Iax2 fonality jobs

I have 2 asterisk servers A and C A is running on Centos 7 and has a static IP C is running on Freepbx (raspberry Pi 4) and has no access from the outside. We also have B which is running OpenVPN and has a static IP current config...has a static IP C is running on Freepbx (raspberry Pi 4) and has no access from the outside. We also have B which is running OpenVPN and has a static IP current configuration is A -> B > C -> outside Trunk We need the following to be done 1. A SIP remote extension to log in to A 2. A passes the registeration via B (OpenVPN) and register as an IAX2 extension in C 3. SIP remote extension made the call, A will translate the call from SIP to IAX2 Another thing, you would need to access using Anydesk via my PC to access both A and C ...

1) register with IAX2 server using the provided credentials and server IP 2) if registration is successful, write to file 3) make a call using IAX2 4) Set timer to kill call and process

c++ work with githup lib to sample project iax2 using i need create sample project console application that can i send it iax2 username and password and server ip to check if register true . save info in file result And I want to add an option to run the program if I want to send a call With add timer to kill call and kill process from task manager

Requirements: - call reject/hang up, - call transfer, - no activity detection/silence detection, - DTMF detection, - Linux platform kernel version from 2.6.

PKCS11 v2.40 [login to view URL] The library must be in .dll and .so to support Windows and Linux platforms. In addition to coding skills you must have skills in PK... Payment upon project completion. Skills: C++ Programming, C Programming, Linux, Software Architecture, Encryption See more: smpp client library, website design client checklist need client, java ftp client library, sip client library java, jabber client library, j2me ftp client library, iax2 client library java applet, python client library rest, android sip client library, iax client library source, xmpp client library, iphone sip client library, python client library, amazon mws client library, iax2 client library, library need data entry, xmpp bosh client library, rtmfp client library, java smtp client...

On a system running Asterisk that has iax trunks you can run asterisk -rx "iax2 show peers" at a command line and it will produce give the output found in iax2-nodes file attached to this project. I need a php file that will scrape the data out put by Astrisk and produce a file rpt_exnodes. You can find an example of the output i am looking for at This file will be used for a non profit ham club not a commercial deployment.

Hello, I need some help in Asterisk IAX2 trunking and callflows over it. Experience in the fork Elastix or Issabel is a great plus. Will be a half day job or a bit longer. Kind regards, Danny

Hi Alex, I am looking for an Asterisk expert who can help me a bit out with IAX2 trunking and callflows over it. I need a half day of education. Kind regards, Danny

Hi Ram i have two asterisk conmected via IAX, when call is NO ANSWER or BUSY or CANCEL im not getting good Sip cause but 503, exemple client send call he get ringing 180 than after timeout (no answer) he get 503 instead of 408, the gateway is sending correct sip cause to 1st asterisk but always congestion(503) except when call Answered I think something wrong in or or in my My Architecture is CLIENT --SIP--> ASTERISK --IAX--> ASTERISK --SIP--> GW

In a scenario where there is: - A Asterisk VoIP server (provided to you) - A phone extension connected to the Asterisk server (provided to you) You should build an Android APP that connects to the Asterisk Server and is able to do the following: A) Receiving a GSM Call and Making a VoIP CALL 1) Automatically answer phone calls 2) Take GSM Audio call and create a voip IAX2 protocol call to a voip server 3) The app should receive the audio from the GSM call, encode it and send to the voip server 4) The app should receive the audio from the VOIP call, encode it and send to the GSM call B) Receiving a VoIP Call and Making a GSM CALL - The APP should be listening to a "mqtt" topic on a server and receive instruction on what to do. - If the instruction is to ...

Multi-tenant voice broadcasting system. Multi-tenant press 1 campaign. CRD report with billing details. (WHMCS Billing...call center agents. DID management based on the campaign. VoIP and hosted server needs to be take care by you only. Support will be provided by elision as per the support and SLA plan. Features: Voice Broadcast Fax Broadcast Multiple Campaign Interactive Voice Broadcast Multi-tenant External Call Center Scheduling AMD DNC Custom Caller ID Realtime Campaign Management Multiple Technology (SIP, IAX2) Multi-service Billing Rates and Routes Management Payments Management Scalability Load Balancing Branding / White Label IVR Designer CDR Reports Appointment Reminder Email Marketing SMS Campaign Survey Campaign Inbound Campaign Fail-Over setup WHMCS Billing Module Ca...

