Opensips opensipsJobs
We are a telecom startup and we’re looking for a PHP full-stack web developers with experience in Twilio and having knowledge of telecom and how telecom systems work especially FreeSWITCH & OpenSIPS. You must have the following skills to qualify for the job: PHP Laravel Javascript/JQuery SSH + Ubuntu Command line MySQL GIT - GitHub Twilio Telecom knowledge FreeSWITCH OpenSIPs AWS These skills are optional: Facebook APIs React We will offer long term work every week but only if you prove your skills in the first week or so. We are only looking for serious developers and to prove that please fill the attached document (we uploaded the same document in 3 file types, use the one you prefer), upload it anywhere and add the link to the project proposal. Your p...
We have opensips running on our cento7 server, we need someone who can install cgrates & cgadmin and integrate it with opensips for real time billing. your bidding price will be deemed as final price.
Hi, We are looking for a consultant to assist in setting up a Kamailio SBC to integrate Teams & Asterisk. Brief overview of requirements: - Centos 8 - Kamailio - Support for multiple MS Teams instances - Support for multiple Asterisk servers - Certain MS Teams instances need to be linked to specific Asterisk servers Required: previous experience integrating Kamailio/opensips with Teams This job is urgent. We need quotes today to be signed off tomorrow (Thusday) and installed Friday. Thanks, Carel
Hi, I am looking for a softswitch class4/5 carrier grade, based on freeswitch, opensips or kamailio, with GUI and billing.
Hello, 1. I have an OpenSIPS configuration. 2. When I get connect OpenSIPS with PostgreSQL that time accrued error in the system: 3. OpenSIPS version: 3.1 Master. 4. PostgreSQL version: 11. 5. Someone who can resolve this OpenSIPS with this error. Thanks.
I am looking for an EXPERT with opensips to build me a carrier grade platform with all the bells and whistles .I also need to an online platform where wholesale and retail customers can purchase DiD and have it automatically provisioned so they can use it on their mobile apps to send and receive calls based on prices on out rate deck. Wholesale customers seeking termination services must be able prepay and send us traffic for termination.. EXPERTS ONLY.
Ola Felipe! Que tal? I would like to finish a job with OpenSIPS (SBC between Asterisk and MS TEAMS). Do you know OpenSIPS? Are you interested?
Hi Aqs Y., I would like to discuss if you could help us. I am seting up an OpenSIPs as a SBC between TEAMS and Freepbx. We are in the middle of the job. Could we hire you? Thanks! Denilson
I am looking for an EXPERT with opensips to build me a carrier grade platform with all the bells and whistles .I also need to an online platform where wholesale and retail customers can purchase DiD and have it automatically provisioned so they can use it on their mobile apps to send and receive calls based on prices on out rate deck. Wholesale customers seeking termination services must be able prepay and send us traffic for termination.. EXPERTS ONLY. I am also thinking of using a2billing for billing.
Hi, I need to get Class 4 switching setup and am taking bids. We currently use FreeSWITCH & OpenSIPs with a cusotm BPX designed by a telecom engineer. I need someone to tell me the best open source software, why it's the best and how long it will take then to do the complete instalation.
We require someone experienced in VOIP, specifically OpenSIPS, MediaProxy, Kamailio, Billing, GUI and Freeswitch to aid in the implementation of a class 5 switch solution and class 4. Proposed design; * Two OpenSIPS proxies - All customer termination and origination from / to customer via proxies * Two MediaProxy media servers - All customer RTP is between MediaProxy and Freeswitch * Two Kamailio SBC's - Acting as PSTN gateways between OpenSIPS and carriers / TDM gateways * OpenSIPS Location / Redirect Server * FreeRadius Server - AAA * Two PBX FreeSwitch * GUI WEB: - Multi Tenant, - Multi Language (Portuguese Brazil, English and Spanish) - Resseles, - Home - Menu - Live System Reports - Rates Management - Routing Managemnt - User Manage...
I am looking for an EXPERT with opensips to build me a carrier grade platform with all the bells and whistles .I also need to an online platform where wholesale and retail customers can purchase DiD and have it automatically provisioned so they can use it on their mobile apps to send and receive calls based on prices on out rate deck. Wholesale customers seeking termination services must be able prepay and send us traffic for termination.. EXPERTS ONLY.
