We use a XORCOM ELASTIX VOIP and have 2 office locations with a VPN tunnel. The second office cant make or receive calls which seems to be a SIP ALG or NAT issue. But need this fixed.
Hello We have SIP,PRI and want to build script where we can make OBD Calling (asterisk),
...VoIP provider. I gave the green color for your easier to understand. -- Executing [s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in new stack -- Executing [s@macro-user-callerid:38] Set("SIP/8902050098-00000024", "CALLERID(name)=+918820094576") in new stack [ログインしてURLを表示]: Caller ID name is ...
...provider. I gave the green color for your easier to understand. -- Executing [s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in new stack -- Executing [s@macro-user-callerid:38] Set("SIP/8902050098-00000024", "CALLERID(name)=+918820094576") in new stack [ログインしてURLを表示]: Caller ID name is ...
...retire our incoming ISDN lines and are setting up to test sip lines. We have an unusual router (peplink) and multiple redundant internet connections. We have spend many hours trying to setup our router to enable SIP connectivity however without success. We are looking for someone with 3CX, SIP and good networking /router skills. Hourly rate to be discussed
Hi Kristen H.,We would like to hire you to prepare a new catalog of @ 170 products currently on a GSA contract to mirror and add to GSA Advantage through the SIP database program. All files will be supplied to you. You will need to organize in the correct format and upload both product descriptions and photos.
We need an HTML based SIP client that can be designed to look and act like a in home intercom. For example, there should be buttons for rooms, that will let you page the rooms, and select either video or audio. This will have to be set up that each "station" can be configured which rooms it can page etc... There are a lot more features and customization
App to register with my Asterisk Server as a SIP extension. My Server will send VoIP calls to the App and the App will make a local call on the GSM network and path both calls together. In other words, the Android Phone will act as a VoIP / GSM gateway. Thamk you.
Objetivo: Provisionar teléfonos Cisco 7911G para plataforma SIP abierta (Voipswitch). Requerimiento: 1 - Selección de firmware compatible con SIP no propietario. 2 - Creación de "[ログインしてURLを表示]" para provisionamiento remoto.
Существующая сеть передачи данных и телефонии по SIP. Необходимо осуществлять мониторинг сети, программировать коммутаторы, медиашлюзы, поддерживать SBC и т.п.
...if VoIP package has enable, if enabled then execute real-time billing logic with client package rate and remove credit from credit balance. If client’s package has free minutes, check if over minutes then execute real-time process. Issue remarks in Manage Credit/Client Description field within table as follows: (you may change your variable names to
Hey everyone, I'm working on a project to develop an interface for a VoIP server to allow users to add their own extensions and modify their call routing. I need a developer that is an expert in Node.js as well as PHP because this project will be developed using both languages. If you have strong experience in both languages please contact me with
Hello, i want script to Test sip accounts with Back SIP response codes Example : HOST = '[ログインしてURLを表示]' SIP_PORT = 5060 LOCAL_IP = '[ログインしてURLを表示]' PROTOCOL = 'UDP' USER = '509' PASS = '509123' and it will return me with [ログインしてURLを表示] (200 OK or 301 Moved Permanently OR 401 Unauthorized etc...) +save
Hello, i want script to Test sip accounts with Back SIP response codes [ログインしてURLを表示] I will provide : Server ip : Port : Tcp/Udb : Username : Password : and it will return me with 200 OK or 301 Moved Permanently OR 401 Unauthorized etc... +save output into text file +Be able to run in multi-thread Job urgent
hello, i have this package : [ログインしてURLを表示] need to...package : [ログインしてURLを表示] need to install on my server windows then build api requests to manage users and add sip accounts SO i will be able later to use on my custom cms
I need a python script to: 1. answer a SIP call using pjsip 2. listen & send the audio to google speech api (file or stream) 3. get the recognized text back Silence should be detected to stop the file recording or the stream to google Websockets might be used as well
VLC server with the ability to stream locally, installed and tested FreePBX with the ability to connect 2 princess phones locally using either MGCP or SIP, installed and tested I have intermediate Linux knowledge and can assist
hi ,i am looking for a android devloper who can help me with opensource sdk for sip client like linphone , csipsimple [ログインしてURLを表示] bid if you have experience with sip app , i do not use microsoft products so bid if you are experience in working with linux [ログインしてURLを表示] will be long term project if satisfied with the [ログインしてURLを表示] budget is 100$ for this project
Customize microsip: Currently microsip allows parameters to be passed via command line Eg: [ログインしてURLを表示] /hangupall [ログインしてURLを表示] /answer [ログインしてURLを表示] 3892014 (...) You must - change the format of the arguments, that will be passed in the format below: [ログインしてURLを表示] msip:hangupall [ログインしてURLを表示] msip:answer [ログインしてURLを表示] msip:38192014 - add new methods that can be called by command line: - ...