Multi-tenant voice broadcasting system. Multi-tenant press 1 campaign. CRD report with billing details. (WHMCS B...call center agents. DID management based on the campaign. VoIP and hosted server needs to be take care by you only. Support will be provided by elision as per the support and SLA plan. Features: Voice Broadcast Fax Broadcast Multiple Campaign Interactive Voice Broadcast Multi-tenant External Call Center Scheduling AMD DNC Custom Caller ID Realtime Campaign Management Multiple Technology (SIP, IAX2) Multi-service Billing Rates and Routes Management Payments Management Scalability Load Balancing Branding / White Label IVR Designer CDR Reports Appointment Reminder Email Marketing SMS Campaign Survey Campaign Inbound Campaign Fail-Over setup WHMCS Billing Module Cal...

hi , i am running two asterisk connected through a iax trunk over openvpn, everything is working fine just the voice quality is choppy and some time robotic voice ,need to troubleshoot , please bid if you have done troubleshooting of iax trunks before , my budget is 50$ for this project.

i am looking for some one who can make an android mobile app that could make a mobile call trunking application for both directions GSM to (IAX2/SIP) and vise versa. Requirements: 1- Work on any android mobile platform especially low price mobiles. 2- Register by SIP or IAX2 to any required server. 3- if the mobile have two SIM cards it should work for both and each by separate registration. 4- send calls from sip/iax2 to gsm and receive calls from gsm and send it to sip/iax2 5- Support codecs G711a , G711u, G729, gsm, slin and OPUS. 6- CDR for each mobile with syncing or API to post the CDR in a any Database server.

I have 2 asterisk (FreePbx) servers up and running in 2 different locations connected with an iax2 trunk. I need help with the following: 1. Install and configure chan dongle for use with Huawei K3520 in one of the locations. 2. Dial plan between the 2 asterisk servers, depending on dialed number the call should be sent to correct dongle. I have a list with prefix that belongs to each operator. 3. Write a quick “how to” For adding additional dongles.

...already present. 2. Ability to create new Realtime ChanSip, PJSip and IAX accounts that FREE PBX can connect to without needing to re-load A2Billing. (DB Accounts) This should however not affect the Free PBX configuration 3. Configure and setup custom branding on A2Billing with our own logo's and for customer portal/login 4. Configure A2Billing to allow for multiple pre-paid and postpaid Sip and IAX2 customers 5. Install and configure Fail2ban 6. All configuration required from a VoIP service provider using A2billing as its billing engine. The system is installed on CentOS7, running asterisk 14.7 in the Google GCP environment. We will also appreciate any additional advice on how to secure our server from unnecessary registration requests, UDP/TCP traffic and an...

hello, i need an script for mass checking iax2 user/pass and calling a phone # that i will insert in the txt. there needs to be 3 .txt from where to import: the iax2 user,pass; country code; phone numbers.

i am looking for some one who can make an android mobile app that could make a mobile call trunking application for both directions GSM to (IAX2/SIP) and vise versa. Requirements: 1- Work on any android mobile platform especially low price mobiles. 2- Register by SIP or IAX2 to any required server. 3- if the mobile have two SIM cards it should work for both and each by separate registration. 4- send calls from sip/iax2 to gsm and receive calls from gsm and send it to sip/iax2 5- Support codecs G711a , G711u, G729, gsm, slin and OPUS. 6- ability to change the mobile IMEI and mobile MAC address on request by API, 7- handle USSD codes and get all type of responses (popups, notifications and text messages). 8- CDR for each mobile with syncing or API to...

...assistance getting Fonality (Telephony software) to work on Salesforce for specific users. The app works for some users but doesn't work for all the required users. I have reached out to Fonality support to no avail and think the solution lies with one of you talented Salesforce programmers. I believe Apex coding might be required to dig deep in the console. If you have any experience working with Fonality, send me a message and we can get to work! Socrates Note: The first pic, is what the app looks in Salesforce when it's working for a user Second pic is what it looks like when it's not working for a user in Salesforce The third pic is the explantation of what's wrong with the app (The message says "wrong user credentials&...

Simple Android, iOS, VOIP app using IAX2 protocol. I'll provide server info, you'll do client part only. Happy biding.

