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    548 pstn jobs found, pricing in USD

    ...his GSM mobile number as the destination for the DID. 4. UserA calls the userB DID and the call terminates on UserB GSM mobile line via a SIP trunk to PSTN. 5. UserB is billed for the DID to PSTN call. 6. UserA is not billed for anything. The above is what I want to achieve. I've been able to do the setup, but right now, only UserA is being billed

    $23 (Avg Bid)
    $23 Avg Bid
    1 bids

    ...odoo follow up options Optimize server configuration for reliable high audio and video quality configure settings for sip, users, codecs, etc for best quality configure PSTN connection(s) for best quality Enable chat and voice notes via soft-phone Configure asterisk/odoo to connect to a translation service, internal or external, and route

    $656 (Avg Bid)
    $656 Avg Bid
    13 bids

    ...odoo follow up options Optimize server configuration for reliable high audio and video quality configure settings for sip, users, codecs, etc for best quality configure PSTN connection(s) for best quality Enable chat and voice notes via soft-phone Configure asterisk/odoo to connect to a translation service, internal or external, and route

    $468 (Avg Bid)
    $468 Avg Bid
    7 bids
    Build a Fax Server Ended
    VERIFIED

    ...to re-send a sent fax. Need to be able to Manage Notifications that a user should receive on successful faxes, on failed faxes, when ATA gets disconnected, etc Integrate PSTN/T1/ Fax Carrier with STS Need to discuss if we should add Standalone Asterisk Server with T1 Card for better reliability / or use Adtran to send calls from T1 to STS Need

    $2666 (Avg Bid)
    $2666 Avg Bid
    2 bids

    We have a FusionPBX Cluster which is working perfectly for our standard clients. We have now established a relationship with a provider that allows PSTN calls to be routed to our SIP server. Our provider has confirmed they can see the following: I can see these calls however we are getting a 404 not found back from your server [url removed, login to view]

    $31 / hr (Avg Bid)
    $31 / hr Avg Bid
    1 bids

    ...connectivity billing problem. The details are as follow we have Grandstream GXW4108. It's FXO Gateway. We want to use this gateway as PSTN gateway it means VoIP call will enter in Grandstream GXW4108 and will hit PSTN FXO lines for call out. We have made all settings. But we have 1 problem. When caller is calling from VoIP to Grandstream GXW4108.

    $166 (Avg Bid)
    $166 Avg Bid
    1 bids

    ...connectivity billing problem. The details are as follow we have Grandstream GXW4108. It's FXO Gateway. We want to use this gateway as PSTN gateway it means VoIP call will enter in Grandstream GXW4108 and will hit PSTN FXO lines for call out. We have made all settings. But we have 1 problem. When caller is calling from VoIP to Grandstream GXW4108

    $26 (Avg Bid)
    $26 Avg Bid
    4 bids

    ...people from the internet to make calls using my PSTN lines throw "MyPBX" router, I want to implement two things: 1- link remote branch VoIP phone to call extensions in HQ and to use local PSTN in HQ, 2- any one can use sip softphone to link to the HQ PBX, and to call extensions in HQ and to use local PSTN in HQ. can you do it, how long, how much

    $221 (Avg Bid)
    $221 Avg Bid
    4 bids

    I am looking for an expert on setting up VoIP PBX as per the following condition:- 1. Install configure and set A2Billing software on ...Ensure all traffic goes via SSL TLS. 4. Test the configuration by connecting hard and soft SIP phone. Both of them will be provided. 5. Configure VoIP trunk for making outbound PSTN calls, preferably google voice.

