Sipuraงาน

...puerto Syslog (UPD 514) y recibir todos los mensajes que envíen los distintos equipos pasarelas de Voz IP (VOIP Gateway) y procesarlos, que muestre en pantalla cuando cuelgan/descuelgan la bocina, marcan, timbra, tarifa, y las demás acciones que se presenten. El programa deberá interpretar los mensajes Syslog de los siguientes equipos como primera fase del desarrollo: - Linksys (ej. PAP2) - Sipura - Grandstream (ej. GXW 4XXX) Ademas deberá contar con base propia local (archivo) para manejar las tarifas establecidas y un pequeño reporte de llamadas (3 días). Como segunda fase de desarrollo: - Soporte con otros equipos: - Cisco - HTTEL - Integración con A2Billing - Desarrollo de un pequeño web API pa...

...puerto Syslog (UPD 514) y recibir todos los mensajes que envíen los distintos equipos pasarelas de Voz IP (VOIP Gateway) y procesarlos, que muestre en pantalla cuando cuelgan/descuelgan la bocina, marcan, timbra, tarifa, y las demás acciones que se presenten. El programa deberá interpretar los mensajes Syslog de los siguientes equipos como primera fase del desarrollo: - Linksys (ej. PAP2) - Sipura - Grandstream (ej. GXW 4XXX) Ademas deberá contar con base propia local (archivo) para manejar las tarifas establecidas y un pequeño reporte de llamadas (3 días). Como segunda fase de desarrollo: - Soporte con otros equipos: - Cisco - HTTEL - Integración con A2Billing - Desarrollo de un pequeño web API pa...

HOJE O SISTEMA ESTA CONFIGURADO COM AS INFORMAÇÕES DESSE SITE: COLOCAMOS AS CONFIGURAÇÕES LOCAIS DE CADENCIA E DISCONECT DESSE SITE: PROBLEMA: AO RECEBER UMA CHAMADA, APOS DESLIGALA O SP3000 NÃO DESLIGA A LINHA, SE ALGUEM TENTAR LIGAR PARA LINHA DA ERRO E SE ALGUEM TENTA LIGAR DA LINHA DA OCUPADO ATÉ QUE REINICIE O APARELHO, NOTAS ISSO SO OCORRE NA RECEPÇÃO DA CHAMADA, AO EFETUAR A CHAMADA E DESLIGA A LINHA É DESLIGADA E PODE SER OPERADA NOVAMENTE, OBS. NO MOMENTO QUE TRAVA O DISPOSITIVO FICA COM TODOS OS LEDS ACESSOS ATE REINICIAR. PRECISO DE UMA SOLUÇÃO PARA ISSO ALEM DE ENCONTRAR UM SISTEMA DE RESTAURAÇÃO DAS CONFIGURAÇÕES

I have a Juniper Networks SSG5 Router. I also have VOIP service. I need my router programed to allow the Sipura 2100 box to communicate from behind the router. It is highly likely that the programming can be done remotely or in conjunction with my assistance.

I have a SPA-3000 Linksys device which I want to use to: 1. Already have a voip account number 2. Need to route all incoming pstn calls to another number via VOIP

I need to perform a complete check on my asterisk pbx-free 1) I have a problem with the trunk that I can make calls but not receive 2) I need to configure a particular number in receipt to respond. 3) the exact configuration of a Portek 2 sim 4) Configuration of a sipura 2002 output

We want to connect 2 buildings by telephone by means of 2 SPA3000 devices and an Asterisk installation. There is an analog telephone line (PSTN, POTS, landline) available in 1 building. The 2 buildings are on the same internal ethernet network, even on the same subnet. The task consists of CONFIGURING the 2 Sipura devices and the Asterisk server. All hardware is up and running. The following is already present: * Building A: - 1 SPA3000 device, connected to a PSTN line - A permanent PSTN line - Ethernet network: the SPA3000 device A is connected to the network with fixed IP * Building B: - 1 SPA3000 device, with one analog telephone connected - A Debian server with a fresh Asterisk install on it, connected to the same network as the SPA3000 devices, with fixed IP

I have a SPA-3000 Linksys device which I want to use to: 1. Receive in my Xlite softphone calls recieved from PSTN 2. Make calls from my Xlite using PSTN phone line If possible determinate pressing a key ex #1 to use PSTN line or # 2 to make calls using VOIP temrination service provider such as , , etc.. ## Deliverables 1) All deliverables will be considered "work made for hire" under U.S. Copyright law. Employer will receive exclusive and complete copyrights to all work purchased. (No 3rd party components unless all copyright ramifications are explained AND AGREED TO by the employer on the site per the worker's Worker Legal Agreement). ## Platform NA

...Web Interface), we'll call it SERVER. ? * Several VoIP SIP phones connected to the network, one per extension. Some extensions are local on the same subnet, some are remote (like mine at home) ? * One VoIP provider (I already have an account configured for IAX, but i can easily change it to SIP) ? * Two regular PSTN land lines (#1 and #2). I can use Linksys (formerly Sipura) SPA adapters to convert those 2 phone lines to 2 local SIP accounts to be used with the SERVER. I believe two SPA adapters are much cheaper than a 2 ports phone card and easier to replace if one fails. ? * Please do not offer me to use hosted PBX solutions. I have my reasons why not and why I need to use those two phone lines in addition to a hosted VoIP account. ** ...

