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 [log ind for at se URL] i want to run this program on a router Router Link [log ind for at se URL] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580
Help programming sip endpoints, applications voice & sms in [log ind for at se URL] I need to connect my [log ind for at se URL] 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 [log ind for at se URL] 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
1) The VAT is not being calculated for my WooCommerce website. 2) I have physical products and downloadable products. The problem is, when I add the downloadable products to the shopping cart, then the cart gives an error message "There are no shipping methods available. Please double check your address, or contact us if you need any help.". 3) The WooCommerce product images are too ...
Configure NextScripts SNAP Pro Plugin for Wordpress, add API and configure for about 10 social networks, test all features, add CronJob to server You must have experience and testimonials to that fact and proof of knowledge. No beginners.
We have an ecommerce website and need support to create new email templates. We also need ongoing support with our wordpress website.
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 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
...billing is live so i require someone with experience. This is my third project to get this done. If you cannot show WHMCS experience you will be ignored. 1. I require my configuration checked as I have issues/error with account creation and upgrades. 2. I require domain api and ssl api to be configured and working. If i can find someone to do this cost
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
...companies. It's a startup centred on cloud technologies. (Voip, Sip) The name is " Caraïbes VOIP " The main color must be blue like the sky. Je démarre une startup dans la téléphonie sur IP à destinations des entreprises. Le projet est axé sur les nouvelles technologies CLOUD. (Voip, Sip) Le nom du projet est "Ca...
...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
Hi , We have a application which uses Solr 5.3 for our searching. These searches are based on th...our searching. These searches are based on the parameters configured, like Matching %, matching data sets alongwith Phonetic search etc. We want to review the current configuration and would like to tune it. We also would like to use zookeeper. Rosh J
need to make postfix mail-forwarder system and be modified dynamically with new addresses. forward x@[log ind for at se URL] to x@[log ind for at se URL] and y@[log ind for at se URL] to yy@[log ind for at se URL] and so on. each of these addresses can also send emails from the smtp of the server.
Cisco voice gateway connection to SIP server and PSTN on E1 PRI
First of all there is a special attribute, called “ebay_title” You should remove the old cron has been set to generate this attribute Here are the datas I found into cron table exception 'Mage_Core_Exception' with message 'Too late for the schedule.' in /home/compuglobals/public_html/app/[log ind for at se URL] Stack trace: #0 /home/compuglobals/public_html/app/co...
...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
Looking for an expert to configure a cisco voice gateway and call manager for a small office network with 15 users.
Overview: We have four(4) apartment buildings and four(4) Dell PowerConnect 5548 switches. We need to configure these switches so that each unit within each building is isolated from all others. We would like to take advantage of the Private VLANS (PVLANs) feature of these switches, thereby sharing the same IP Subnet for all units. The switches will need to be connected to each other via a "d...
We're looking for a Prestashop expert for some help in configuration of shopping cart / checkout process and some minor bug fixes on our website. Also need advise on upgrading the website from 1.6 to 1.7 OR migration to another platform. If you are able to get things done quickly, pls get in touch.
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
...that are connected to a bridge on a specific tagged VLAN. - I must be able to add another Bridge/VLAN. - We are not willing to use an untagged VLAN/default Virtualizor configuration. I would expect whomever accepts this project has knowledge of Linux Bond interfaces, VLAN interfaces, and Bridge interfaces, and can help me troubleshoot this scenario
...tion] The "[log ind for at se URL]" file could not be downloaded: allow_url_fopen must be enabled in [log ind for at se URL] (https:// wrapper is disabled in the server configuration by allow_url_fopen=0 failed to open stream: no suitable wrapper could be found) update [--prefer-source] [--prefer-dist] [--dry-run] [--dev] [--no-dev] [--lock] [--no-custom-installers]
I’m looking for a logo for business cards and letterhead I’m looking for some flyers so that I can advertise VoIP and SIP services the Name of the company is JT&T Communications we offer best in class SIP and VoIP. And phone system analysis design and implementation