lines. We have an unusual router (peplink) and multiple redundant internet connections. We have spend many hours trying to setup our router to enable SIP connectivity however without success. We are looking for someone with 3CX, SIP and good networking /router skills. Hourly rate to be discussed
...The purpose of the program is to extract data into CSV files from the Xero accounting package. It has been working perfectly for several years. But now Xero require the use of TLS 1.2. Hence we must update the program. Note: You do not need to program new functionality, as the program does exactly what is needed. This job it to get it working with a new
Hi Kristen H.,We would like to hire you to prepare a new catalog of @ 170 products currently on a GSA contract to mirror and add to GSA Advantage through the SIP database program. All files will be supplied to you. You will need to organize in the correct format and upload both product descriptions and photos.
Hello, We need to add a functionality to our IVR which is based on Asterisk V 13.14.0 / PhpAGI. Os is Debian 8, database is MySql 5, Php is also 5. Simple functionality: - Inbound call accepted (client who needs support) - IVR (PhpAGI) says "welcome" - Call is forwarded to 1st level agent (already done by DIAL command) - 1st level agent takes call
...We offer: - hourly wage: 15 USD/hour; - wages minimum 10000 UAH per month; - work in a prospective company: we introduce automation systems based on open-source products Asterisk IP-PBX and CRM VTiger: open API, easy integration with other systems, more than 30 own developments for CRM VTiger, VTiger has wikipedia and community community developers;
...(landline 1) and I have my own server too. I'd like to redirect calls made to landline 1 to another landline (landline 2) that I don't own. I've already made a vocal robot on Asterisk so that depending on the digit the caller presses, it does something different. What I need now is the last part : 1) Depending on the digit pressed, forward the call to a
I...Xero API. This worked well for a few years. But now Xero requires that TLS 1.2 be used. I need my program to be changed so that it is compatible with TLS 1.2 In the pictures I show some of the source code files and the compiled files. You will update these to be compatible with TLS 1.2. You will supply the updated source code and compiled files.
We need an HTML based SIP client that can be designed to look and act like a in home intercom. For example, there should be buttons for rooms, that will let you page the rooms, and select either video or audio. This will have to be set up that each "station" can be configured which rooms it can page etc... There are a lot more features and customization
App to register with my Asterisk Server as a SIP extension. My Server will send VoIP calls to the App and the App will make a local call on the GSM network and path both calls together. In other words, the Android Phone will act as a VoIP / GSM gateway. Thamk you.