Im looking for someone to help me set up my sipura 2002 voip adapter. my trunk register, but Incoming calls are not. I know its sipura, because I do receive calls on x-lite softphone.

1. We need our lines that we already have to be connected to our up and running trixbox server and sipura units on the other end and working, 2. I want to be able to redirect my calls and record them if possible when I want to. 2. Also we need to learn or to set up a system for additional line setup if needed on our OWN without any outside help for possible new clients. 3. International calling would be a big plus. That's it as of now.

I have issues with Linksys PAP2 firmware. I have a need for a Firmware recovery tool specifically for Linksys PAP2. There is already one available for Sipura SPA2000 which was the predecessor to the Linksys PAP2 boxes and it talks perfectly to the Linksys PAP2's however the correct firmware is not in the exe file. I need someone to take this firmware recovery tool in exe file format and embed the firmware bin file so that I can recover corrupts Linksys PAP2 firmwares. I can supply as many PAP2 boxes with corrupt firmware as necessary. Regards, Paul Lewis-Brown

Need assistance configuring the PSTN interface of SIPURA SPA3000 to work as Trunck (For outbound call)in FreePBX system. The spa-3000 is located on a different subnet than the FreePBS server and SIPURA only have a dynamic IP while FreePX with a static IP

Hello I need help in setting up our openser. We are trying to integrate Microsoft speech server and our sip agents (ASterisk box, sipura phones etc.). All communication goes through openser. Speech server only supports tcp and a translation needs to be done which is completed and in a working stage. However, speech server sends 302 redirect and openser sends it back to the UAs and they get confused and moreover when they try to follow the new address in 302, they fail as it supports tcp and ua (user agents) can support only udp. I want some openser developer who can redirect that request within openser itself. Remember that the new contact is sent in maddr attribute in contact header. You will have to parse. There is code lying in the forums which is supposed to redirect and...

...Computers 2. Servers ??" 6 Servers with custom applications that require internet connection. 2 out of the 6 servers are web/app servers that will be open the Internet and should be placed in a DMZ. All servers have 2 NIC s (Currently one is not connected however can be connected if needed) 3. VOIP ??" I have one Linksys PAP-NAT adapter for Vonage and one Sipura 3000 box. The Linksys adapter requires DHCP and an outgoing connection while the SIpura box requires outgoing and a range of incoming ports that should be opened. 4. KVM over IP ??" Avocent DSR4020, allows me to use KVM functionality over IP. Access to KVM should be available from a VPN connection and from the internal network. 5. Wireless Linksys WRT54G ??" My Cisco router does not include the ...

Need assistance configuring the PSTN interface of my spa-3000 / 3102 to work as a line in my trixbox / FreePBX system. the spa-3000 is located on a different subnet than the trixbox server and will be located behind a router. The trixbox server has no Firewall / Nat to worry about. PM me for more info.

Looking for someone to install asterisk in an office. Asterisk will be used for 3 companies. we have one VOIP line each for two companies using sipura 2000. the 3rd company has two ptsn lines. for ist two companies we need standard PBx features . The 3rd company is sort of a doctors office. people should have option to contact the doctor by pressing a #. doctor should get hunted on his phone, email cell phone we have digium FXo card and will be using Cisco 4 line voip phones. Idea is to keep VOIP for incoming and outgoing calls for first two companies. For 3rd company , we would like to keep the pTSN lines but will use VOIP for outgoing calls ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of al...

I have already installed Trixbox on my server, the project now has changed to provide a bare minimum of functionality (I am dreaming big, but let's get it working and improve functionality later): hook the machine up with connect the Sipura 3000 to the POTS line and configure it correctly (I have done that before, should take an hour). Provide the following functionality System needs to distinguish between three DIDs (2 , one POTS line) according to DID voice announcement for POTS line: You have reached XYZ company, please wait while I connect your call voice announcement for DIDs: You have reached ABC company, para Espanol, oprima el numero 2 en este momento. IVR # 2 espere por favor, tratamos de conectar su llamada a la persona apropiada IVR #1

I am wanting someone to configure my Trixbox ASterisk VM machine, and Sipura 3000. I will provide a WWW interface on both units, and VNC/Remote Destop as required. This is all for a home PABX... Existing: Three SIP phones connected OK. Each has Voicemail. I also have an IAX2 trunk running, such that anything of 8 or more digits is dialed on the trunk. Required: Sip Phones Extensions 2000, 2001, 2002 etc PSTN-1 through SPA-3000 Outgoing via 9+numbers dialing Incoming to group 3005 by default If caller ID is 0412929634 then ring direct on extension 2000 PSTN-2 through SPA-3000 (Possible future) Outgoing via 8+numbers dialing Incoming to Group 3006 IAX2 through external company Outgoing via 7+numbers dialing Incoming to group 3007 Desirable One member ...