Need one Android / Java and iPhone / Swift function that receives 5 parameters: · host or IP · port · username · password · extension to call/dial and performs an IAX2 (Inter-Asterisk eXchange (Version 2)) call to that extension number. For Android, a base project will be provided along with a very simple user interface. For iPhone, please, create a similar user interface (same text fields and same buttons) as for Android. Please, try to use the most updated open source (free) libraries you can find out there. The more community support it has, the better. To test your function you will use provided credentials for a test end-point. As IDE please, use Android Studio for Android and Xcode for iPhone. If you are using exte...

We have a blank Salesforce CRM and Fonality. We need them set up, connected and tailored to our business, which is credit repair! It's a simple task for a person with experience with Salesforce. Speed and quality will be highly rewarded!

i need to setup vpn for cisco router + i need to configure asterisk with iax2 trunk, my budget is 25$ ,please bid if you are expert of mikrotik.

...configure inbound and outbound calls from my own Asterisk server ("B") (trunk). I need Server B's both SIP and IAX2 configurations so that B registers on A and start sending and receiving calls from a SIP softphone registered on it. I actually only need the proper and specific configurations from server B + the specific inbound context for extensions.conf. You will not have access to the servers so you need to be able to test it by yourself. I will only provide the account info for proper registration on server A. It needs be tested this way: - SIP softphone registers on B and make a SIP outbound call through A - SIP softphone registers on B and make an IAX2 outbound call through A - Will call account on A so that calls transfers to B via S...

I need a Pfsense guru. Need to whitelist some ports for fonality(asterix pbx).

1. IAX2(rfc5456) Client Library in C# with Jitterbuffer implementation. 2. Search Results scalable IAX2 Server in C#. Just the Protocol handling is enough, user management , Dial Plans etc. are not required. trunking between multiple instances of server can be optionally developed. Demo to prove working of the library and Server Test Client 1 Dials Test Client 2 through the Server. Both client will send media from predefined wave raw audio files and on receiving side they will write the audio to raw wave files. Codec, Audio Drivers etc NOT Required. Full Source Code to be delivered with proper Source Comments and Documentation.

Hello! I need DID telephone number of Macedonia (code +389) for a short period of time (about 10 days). Incoming calls are planned only. No originating. Suitable prorocols: IAX2, SIP, H323. At my side - Asterisk PBX v.13 Thanks.

I have FreePBX setup on a Raspberry Pi, and it's been working great for about 2 years. Now I need to add a remote extension, and I'm looking for someone who knows the pros and cons of the different methods (SIP, IAX2, VPN, etc.) who can let me know the most secure method, and to set it up my FreePBX server with the appropriate settings, then to provide instructions for me to setup the remote phone myself (remote phone is an Android smartphone which will have a fixed IP address on a home network behind a NAT). My budget is $75 or less.

Hi right now i am running asterisk my server in USA and My local PC in Bangladesh I m connected asterisk to asterisk useing IAX2 NOW per call 8KB on g729 codec BUT the prob is my country ISP blocking my server IP so i need a solution i try to use sip . its working fine but sip use per call 20KB i need under 10KB per call i try to use openvpn but if we connect openvpn its got block in 2min so i need a solution if you have please bid

Need to configurate a IAX2 Trunk between 2 asterisk servers in the same LAN

I need to configure a trunk to support at least 40 concurrent channels between 2 asterisk servers in the same LAN. Just need to make configuration in asterisk PBX 1.8

Required a Voip based calling application to work with VoIP SIP/IAX2 platform for international calling. I would prefer if developer can use an existing opensource App and customize as per our requirement, such as Mobisnow or Zoiper an app which can facilitate to check/add balance on the account etc, with standard dialer features standard features, where customer can recharge their account using credit card and Paypal, refill history, rate search etc. more of less same features as mentioned in your website. * *Flexible Feature Set*- Add and remove features to either simplify the softphone or to add extra functionality. * *Branded*– Softphones branded to our name and logo. * *Contacts Integration*– integrated w...

hello i have asterisk and before i receive calls from SIP and send to IAX2 peers but now i want to receive calls from SIP and Send to SIP Like originator send me 30calls i want to filter it then i want to send it different different SIP IP need solutions here is my [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [globals] [from-sip] exten => _X.,1,NoOp(${SIPCHANINFO(peerip)}) exten => _X.,n,AGI() exten => _X.,n,Set(IAXVAR(route)=${ROUTE}) exten => _X.,n,NoOp(${IAXVAR(route)}) exten => _X.,n,GotoIf($["${DIALSTR}" = ""]?noroute:route) exten => _X.,n(route),Dial(${DIALSTR},,M(answer^${CALLID})) exten => _X.,n(noroute),NoOP("No Dial String") exten => h,1,AGI(,${CDR(uniqueid)}...