I am looking for an EXPERT with opensips to build me a carrier grade platform with all the bells and whistles .I also need to an online platform where wholesale and retail customers can purchase DiD and have it automatically provisioned so they can use it on their mobile apps to send and receive calls based on prices on out rate deck. Wholesale customers seeking termination services must be able prepay and send us traffic for termination.. EXPERTS ONLY. VOIP opensips CGrates, webrtc asterisk
Hi, We are stuck and need urgent help in getting OpenSIPs 3.0.2 working with TLS/WS/WSS listeners. Version: opensips 3.0.2 (x86_64/linux) It's getting stuck on OpenSIPs startup when using LetsEncrypt certificate. Thank you! :)
...existing customers in beta and ready to launch soon) and we’re looking for a PHP full stack web developers with experience in twilio and having some knowledge of telecom, not 100% required but would also be great if you have knowledge with FreeSWITCH & OpenSIPs (both open-source telecom software & PBX / SIP Server). You must have the following skills to qualify for the job: PHP Laravel Javascript/JQuery SSH + Ubuntu Command line Ubuntu/Linux/SSH MySQL GIT Twilio These skills are optional: AWS Facebook APIs React Telecom knowledge FreeSWITCH OpenSIPs We will offer long term work every week but only if you prove your skills in the first week. We are only looking for serious developers and to prove that please fill the attached document, upload it anywhe...
I need integration of opensip over asterisk. NOTE :- DONOT BID UNTIL YOU DONOT KNOW ABOUT OPENSIPS
I have installed openSIPS server but need to configure it so that it does the necessary basic functions of SIP server with modules such as the SMPP for sms activated ( & ).
Chào bạn, Tôi tên là Tiến, ở tại Sài Gòn; cần tích hợp OpenSIPs và FusionPBX (FreeSWITCH). Regards,
I have installed openSIPS on my server to the best of my knowledge but have problem concerning modules tables not showing in database. I do not know enough to solve that.
i just need openSIPS installed on my Debian 9 server whereas I can use via web GUI
I want to learn opensip integration in asterisk . I already install asterisk server . Thanks in Advance
I want to install opensip in asterisk... Note :- Donot waste your bid until you donot know about opensips
I need to configure Kamailo SBC to connect multiple Microsoft Teams account in the same SBC server. - Install Opensips - Install Opensips-cli - Install Opensips-cp - Configuration TLS certificates - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk
Want to setup the CGRates billing module with the Opensips with the AIP. both are already installed in machine now have to configure the CGRates billing module. so, I can be able to charge properly. Type first word in your proposal that "WOLE_CG". So, I can understand you are not bot. Ref :- Budget will be changed ** Thanks
Hi, I have an Asterisk server running a small telecom operation. I reached a number of users that is beginning to affect Asterisk, so I need a more versatile SIP router. I would like to maintain asterisk to implement IVR logic, voicemail, conferences but I would like to pass functions like SIP contact management, authentication (Invite and Registration) and even subscriptions to either opensips or kamailio.
i need With modules callcenter dialog CDR load balance NAT will Work With opensips-cp
We are looking for a virtual phone solution to be accessed with iPhone that can have an auto attendant (option 1: Sales and 2: Support). I need to build our own solution, especially we don't receive many calls. I expect that the solution would be in the cloud. The system wil...in the cloud. The system will have a US number. The developer may utilize technologies and system components like soft PBX, SIP, Asterisk, DID, etc. I need a stable system that would have a cheap running cost. Please give me a plan of what you will do and a fixed rate. Example of optional technologies: Asterisk, FreeSwitch, Kazoo VoIP Cluster, Kamailio, callweaver (faxing), ASTPP, Auto Dialer, opensips, kamailio, Call Center, Elastix, Vicidial, VoIP, Ringless VoiceMail, 3CX, FusionPBX, Issabel, OSDial, G...
I have flexisip , Asterisk , Opensips working well . I want experienced people flexisip , Asterisk , Opensips Who can support configuration document. Client <--->Flexisip ( Frontend )<---> Asterisk ( Backend )<---->PSTN Please check this link and If You can configure and support contack to me.
i need With modules callcenter dialog CDR load balance NAT will Work With opensips-cp
I have flexisip and Asterisk , Opensips working well. I want experienced people Who can support configuration document. Flexisip () Frontend Asterisk , Opensips Backend
I want to be able to respond with PRACK with an opensips proxy.
[OpenSIPS/opensips] Can opensips generate update messages for refreshing session timers? (#1274)
We have database server on aws, our special script check every caller IDs which comes in and goes out with some APIs to identify if caller is a spammer or not. Opensips serves as a proxy for IP authentication and SIP registration and once Opensips receive the calls and match everything as per configuration it sends call to freeswitch for further processing. Now, in database we have table which stores all the blocked caller IDs, now what we need to do is create a statement in file that if call comes in call should first get checked in block list if caller ID is present in block list then route the call to specific freeswitch and if caller ID is not matched in database then call should processed normally. * Please go through description carefully i will not negotiate and pay...
