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    167 excel 5060 công việc được tìm thấy, giá USD

    ...answer function for external call from PBX system. Before auto answer play AUDIO FILE 7 (incoming call), auto answer start in 2sec. SIP SETUP config file Sip pbx ip address/port 5060 Sip User/Pass Operator Calling number Time to auto answer Relay Unlock period Sip client must have VPN configuration to connect to Asterisk PBX over internet. Hardware

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    ...function for external call from PBX system. Before auto answer play AUDIO FILE 7 (incoming call), auto answer start in 2sec. SIP SETUP config file: Sip pbx ip address/port 5060 Sip User/Pass Operator Calling number Time to auto answer Relay Unlock period Maximal time for call Sip client must have VPN configuration to connect to Asterisk PBX over

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    ...des couturelles B/6224 Fleurus Tel : +32 81 11 77 67 GSM : +32 495 88 88 68 [đăng nhập để xem URL] (exemple ) Entrepôt de stockage et ateliers Rue des glaces nationales 5060 Sambreville Boîte mail Mail : contacticpbelgique@gmail.com...

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    ForiGate config Đã kết thúc left

    Disable ALG in FortiGate 80E and setup a NAT rules. Ports 10000-20000 UDP to Any And 5060 UDP to a specific range

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    ...спільну IP адресу з сервером pfSense для вихідного. Для вхідного трафіку відбувалося прокидання лише окремих портів на ці машини (приміром на сервер з Asterisk прокидаються порти 5060, 5061, 10000-20000 і т.д.) 3) всі віртуальні машини мають перебувають у спільній локальній мережі pfSense (з можливістю обмеження трафіку між окремими машинами); 4) основна

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    hi i have a ASTERISK SERVER and it have a SIP Account i have a kamailio5 installed on my local system now you need to register kamailio5 to ...to register kamailio5 to Asterisk using SIP account when asterisk server send call to kamailio5 ==========kamailio5 have to send call to local gateway IP like [đăng nhập để xem URL]:5060 if you can make it please bid

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    Asttp Support (Freeswitch) Đã kết thúc left

    ...Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;maddr={local_ip_v4};transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC) 2019-01-08 01:07:16.671190 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;maddr={local_ip_v4};transport=udp,tcp) ATTEMPT 3 (RETRY IN 5 SEC) 2019-01-08

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    Secure Issabel PBX (asterisk) Đã kết thúc left

    I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

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    Project for Prostitov A. Đã kết thúc left

    ...A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [đăng nhập để xem URL] secret GZD397XU Host [đăng nhập để xem URL] port 5060 Register String 390415541985@[đăng nhập để xem URL]:GZD397XU:390415541985@[đăng nhập để xem URL]:5060 B) TRUNKS MESSAGENET Protocollo SIP username 5406073184 From Domain [đăng nhập để xem URL] secret ajFFzpNj Host [đăng nhập để xem ...

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    multi asterisk in one server Đã kết thúc left

    ...different name and connect with different database 1main server other all asterisk server work like VM host1 IP [đăng nhập để xem URL] SIP 5060 DB NAME datastore1 host2 IP [đăng nhập để xem URL] SIP 5060 DB NAME datastore2 host3 IP [đăng nhập để xem URL] SIP 5060 DB NAME datastore3 host4 IP [đăng nhập để xem URL] SIP 5070 DB NAME datastore4 host5 IP [đăng nhập để xem URL] ...

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    Qos implemenation DSCP values Đã kết thúc left

    ...to implement skype for business in our infrastructure ... I would like to mark and traffic with these DSCP Values : Skype for Business Signaling, DSCP 28, source port TCP 5060:5079 Skype for Business Audio, DSCP 46, source port TCP/UDP 50020:50039 Skype for Business Video, DSCP 34, source port TCP/UDP 58000:58019 Skype for Business Application Sharing

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    ...Ubiquiti Edgerouter at the remote site, establish an IPSEC VPN to a Ubuntu 16.04 VPS server and then have all SIP traffic route over the VPN to allow 5060 traffic out of Dubai which blocks SIP on 5060. Install IPSEC on the Ubuntu server (currently in production hosting web app) Configure VPN Ensure only SIP traffic flows over VPN lock down the entire

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    Project for Yasin A. Đã kết thúc left

    ...1) Upon installation of Softphone, prompt for username and password 2) It will then login to a website (MySQL) and get the pbx server hostname, username(ext), password, port (5060 or 5160) and URL for customer logo 3) configure the phone with the provide info and replace default logo with customer logo. If customer logo do not exist use default logo

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    ...ails&id=1&Itemid=621 Here's an example of how I accomplished this years ago, but with a Jsom, if you look at a field Swap for My listing: and enter a valid listing ID (try 5060), it will then be imported via RSS and Jsom and become part of the message HTML: [đăng nhập để xem URL] I need a more visual and modern aproach, ideally when a user

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    Simple SIP redirect Routing system ,listen on port 5060 .It authenticate the incoming SIP invite source IP address , retrieve data by connecting to Mysql or Sql server, get values that should be appended to sip contact header edit/append sip contact : content ,then send it back to source of invite ,wait for ACK and send invite again if no ACK

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    I have a FreePBX server and I need to route all VOIP traffic by port. 5060 and 10000 through 20000 through 1 gateway and all other ports through another gateway. All in same subnet same interface. Our ITSP (Spectrum enterprise) will not allow anything other that VOIP traffic through their Voice network. Diagram enclosed. Something like this. http://www

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    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

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    Project for GrArtem Đã kết thúc left

    ...matter what I try, I simply can't get the outgoing calls to work. I get errors like this: Call to <sip:14057577446@<xyzname>.[đăng nhập để xem URL]:5060> has failed; Cause: 403 Forbidden/INVITE from [đăng nhập để xem URL]:5060 I really need to get this fixed soon. Please let me know if you can assist and how much you think it might cost to fix the outgoing ca...