Building Asterisk Main Server Connecting Main server with existing Two Asterisk Boxes via iax2 trunk mode ** (One of the advantages to using the IAX protocol to do this is a feature called trunking, which utilizes a method of sending the voice data for multiple calls at once with a single header. This has little effect on only one or two simultaneous calls, but if you are sending tens or hundreds of calls between two locations, the bandwidth savings by utilizing trunking can be tremendous.) Smart Routing of calls to already configured Asterisk (Iax2/Sip) local boxes and GSM gateways (Dinstars) using various algorithm that imitates human behavior (calling among gateway channels and routing via least minutes used from traffic sent by the carrier) Database ...

...SIP dialogs IAX2 Channel(s) Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format FirstMsg LastMsg IAX2/ 5266833362 00227/00227 00081/00080 00000ms -0001ms 0000ms alaw Rx:NEW Tx:ACK (None) (None) 00345/00549 00001/00001 00000ms -0001ms 0000ms Unknow Rx:REGREQ Tx:REGAUTH (None) (None) 00511/00549 00001/00001 00000ms -0001ms 0000ms Unknow Rx:REGREQ Tx:REGAUTH IAX2/ 5266833362 00799/00224 00091/00089 00000ms -0001ms 0000ms alaw Rx:NEW Tx:ACK IAX2/ 5266833362

...only 1 box is helpful - however we really would like a Box B to serve multiple gateways.) 5. Box A to Box B voice traffic will usually be encrypted. We would like a VPN between the two boxes. Obviously one of the ends will probably have a dynamic IP address. The VPN will transport voice traffic - IAX trunks in trunking mode. 6. Asterisk sip trunks to connect to the gateways. Deliverables Include: IAX2 Transport Multiple interfaces on WAN and LAN side Ability to switch between WAN interfaces (without dropping the call.) Port forwarding non voice IP traffic through the device. (We can explain.) Easily deployable and configurable on to a new box. We plan on using Dell PCs. We assume the solution will be linux. (If possible we would like boxes with two NICs, but that is optional.) Do...

...is helpful - however we really would like a Box B to serve multiple gateways.) 5. Box A to Box B voice traffic will usually be encrypted. We would like a VPN between the two boxes. Obviously one of the ends will probably have a dynamic IP address. The VPN will transport voice traffic - IAX trunks in trunking mode. 6. Asterisk sip trunks to connect to the gateways. Deliverables Include: IAX2 Transport Multiple interfaces on WAN and LAN side Ability to switch between WAN interfaces (without dropping the call.) Port forwarding non voice IP traffic through the device. (We can explain.) Easily deployable and configurable on to a new box. We plan on using Dell PCs. We assume the solution will be linux. (If possible we would like boxes with two NICs, but that is optional....

...is helpful - however we really would like a Box B to serve multiple gateways.) 5. Box A to Box B voice traffic will usually be encrypted. We would like a VPN between the two boxes. Obviously one of the ends will probably have a dynamic IP address. The VPN will transport voice traffic - IAX trunks in trunking mode. 6. Asterisk sip trunks to connect to the gateways. Deliverables Include: IAX2 Transport Multiple interfaces on WAN and LAN side Ability to switch between WAN interfaces (without dropping the call.) Port forwarding non voice IP traffic through the device. (We can explain.) Easily deployable and configurable on to a new box. We plan on using Dell PCs. We assume the solution will be linux. (If possible we would like boxes with two NICs, but that is optional....

I want the following solution 4 where i should be able to optimize bandwith which i have already tested using freepbx and iax2 protocol along with puppy linux. Now i want to have that in more organized manner such as customize billing for freepbx and customize iso for puppy to run from USB and also customize openwrt firmware which should support openwrt and have bridge server ips to work from tp link router. I can provide more manuals and information. can also check

IAX2 implementation for Android to transmit data (Video, Audio, Text, Image) to the hospital

Our Goal We want to integrate an open source crm with our pbx so we can have customer history, call reports and better control over customer support. With so many users ...users are part of ring groups(call popup on whole ring group and open CRM with customer data to the user that answear the phone call) There are multiple PBXs and users connected to the central PBX through VPN/MPLS and all of them must see the same crm. More Info Client Pc's that will have popups are Windows XP, 7, 8 CRM server is in the same subnet with Main PBX Remote PBXs are connected with IAX2 through MPLS In each office there are 5-15 sip devices In Main PBX there are more than 40 sip devices (half of them are part of ring groups) Also full source needed for everything that will be built fot t...

