Pstn jobs

We have a PSTN design that is failing one test for China NAL testing. It passes in all other countries. The only failure is Return Loss at 2km, and it is failing by 0.5 dB We use the Silicon labs Si3050 and Si3019 chips and their standard reference design for their DAA We are looking for someone who direct experience using these ICs and the SiLabs reference DAA and passing China NAL.

Hi, I'n new in the collaboration area, so please bare with me. I have CME 11.5 installed on 4331 router, I have configured the auto attendant and it's working perfectly when testing from inside the network, but when I call from outside (PSTN) I hear the welcome message but I can't enter any extension, like when it says press 1 for sales or 0 for help, it doesn't matter what I choose, I keep pressing 1 and 0 but it's not redirecting me, it just keep reading the message and repeating it. remember, this issue is only when call is coming from outside, when I test it from another ip phone it works normally. I'm not sure if it's related to transcoding (that I have no experience about it) show run file is attached thank you

...advanced technology about to be launched. It is important that the logo adheres to the strengths of the product, the solutions it provides, and the image of Media5 Corporation. Media5 - Who are we? Media5, the manufacturer and vendor of the well-known carrier-grade Mediatrix gateways, is a global trustworthy partner supplying VoIP gateways for the Telecom industry focusing on SIP Trunking, PSTN/TDM replacements, Unified Communications, and Hosted Services. Its portfolio of VoIP gateways allows businesses to implement and manage reliable, robust, secure, cost-effective communications while providing the most flexible, feature-rich, diverse, and up-to-date capabilities that can enhance and transform productivity and collaboration. About Virtuo Powered by Media5 Corporatio...

create voip application working on PSTN and voip service, in which we generate virtual number, for example - - -

I currently have two FreePBX instances, PHX-PBX and BDQ-PBX. PHX-PBX is my main PBX, and has most of my configuration (e.g. Extensions, IVR, MeetMe, etc.). On BDQ-PBX, however, I have a Sangoma A200 card and a single PSTN connection on Port 1. What I need to do is the following: 1. For all inbound calls to the PSTN line, I’d like them routed from BDQ-PBX to PHX-PBX to the Main IVR; and 2. For all outbound calls from PHX-PBX that start with 01191, I’d like for those calls to be routed as outbound dials to the PSTN line on BDQ-PBX.

1. Install FreePBX v.13.0.121+ in to Linode server 2. configure SIP trunk engin (Australia) , (can be copy from existing PABX) 3. Connect PSTN to sip using ATA adaptor 4 Connect with Britex 24

Hello, We are looking for someone to help us start the development of an Asterisk Management System. The control panel must be developed in PHP. I am not the best with Asterisk and I am unsure how the control panel would communicate with Asterisk. This is what we need the control to do: - Add a SIP Trunk Channel (For PSTN connectivity) - Remove a SIP Trunk Channel - Add Telephone Numbers to the database which can be used by extensions for outbound and inbound calls - Remove telephone numbers - Create a telephone extension - Delete a telephone extension - Create a multi-level IVR -- By multi-level, I mean the IVR will play a sound, expect key press, then it can play another sound, and expect another key press. The control panel should have all data stored in a MySQL Database, I d...

...customer phone number. Customer may have multiple concurrent deliveries, in that case, text to speech may prompt press 1 for delivery reference XXX, press 2 for delivery reference YYY… If not delivery server would prompt error message. All calls must be recorded. I need to be able to query CDR to check all communication between parties + open recorded communications. Incoming calls will come from PSTN, SIP trunk, and all outgoing call through sip trunk. Script may be based on freeswitch as it’s an xml based solution, faster for administration....

i am working on an app that can provide unlimited calling from Canada to India( same functionality as Rebtel). I looking for someone who can provide pstn gateway solution in Canada with unlimited incoming. Any type of pstn gateway solution can be considered( ex. pri, sip......)