I know it looks like a big project but it is not. I am just trying to be detailed. These are currently features supported within Asterisk. Nothing should be foriegn ...after greeting after business hours unless user dials extension directly. Home setup will only have a greeting message and then go to voicemail on asterisk when busy. Outgoing phone calls for home Or office will use 1 voicemodem card for US calls OR 1 sip account for international calls. Business calls will be routed to PC where I will have Xlite setup. Home Calls will be directed to sipura during day. Conferencing - Must have a password protected area to hold voice conferences. We will use escrow and will not require anything be done outside of what is posted or agreed to online for your protection...

//*If the following text is garbled or unreadable, there is a ms-word file attached with the same content to this project*// -Situation Diagram: Carrier 'SIP' asterisk 1.2.4 'SIP' sipura-spa3000 'FXO'PSTN OR Carrier 'SIP' asterisk 1.2.4 'SIP' grandstream ht488 'FXO'PSTN OR Carrier 'SIP' asterisk 1.2.4 'SIP' [Any other single SIP FXO port] 'FXO'PSTN Problem: FAS on asterisk ' spa3000 call leg In words: Sipura/grandtream/clipcomm manufacture FXO sip gateways. When placing a call from asterisk to any of the above FXO ports with this command in - Exten => _.,1,Dial(SIP/${EXTEN}@FXO-one_of_ht488_spa3000_CG410-FXO) And using the FXO in single stage gateway mode...

[![Free Image Hosting at ][1]][2] I need an application that can read (polls) the CallerId for incoming phonecalls in my sip hardware: a Sipura 2002. The incoming phonenumbers should popup and also be stored in a logfile. Sipura adaptors have a buildin webpage () and there you can see the last callers. Please note that this model 2002 has 2 lines. The second line is just of the image but its there as well. Sipura Internal Web [][3] **General Info** The Sipura 2002 is a analog Telephone Adaptor that connects a normal phone to the Internet. Sipura is a Cisco daughter. Sipura Homepage: I use this model: ## Deliverables 1) Complete and fully-functional working program(s) in executable

...platform(s) specified in this bid request. 3) All deliverables will be considered "work made for hire" under U.S. Copyright law. Buyer will receive exclusive and complete copyrights to all work purchased. (No GPL, GNU, 3rd party components, etc. unless all copyright ramifications are explained AND AGREED TO by the buyer on the site per the coder's Seller Legal Agreement). ## Platform With multiple Sipura 3000 endpoints at several locations, have a VoIP provider[VoIP#1] provisioned to cover the US calling region. And whenever calls are made to Int'l destinations from the same endpoints I would like them to be routed thru a 2nd VoIP provider[VoIP#2]. Since I'd like to share the VoIP#2 service plan with the various endpoints I believe Asterisk would be...

The goal of this project is to provide secure voice communications with a VoIP phone. The phone in question is the Sipura SPA-2100 which already has all the functionality needed. The only missing part is the generation of 2 keys for the SRTP (Secure Real Time Protocol). There is already a public program that generates such keys here: But I would like to have my own keygen program, with source code. The program mentioned above will be used as a reference to make sure that your program provides the same functionality -except that yours will not be web-based. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows

I have an Asterisk at Home box that is all setup and works fine with my Sipura SPA-3000. One trunk is PSTN and the other is VOIP. I have a problem with incoming calls. For whatever reason, I cannot get them to be routed properly to Digital Receptionist. I can only route them to an extension. I also want to setup Kall8 VOIP termination with my Asterisk box. I will provide SSH and web control panel access to my Asterisk box to the selected bidder. I not only want those things to be done, but I also want to know "why" it didn't work so I can fix it if I ever run into the same problem. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condi...

...The POTS line connects to all my regular analog phones with standard wall jacks. I just want to have Asterisk "watch" the analog line, and if no person picks up the call with an analog phone within 5 seconds then asterisk will answer the call and go through the Script (attached). It is a pretty standard “company?? script. The asterisk server will be connected to the analog POTS phone line via a Sipura 3000. see attachment for all the details. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows (depending on the nature of the deliverables): a) For web sites or other server-side deliverables intended to only...

Hello, I have a working installation of Asterisk with AMP, as well as a Sipura 2000. Everything is connected through sixTel, and was set up via the AMP interface (which itself is working beautifully). I need an Asterisk expert who is extrememly well versed in Asterisk installations, especially concerning AMP, that can help with tweaking the configuration and answering my questions. A few specifics: - How do I access voicemail? - If I want to connect multiple boxes to my system via SIP, what is the best protocol? - I want to have several Sipura (or similar) boxes, how do I provision ports for them (5060, 5061..?)? - Right now when I make a call while watching the asterisk command line interface, it takes 9 seconds before asterisk will respond to the key press combination. That...