We need someone to assist us remotely for deploying Docker Enterprise in both testing and production environments. Once the infrastructure high availability deployment for Docker/Kubernetes (3+3 nodes in both testing/Production for UCP and DTR) will be achieved, we should then focus on the customization of our VoIP application (contact center and pre-paid solution) using either OpenSIPS and/or ASTPP software and its integration to the Twilio network.
I have opensips installed with 4 Freeswitch servers, IP authentication is working fine but in SIP authentication (i.e., Register) i need to append following code. # Setting accountcode in P-Accountcode variable remove_hf("P-Accountcode"); remove_hf("P-Pricelist_id"); cache_fetch("local","accountcode_$fU",$avp(accountcode)); cache_fetch("local","pricelist_id_$fU",$avp(pricelist_id)); if(!$avp(accountcode) && $fd==$od) { avp_db_query("select accountcode,pricelist_id from subscriber where username='$fU'","$avp(accountcode);$avp(pricelist_id)"); cache_store("local","accountcode_$fU","$avp(accountcode)",3600); ...
I am looking for a detailed instruction for the installation of Opensips, Opensips-cp, freeswitch, homer, cdr/billing(eg cgrates with gui), all in a single machine (real or VM). The host system would be Debian 10 64bit. The documentation should be taken from a novice engineer and get reproduced in a step by step manner. Everything would be in LAN and the IP would be with gateway/dns 192.168.178.1. Configuration of the users should be described how to get started. Describe how IVR, Voicemail, faxing is enabled. Description of how to create cost plans(pre-paid and post paid) for different extensions and how to extract CDRs and billing info for each extension. More jobs to come!
We are implementing a VIOP system. We need a consultant in OpenSips or similar
Hello, I need opensips 2.4 to be installed with control panel integrate to ASTPP 4.01 billing system using PosgreSQL and setup PSTN service on ASTPP.
Looking for someone with experience in OpenSIPs. We need a basic SIP <-> SIP-I gateway built in OpenSIPs (or similar). It's important to have a config section were we can manipulate ISUP configs. We need someone who has experience in Carrier Interconnection using SIP-I and had done this job before.
I want configure a OpenSips or Kamailio SBC for load balance with mu freeswitch servers in a NAT (Amazon AWS). The users register can be use a mysql database.
- Download and install Opensips; - Create template configuration for Customer (originator) and Vendor (destination); - Customize origination peer to analyze incoming destination number and identify destination number area code; - Based on identified area code, search randomly for a number in a specific file for a number to replace at FROM field (caller ID); - Send call to destination;
VOIP Reseller Setup: Able to do installation and configuration Asterisk, A2billing, and openSIPs on my server.
Hi I need to install Opensips. It have to Register clients and make load-balancing by IPs and Web-interface to manage will be located behind NAT. so it's not so easy to do.
We need and expert with Opensips in order to configure and connect it to Asterisk PBX to configure Opensips for handling SIP messaging to SMS provider and vice versa. a complete Opensips solution to send/receive sms from peer-to-peer SMS provider. Encoding and splitting the messages by Opensips with following guideline: All software must auto start on server boot We already installed Opensips 3.0 on Centos 7.6 (FreePBX 14 and asterisk 16.3.0) Provide full list of commands to install the above and files edited to be able to install on another server at a later date Give us solution for backup and restore all config files to our backup server. The project is to connect our Opensips server to a peer-to-peer SMS provider by smpp The following gu...
We want below thing to setup in Opensips : When User A is offline and User B calls to A, then Opensips should send Push notification to User A. (Opensip Side) User A will send SIP REGISTER to Opensips. (App side) Opensips will receive the SIP Register success event and send SIP INVITE to User A. (Opensip Side) User A will ring and answer the call. (App Side) In Above flow, App side is implemented. We need to add route in Opensips for above want below thing to setup in Opensips : ,
I need someone who can enable/configure incoming calls load balancing on OpenSips among 4 Asterisk instances, attached is my current opensip conf.
We want below thing to setup in Opensips : When User A is offline and User B calls to A, then Opensips should send Push notification to User A. (Opensip Side) User A will send SIP REGISTER to Opensips. (App side) Opensips will receive the SIP Register success event and send SIP INVITE to User A. (Opensip Side) User A will ring and answer the call. (App Side) In Above flow, App side is implemented. We need to add route in Opensips for above want below thing to setup in Opensips : ,
We need an expert in OpenSIPS that can change the SIP response of 200 ok to something else or delay it full details will be given later.