We are building a freepbx server in Canada that can receive inward and outward calls and need some advising setting up a pstn gateway. For example pri line would be better or a sip trunk or if there is any other option ? Looking for a knowledgeable person who can get this project started. p.s. we will be posting some more jobs for the same so will need some help in further development.

use usb modem or an artech AD102 to gain caller id on android device - the modem will be connected via USB I just need that done You have to do your own research and develop I have access to Artech AD102 and a usb voice fax modem to test with which will connect to my android device This will be used in UK pstn line

Hi. I want to setup kazoo from source such as contact center and config to perform outbound, inbound call with PSTN network and softphone. And have some other requirements. Can you do it?

I have a FreePbx which is used for...extensions with simple voip phones. Most of these extensions are forwarded to a cellphone. I use a Grandstream Gateway (GXW4104) for 3 PSTN lines. The Gateway is accessible but for some reason it stopped working for 1 week now. I tried different things but it's not dialing out. I added temporarly a SIP trunk so the systems continues to work. I need to get this Gateway working again. I can provide a Team Viewer access to a computer on the network and which has access to the PBX and the Gateway. You can configure an extension to forward on a number I can provide you for testing purposes (which will have a IVA setup so you know it'S working to dial out). I know the PSTN lines work as I have plugged a analog phone to each one and was...

I have a running pure asterisk sytsem. I need somebody to write an asterisk transfer dialplan. I need to transfer the PSTN call landed in one zoiper extension to another PSTN Number.

Hi I want to modify Linphone into Tablet 7" Andriod 5.1 use for Free SIP to PSTN Kiosk - Auto open app in fullscreen mode (portrait) without close button or switch to other app after device power on - Full screen vdo/jpeg/gif file from URL List - after any touch from above will show mainpage : Separated square layout and pull content from Specific URL (Video, weather, Jpeg, GIF, Google map) and drill down to subpage - Google Maps will use GPS start location in setting page and searchable destination - NOT ACCEPT Incoming call - NO Contact list address book - Dial Page Separate screen as block layout to show (Video, Jpeg, GIF) - Can sent DTMF tone while on call (e.g. press for extension number) - History will sent as

Free PABX error fix and PSTN Gateway setup ones it fixed transfer 2 x Free PABX from digital ocean droplet to Linode Setup maximum security or firewall call recording setup to cloud drive

Kazoo konami transfer via api for mobile extension. _incoming call comes from PSTN and is connected to mobile phone via PSTN -we want to find the channel from kazoo using konami and transfer the call, by sending command via mobile app. WOuld like to have help with konami, in order to transfer the call.

I am using HT503 to automate calls to PSTN. I am not sure what settings to use for this to work in India. I am able to make calls from one sip to another, so that's not an issue. I have also successfully connected FXO of ht503 to asterisk(Registered). This is the output I see everytime I try to make a call: > Event: Hangup > Privilege: call,all > Channel: SIP/amit-0000003a > ChannelState: 6 > ChannelStateDesc: Up > CallerIDNum: amit > CallerIDName: Amit > ConnectedLineNum: <unknown> > ConnectedLineName: <unknown> > Language: en > AccountCode: > Context: phones > Exten: 995XXXX124 > Priority: 2 > Uniqueid: 1522158641.88 > Linkedid: 1522158641.88 > Cause: 127 > Cause-txt: Interworking, unspe...

... 2. configure the API (vonnect the virtual number to exist number) like this: you do it like this: ------------ Pointing number to PSTN --------------- - CREATE TRUNK (), as in documentation provided 'Sample 2', create PSTN forwarding to your defined number. - UPDATE DID (), referring to 'Sample 2' you should assign one step back created PSTN trunk to your DID(-s) 3. to get its stats like this: you do it : ------------ Stats summary / CDRs ------------------ - CDR EXPORT ()

... 2. configure the API (vonnect the virtual number to exist number) like this: you do it like this: ------------ Pointing number to PSTN --------------- - CREATE TRUNK (), as in documentation provided 'Sample 2', create PSTN forwarding to your defined number. - UPDATE DID (), referring to 'Sample 2' you should assign one step back created PSTN trunk to your DID(-s) 3. to get its stats like this: you do it : ------------ Stats summary / CDRs ------------------ - CDR EXPORT ()

Hello, We are looking for someone for long term projects with expertise in freePBX. We have freePBX account with plugins. We want to setup conference calls, we have plugin for that, we need someone to set it up and give manual how to use it. Services that we need are: Setup 3 DID providers Setup freepbx to be able to email PIN to users free PBX...with expertise in freePBX. We have freePBX account with plugins. We want to setup conference calls, we have plugin for that, we need someone to set it up and give manual how to use it. Services that we need are: Setup 3 DID providers Setup freepbx to be able to email PIN to users free PBX to identify caller based on CLI freepbx to connect participant by dialing out ( charging user pstn rate fax-to-email em...