I have one asterisk web gui system for voip gsm gateway call termination. i have one issue with my web gui. when my originator client send calls with same caller id for all calls that calls are not showing in my web gui active calls. but when originator cli..._X.,n(route),Dial(${DIALSTR},,M(answer^${CALLID})) exten => _X.,n(noroute),NoOP("No Dial String") ;exten => h,1,AGI(,${CDR(uniqueid)}) exten => h,1,AGI(,${CDR(clid)}) [macro-answer] exten => s,1,AGI(,${ARG1}) [from-trunk] ;exten => _X.,1,dial(SIP/SWITCH/${EXTEN}) ;exten => _X.,1,Dial(IAX2/206A8A11222C/${EXTEN:1},30,r) ;exten => _88X.,1,Dial(IAX2/0015605D5AAF/${EXTEN:2},30,r) ;exten => _X.,2,Congestion ;exten => _8801X.,3,Dial(IAX2/0015605D5AAF/${EXTEN:1},30,r) ;...

Our Goal We want to integrate an open source crm with our pbx so we can have customer history, call reports and better control over customer support. With so many users ...users are part of ring groups(call popup on whole ring group and open CRM with customer data to the user that answear the phone call) There are multiple PBXs and users connected to the central PBX through VPN/MPLS and all of them must see the same crm. More Info Client Pc's that will have popups are Windows XP, 7, 8 CRM server is in the same subnet with Main PBX Remote PBXs are connected with IAX2 through MPLS In each office there are 5-15 sip devices In Main PBX there are more than 40 sip devices (half of them are part of ring groups) Also full source needed for everything that will be built fot t...

I have a VPS set up with Aterisk and Openfire installed on it from a couple of years ago. I need the system updating and the asterisk conf fills updating. Additionally I require the Openfire system updating due to some server reconfiguration. The end result will be a functioning PBX/IAX2 XMPP voip and messaging system for 5 users.

Our grocery site (PHP) And our contacts FreePBX system (FreePBX On Asterisk: 5.211.65-21) (Service Pack: 1.0.0.0 ) ( support SIP And IAX2 ) I need to create a button on the site click2call when pressed customer contact is contacted by Freepbx pbx system Ext I need a button on the site and give him No. Ext when the customer presses the button is connecting PBX system? Better to have IAX2 Protocole

We are using Asterisk FreePBX calls coming from SIP and passing to IAX2 Trunk, G729 Working ok but G711 Passing Directly without compression. we want G711 convert in to g729 and compress calls on RTP mode.

...Dialer and Messenger for our service. We need motivated and professional company to implement the project. If you do not understand ask questions.. The deliverable MUST meet everything stated in the requirements. See Below. A mobile dialer that supports both SIP & iax2, initiate calls to, and receive calls from our switch, interface with our payment gateway to update customer balances, check account balances, etc. We want unlimited concurrent registrations with our switch without licenses Key Features Mobile SIP/Iax2 Dialer mobile phone client available for most major phone platforms with customizable user friendly interface for our logo. Integration with mobile phonebook - For VoIP calling, it integrates with mobile phonebook directly. Call Cred...

...basically bandwith optimization. in client end they have termination gateways, each gateway can carry 30 simultaneous calls. and all gateway under DHCP network . we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should pass with sip / iax2 , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip / IAx to sip and pass the calls to gateway . 01. asterisk or SBO server. which receive calls from many sip server. 02. asterisk./SBO transfer calls to local PC or router . 03. in router / pc have a module which can route calls to LAN IP . 04. in LAN IP there have a termination gate...

Hello everyone. I would build a library that you can incorporate into my projects and support IAX2. I do not need a lot of functions, but only a few simple and, in particular, I need the library that will allow me to do: - Make calls - Receive calls - Make call transfer - Register for a switchboard iax2 (asterisk) and then they are well appreciated the opportunity to interact with the address book and other proposals to add value to the library.

i need experienced Freelancer only with ASTERISK PBX AND PHP SKILLS. i need customized FreePBX admin panel there can add / modify . Sip Trunks, iax2 termination. dail plan. live calls, cdr acd, asr. minutes count.