Asterisk sip client application open source jobs

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15,007 asterisk sip client application open source jobs found, pricing in USD

...experience is required Requirements for the role: Fast and stable internet connection (For receiving voice calls) Computer with web browser for completing online scoring SIP Softphone (such as xLite or Talk, xLite is recommended) Working hours are 8am - 9pm UK time, Monday to Friday. We expect each operator to do a minimum of 4 hours per day.

$5 / hr (Avg Bid)
$5 / hr Avg Bid
19 bids

Im looking for someone to modify an open source app (linphone) to connect to our servers via API. We have our own phone servers, so we would like for the app to use our API for user login and once logged in, pull the user SIP device via API. The call log and voice mail will also pull from our API. Being the entire app is already built, it should be

$205 (Avg Bid)
$205 Avg Bid
29 bids

Hi I'm looking for people with knowledge of Asterisk, [url removed, login to view], communication software (or similar solutions for call center management). The goal is to build software which would allow web based video conversations and basic call center management. More details via pm.

$3736 (Avg Bid)
$3736 Avg Bid
16 bids
Asterisk Expert 5 days left
VERIFIED

We are in need of an Asterisk Expert to Implement the designs we already have into Vicidial. This can be an on-going project for right candidate. Happy bidding !

$163 (Avg Bid)
$163 Avg Bid
10 bids

Configure Polycom Soundpoint ip601 SIP phones with SIPCITY cloud PBX. As above. We have three phones to be configured. This should be quick and easy for the right person.

$35 (Avg Bid)
$35 Avg Bid
2 bids
Opensip/cgrates 8 days left

Hello, I need to have an opensips server installed with cgrates. This will be for SIP trunking with nat support. I will need to be able to route inbound DIDs to customers. Also will need to be able to setup accounts in cgrates so that we can have about prepaid calling on the account. I think we have several other project that you might be able

$500 (Avg Bid)
$500 Avg Bid
1 bids

Linux Otra o no estoy seguro Desarrollo informes sobre Asterisk

$884 (Avg Bid)
$884 Avg Bid
10 bids

...andriod voip client done using linphone. the job was not 100% completed. I need to: 1- fix / change the logo and 2- turn off access to settings , 3- turn off the sip address when dialing a number, and 4- make the app recognize only the last 5 digits of the phone number. 5- Publish to google play. 6- give me the final source code. 1- You

$63 (Avg Bid)
$63 Avg Bid
12 bids
Expert in WEB-RTC -- 2 5 days left
VERIFIED

Experto en WEBRTC Tenemos un servidor asterisk y un servidor web queremos hacerlo correr en HTML5 con un cliente que tenemos hecho. You have to be a Expert in WEB-RTC we need to connect an app to asterisk via sip we already have a client and server we need to get alive.

$6 / hr (Avg Bid)
$6 / hr Avg Bid
8 bids

Necesito crear un softphone que trabaje bajo protocolo sip para telefonia ip, con un proxy predefinido fijo, donde el cliente baje el softphone ojala de google play, y solo ingrese su usser (la pass sera la misma que el usser) y el proxy siempre sera el mismo, tenga la opcion de registrar la cuenta, y poder llamar.

$232 (Avg Bid)
$232 Avg Bid
17 bids

Want to allow my shopify cart to be able to search partial search (add wildcard asterisk to front and back of search word? or something better). Autocomplete, and instant search (starts showing some results of items/pictures when searching. I know this code is out there, just want it on my site.

$175 (Avg Bid)
$175 Avg Bid
15 bids
Expert in WEB-RTC 3 days left
VERIFIED

Experto en WEBRTC Tenemos un servidor asterisk y un servidor web queremos hacerlo correr en HTML5 con un cliente que tenemos hecho. You have to be a Expert in WEB-RTC we need to connect an app to asterisk via sip we already have a client and server we need to get alive.

$2 - $8 / hr
$2 - $8 / hr
0 bids

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Need to build Calling Card VoIP server based on Centos 6, Asterisk and A2Billing

$249 (Avg Bid)
$249 Avg Bid
6 bids

Existing jsSIP dialer needs guru troubleshooter to help resolve bug. Upon a successful connected call from web app through fpbx to sip trunk using webrtc jsSIP. We have a lag upon connection where no sound between caller and called. The lag/no sound lasts 5 seconds usually then both parties on call can hear each other. We require all work be

$204 (Avg Bid)
$204 Avg Bid
10 bids

...with FPBX to SIP trunk working. Just small delay on some #'s, not all calls, when connects until each person can hear each other. Work MUST be done via AnyDesk or TeamViewer. Here is where the delay occurs in CLI: (the lag is between these two lines) Sequence: 1. Line Posts in CLI: 0xb7503018 -- Probation passed - setting RTP source address to

$181 (Avg Bid)
$181 Avg Bid
8 bids

...install FreeSwitch on my server and install ASTTP on that. then doing this configs: 1- Activate G729 Codec on that 2- Change transfer protocol to UDP 3- activate just SIP protocol on server. (disable another protocols) 4- Enable Video Call feature on freeswich if you not robot please add "Swh" in your bid. Thanks for your attention

$206 (Avg Bid)
$206 Avg Bid
9 bids

I am using Express Talk VoIP Softphone in my Windows PC (WIN7 OS)But the Sip is not connecting properly all time Need Guidence to connect quicky all time to sip account

$5 / hr (Avg Bid)
$5 / hr Avg Bid
10 bids

...custom andriod voip client done using linphone. the job was not 100% completed. I need to: 1- fix the logo and 2- turn off access to settings , 3- turn off the sip address when dialing a number, and 4- make the app recognize only the last 5 digits of the phone number. 5- Publish to google play. I only have the app no source code. You can download

$94 (Avg Bid)
$94 Avg Bid
15 bids

...VG224 en SIP contra un Server Asterisk. Actualmente lo tenemos configurado pero con algunos problemas como que solo algunos puertos se registran y otro no. /////////////////////////////////////////////////////////////////////////////// We are looking for someone who has the ability to configure a CISCO VG224 gateway using SIP with an Asterisk server

$40 (Avg Bid)
$40 Avg Bid
6 bids