Hi, I have an old PABX telephone system that is playing up. I've bought a second hand system that I would like installed in its place. The business its getting installed in has 3 or 4 pstn lines coming in that are used for calls with line hunt, fax, internet and chubb security. The lines are currently playing up so it might be the pabx or some of the joins which you will need to problem solve as well. Telstra say the problem is not external to the building. The system has one main reception phone and will have 6 extension phones. All cabling is in place to the extensions.

Hi, I have an old PABX telephone system that is playing up. I've bought a second hand system that I would like installed in its place. The business its getting installed in has 3 or 4 pstn lines coming in that are used for calls with line hunt, fax, internet and chubb security. The lines are currently playing up so it might be the pabx or some of the joins which you will need to problem solve as well. Telstra say the problem is not external to the building. The system has one main reception phone and will have 6 extension phones. All cabling is in place to the extensions.

Chat App with PSTN/ GSM calling capabilities To be able to - Call PSTN lines - Call in-app (peer 2 peer) - Voice Notes - Instant Chats Media SDK IOS/ Android/Web Media SDK, integrated via griddle to android app and via Cocoa Pods to iOS app to enable communication between mobile applications or web for desktop to PSTN. Integration of the media SDK will be done by the customer or contracted professional services.

Feature Description: Media SDK IOS/ Andro...IOS/ Android/Web media SDK, integrated via gradle to android app and via Cocoa Pods to iOS app to enable communication between mobile applications or web for desktop to PSTN. Integration of the media SDK will be done by the customer or contracted as professional services. media platform (hosted) media platform that enables users to communicate securely using VoIP on mobile or web. additional features can be such as recording, conferencing, network error handling, security and more… Summary of the App: 1. Voice Calls to any GSM phone via dial pad 2. VoIP is the critical factor (The Call must not be completed on PSTN lines only) 3. Make it run Voice Conferencing much like Skype without the video 4. Loading it to...

...Linphone. - Delete chat feature - Delete assistant feature - No account creation form (this is a very private app, we will provide access to user ourselve by email) - SMS Feature (we will use design of chat feature but for sms) : Our server already manage this feature, so app need to pass to our server url text message + number phone to send to). - Call feature : no SIP call we will call only PSTN Number : again here you will need to send to our server - the number user want to call - login/password account of the user (this login/password is already stored in the app because user need to set it up to use the app) - finally a gateway ID (user will choose to initiate the call by choosing a given gateway id. Ids of gateway will be hard coded in the app, we will make it dy...

Hi , I am looking for VOIP switch developers for calls routing on apps and PSTN

...repairing, resolving, and documenting end user technical issues for basic desktop/laptop/workstation support, basic connectivity support (wired and wireless), PDAs, BlackBerrys, Smartphones and basic printer support - Support users with Apple Mac and IOS devices - Support Multifunctional Devices (MFD) for issues like Scan to Email, Scan to Fax, Email to Fax etc. - Support In-country PBX networks and PSTN interfaces - Support Hardware/Software selection and Procurement effort - Support Hardware Refresh, Redeployment and Disposal activities - Troubleshooting and resolving software issues. Ability to install, configure, reconfigure or reinstall software including remote support - Reimaging computers/hard drives in accordance with customer standards - IMACD function includin...

we have SippySwitch as main SoftSwitch. we plan to put OpenSIPS front of it as to support Tls and media encrypted. APP-->OpenSips-->SippySwitch our APP talk with OpenSips via Sip TLS. OpenSips talk ...plan to put OpenSIPS front of it as to support Tls and media encrypted. APP-->OpenSips-->SippySwitch our APP talk with OpenSips via Sip TLS. OpenSips talk with Sippy vai normal UDP sip message. APP-->media relay-->Provider our APP talk with media relay via encrypted RTP, media relay server talk with provider via normal RTP we need these works. APP make outgoing call to PSTN APP make outgoing call to APP OnNet APP/DID make incoming call to APP both incoming and outgoing need to through OpenSIPS using TLS/ media encrypted it works as a g...