    $196 (Avg Bid)
    $196 Avg Bid
    21 bids
    Linux Ended

    ...to odoo follow up options Optimize server configuration for reliable high audio and video quality configure settings for sip, users, codecs, etc for best quality configure PSTN connection(s) for best quality Enable chat via soft-phone Configure asterisk/odoo to connect to a translation service, internal or external, and route voice traffic through

    $30 - $250
    $30 - $250
    0 bids

    ...router to SPA400 IP address. The SPA400 can register to asterisk server. The problem is that: 1) huge delay (more than 20 sec) for outgoing PSTN calls. There is no big delay for incoming calls. 2) PSTN called/caller are unable to hear anything apart from the ring Please apply only if you have worked on similar setup before. The project will be

    $158 (Avg Bid)
    $158 Avg Bid
    7 bids

    I need some help with finding some leads. Swiping Mechines Types:...processing fee Rent:- mpos:- 225/- p.m Gprs:- 700/-p.m Land line(pstn)300/- If any want it for lifetime free Mpos:-8500/-(entire life) Gprs:-21500/-(entire life) Pstn:-11800/-(entire life)

    $9 - $23
    $9 - $23
    0 bids

    We need to install th...devices having the app installed and vice-versa - Chrome to IOS devices having the app installed and vice-versa - also includes a Chat function and the ability to call the PSTN For video calling purpose we need the help of webrtc. The video call, voice call, voice chat should work seamlessly in both 2G and 3G networks.

    $495 (Avg Bid)
    $495 Avg Bid
    10 bids

    I am in a country they block SIP and VoIP, I already have myPBX server in HQ, I want to implement two th...in HQ, I want to implement two things: 1- link remote branch VoIP phone to call extensions in HQ and to use local PSTN in HQ, 2- any one can use sip softphone to link to the HQ PBX, and to call extensions in HQ and to use local PSTN in HQ.

    $126 (Avg Bid)
    $126 Avg Bid
    9 bids

    ...integrating Twilio (and previously [url removed, login to view]) as the outgoing provider. Receiving calls from a PSTN works fine, but outgoing calls always fail (i.e., "forbidden/not found/etc"). I "can" dial internal extensions without any problems, but the outgoing calls to a PSTN fail every time. I plan on having several extensions (both softphones and s...

    $75 (Avg Bid)
    $75 Avg Bid
    1 bids

    ...working with the following features; a. The Chat room must have two (local) extensions. One for the "Internal Participant" and another for the Admin. b. Persons (from PSTN) who want to join the conference will call the Admin extension and he will add the caller to the chat room. Only Admin can add people to chat room. c. Is there a way to monitor

    $155 (Avg Bid)
    $155 Avg Bid
    17 bids

    ... The FS extension 101 should NOT be able to call outside (only internal calls to other extensions). For incoming calls from the external SIP trunk, there are 2 external (PSTN) numbers (e.g. +1 905 878 5000 and +1 905 878 5001) - when receiving a call for the first number, it should be redirected to the ext. 100 at FS, for the second number to the

    $256 (Avg Bid)
    $256 Avg Bid
    7 bids

    Looking for an experienced Linux Administrator with Asterisk Voi...Mandatory skills: Centos7,6 Asterisk Optional Skills Expeience with 5 Softswitch Asterisk, Freeswitch, Kamailio,Opensips,vicidial,A2billing, Free pbx, elastix, PRI & PSTN card installation, call centre setup, Advanced IVR, Vmwire Esx, web-meetme ,Cloud solution using asterisk

    $12 / hr (Avg Bid)
    $12 / hr Avg Bid
    34 bids

    ...endpoint at extension 101 should NOT be able to call outside (only internal calls to other extensions). For incoming calls from the external SIP trunk, there are 2 external (PSTN) numbers (e.g. +1 905 878 5000 and +1 905 878 5001) - when receiving a call for the first number, it should be redirected to the 100 extension, for the second number to the

    $229 (Avg Bid)
    $229 Avg Bid
    9 bids

    ...Ideally I would like to activate that same-DID-line on another phone (pickup handset) to become party to the conversation without the recipient initiating call transfer, like pstn phones of old. I also want the intercom feature to function. The voicemail available notification must continue to function. I want our samsung smart phones (LibrePhone

    $225 (Avg Bid)
    $225 Avg Bid
    9 bids