Our small company is migrating to Office 365 E5 Cloud PBX and we've been trying to setup the system outselves. We've run into an issue that Microsoft Support cannot seem to fix (because they don't know anything about the Cloud PBX, Phone System, or Unified Messaging)....does not recognize our dial plans. We do not know how to properly set up the dial plans and it's causing trouble with our ability to access our voicemail. Our problem is mostly centered around extension dialing. This should be a pretty simple job for someone who knows how to administer Office 365 E5 Cloud PBX. We do not have an on-premise IP PBX. We are only using the Office 365 E5 Cloud PBX w/ PSTN calling plans. We are serious about moving forward with this project. We'd like to hir...

...repairing, resolving, and documenting end user technical issues for basic desktop/laptop/workstation support, basic connectivity support (wired and wireless), PDAs, BlackBerrys, Smartphones and basic printer support - Support users with Apple Mac and IOS devices - Support Multifunctional Devices (MFD) for issues like Scan to Email, Scan to Fax, Email to Fax etc. - Support In-country PBX networks and PSTN interfaces - Support Hardware/Software selection and Procurement effort - Support Hardware Refresh, Redeployment and Disposal activities - Troubleshooting and resolving software issues. Ability to install, configure, reconfigure or reinstall software including remote support - Reimaging computers/hard drives in accordance with customer standards - IMACD function includin...

Hi, We have 3cx VoIP installation on the main site, I want to implement PBX for other sub sites preferably Grandstream 61xx UCMs. My goals are: - Configure bridging in 3cx to be able work with grandstream - Users from main site should be able to call directly the extension number of the Grandstream site and vice versa. All sites are connected private VPN - Configure fail over PSTN Telco line in the Grandstream site. I would prefer an expert on this to work remotely and with competitive price. :) Thank you.

We would like a quote on an Upgrade for an existing Chat App that is already in the market. The app is very much like Whatsapp. The main features to add includes: 1. Voice Calls to any GSM phone via dial pad 2. VoIP is the critical factor (The Call must not be completed on any PSTN lines) 3. Make it run Voice & Video Conferencing much like Skype 4. Loading it to Play Store & iStore The Current App features already include: + Instant messaging Chat + File Sharing + Voice & Video Calls (***In-App only) Or you can download the Giant Call App

I'm integrating a2billing to my asterisk platform, this system will require me to bill the recipient or the callee for receiving calls. Below is the detail scenario of my application: ...a2billing platform. 3. UserB setup a DID on the a2billing platform with 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 for both A-Leg and B-Leg of the calls. But what I want is for only UserB, the owner of the DID to be billed for any calls to his DID that terminates on the ...

...from website Configure auto-dialing groups in odoo using contact, leads, or any other editable/custom group. (with option to dial a direct number or extension in IVR) Add calling, sms, multimedia and voice broadcasts 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 and voice notes via soft-phone Configure asterisk/odoo to connect to a translation service, internal or external, and route voice traffic through the translator Will need some way to identify/specify the language of each endpoint Create extensions and supporting objects (times, applications, etc) for the below functionality. The...

...from website Configure auto-dialing groups in odoo using contact, leads, or any other editable/custom group. (with option to dial a direct number or extension in IVR) Add calling, sms, multimedia and voice broadcasts 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 and voice notes via soft-phone Configure asterisk/odoo to connect to a translation service, internal or external, and route voice traffic through the translator Will need some way to identify/specify the language of each endpoint Create extensions and supporting objects (times, applications, etc) for the below functionality. The...

...can use a month etc) Companies Admins need to be able to Diagnose faxes from the GUI Companies Customers should be able to Send, see incoming, Delete faxes, Set Up a Cover Page Companies Customers should have the option 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 to be able to handle which carriers the STS should use for Outgoing Faxes (sometimes we may want to use a different carrier to send out the faxes, for example either the T1 or Peerless etc) and it should not be the...

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 85.10.244.150. You will need to ensure to configure your system to accept calls with cps prefix: Request-URI User Part: C00344442070397187 as we route CPS calls to you as: sip:C00344442070397187@ We need to configure our FusionPBX to allow calls sent to it from our provider then get routed correctly out of the PBX via the SIP trunks to get the benefit of the cheaper calling costs. This should be a very simple config change

Looking for Grandstream expert/ VoIP engineer who can fix call 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. It's ringing twice at caller end(PC 2 Dialer) and on 3rd ring showing status call connected leg A billing is starting(onPC 2 Dialer). But on other side leg B side is still ringing and no billing starting at leg B(on Grandstream Gxw4108) . We want to solve this issue, Support for Grandstream Gxw4108 settings VoIP to Grandstream GXW4108 is required.

Looking for Grandstream expert/ VoIP engineer who can fix call 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. It's ringing twice at caller end(PC 2 Dialer) and on 3rd ring showing status call connected leg A billing is starting(onPC 2 Dialer). But on other side leg B side is still ringing and no billing starting at leg B(on Grandstream Gxw4108) . We want to solve this issue, Support for Grandstream Gxw4108 settings VoIP to Grandstream GXW4108 is required.

Hi bro, I want to configure, remote VOIP over VPN, I have on my local network "MyPBX" router, WRT router (OpenVPN client), I also have a VPS Centos7 (OpenVPN Server), I want 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,

...module to route through asterisk configure asterisk to accept and route webrtc Configure auto-dialing groups in odoo using contact, leads, or any other editable/custom group. Add calling, sms, multimedia and voice broadcasts 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 the translator Will need some way to identify/specify the language of each endpoint Create extensions and supporting objects (times, applications, etc) for the below functionality. The basic hierarchical stru...

Hello, I have an FXO gw SPA400 behind a firewall /home router. It has a local IP : 192.168.1.x. The asterisk server has a public IP address. I have forwarded UDP ports 5060 and 10000-20000 in the 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 considered as successful when both issues are resolved. Thank you.

I need some help with finding some leads. Swiping Mechines Types:- mpos Gprs Land line Any Mechine just 2500/- 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)

We need to install the latest stable release of OPENSIPS and ASTPP in the server and then establish calls between users using SIPML5. Calls should be established between - Chrome to Chrome browsers - Chrome to Android devices having the a...need to install the latest stable release of OPENSIPS and ASTPP in the server and then establish calls between users using SIPML5. Calls should be established between - Chrome to Chrome browsers - Chrome to Android 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...

I am in a country they block SIP and VoIP, I already have myPBX server 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.

...part-time business, but I'm very new to VOIP systems like FreePBX/3CX/etc (and trying to configure by trial and error is very difficult and frustrating). I currently have 3CX installed on a small Vultr instance, and I've tried integrating Twilio (and previously ) 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 some new Yealinks that are arriving on Monday). Ideally, the inbound calls would be answered automatically by auto-attendant with music on hold while all of my phones ring (then roll over to...

I will like to use either of the asterisk versions specified in my title above to get a conference chat room 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 all channels in the chat room in real time to either remove or mute any channel for whatever reason including a noisy channel. d. Unlimited Participants expected Regards

...trunk, all numbers that are longer than 3 digits (i.e. that are not internal extensions) but don't match the international format should be rejected as invalid with a voice message (e.g. "the number you dialed is incorrect"). 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 ext. 200 at Asterisk (i.e. +1 905 878 5001 coming from external SIP trunk → FS → Asterisk → ext. 200). Endpoints connected to Asterisk should be able to make outside calls too via ...

Looking for an experienced Linux Administrator with Asterisk Voip experienced. Min. 5 years hands on experience. 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