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    ...back to ISP1 AS SOON AS it’s back online. 2- Need redundant ISP Internet output on Eth0 and Eth3 with DHCP Enabled [đăng nhập để xem URL] Leasing from 100 to 200. 3- Port forwarding 5060-5061 UDP and 10000-20000 UDP to IP [đăng nhập để xem URL] from both ISP interfaces to mention IP through eth3. 4- Need another Internet connection eth4 from ISP1 only with DHCP 192.168

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    Hire a PHP Developer Đã kết thúc left

    ...8029299 Content-Disposition: form-data; name="PageCount" 3 -------------------------------28947758029299 Content-Disposition: form-data; name="FaxImage"; filename="[đăng nhập để xem URL]" Content-Type: application/octet-stream {Image_data} -------------------------------28947758029299-- The sample XML output should look like: <SendFax>

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    ...ALG router / firewall setting disabled. 2. Add an "any/any" rule using the proxy IP of [đăng nhập để xem URL] 3. Open these ports on the switcher -Sip Signaling ports: 5060-5090 UDP -Audio Ports: 10000 - 64000 UDP 4. Block the incoming traffic to prevent the switch from being attacked. 5. I need this done without knocking out my voip

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    Configure Cisco 1921 Đã kết thúc left

    Please create a Cisco 1921 config for me. It is for fibre internet; - WAN Port GE0/0 - ISP ...[đăng nhập để xem URL] - LAN DHCP Range [đăng nhập để xem URL] to *1.20 - Please secure the router with new passwords - Setup NAT - Allow SSH access locally only - Set up example port forwarding 5060 external to [đăng nhập để xem URL] Please set up other features you think will be good.

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    I have 3 Routers Mikrotik in different places Now the 3 Mikrotik are connected in a VPN PPTP. Have access very good This is the connection Mikro...Client PPTP I need setup of QUEES (For QoS) I need manage the bandwith for example: Total Bandwith (2MB) 726K for PPTP Service, SQL SERVER 724k for 80, 443, 448 700k for 5060 (SIP Service)

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    i have an asterisk server for call termination in a country where voip ports are blocked (5060 to 5090) so i need to create a vpn sip tunnel just like [đăng nhập để xem URL] service plz refer to their website for info. data will go encrypted using ssl from my ubuntu linux server, on https port 443 then on my other remote server will be decrypted and converted

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    ...KAMAILIO sur un port différent du 5060 et que ce dernier relais les appels entrants et sortants entre le serveur ASTERISK d'OVH (sur lequel est installer ma ligne SIP) et mon téléphone IP. Le but est d'utiliser sur KAMAILIO un port commun comme le 443 ou autre pour bypasser le blocage du FAI qui semble bloquer le port 5060/5062. J'ai vu que ...

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    ...with the Axis IP C3003-E speaker, we cannot use VPN so we use the public port of the server running asterisk and have the client open the outgoing and incomming ports needed (5060 tcp and 10000-20000 in udp). The problem we are experiencing with a customer is that the sound is not coming out of the Axis speaker and we have tried everything but nothing

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    ...want to create firewall Policies for each of those VLANS. 1 VLAN - Allow all incoming and outgoing 2 VLAN - Only allow our VOIP Server, so block all ports except for 5060 and 80 3 VLAN - This is a guest network, mainly want to block torrenting and adult sites. There will be ongoing work as the network will be developed further. Also open

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    ...168.1.2:5060 SIP/2.0 Via: SIP/2.0/UDP [đăng nhập để xem URL];rport;branch=z9hG4bKppggmt5mg1avj Max-Forwards: 70 From: <sip:01152588200 @ [đăng nhập để xem URL]>;tag=0tpgSjXtr3ctD To: <sip:carlos @ [đăng nhập để xem URL]>;tag=as00a203e3 Call-ID: 3dccd3dc4a295a7c5391c95377a45a0d @ [đăng nhập để xem URL]:5060 CSeq: 96649882 INFO Contact: <sip:011525882XX ...

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    For example a ITSP currently running the VoIP Service on a server which in US: [đăng nhập để xem URL], the port is 5060, provider want extend the VoIP service to UAE(the country which blocked the VoIP). Provider just need to set up a PERS server and place it in a country which not block the VoIP, such as Japan, US, Germany etc., assuming the PERS server

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    Asterisk Call Forwarding Đã kết thúc left

    ...sdpsession=Zoiper ra47a93e context=Zoiper ra47a93e allowguest=no directmedia=no allow=all allow=alaw allow=gsm trustrpid = yes sendrpid = yes udpbindaddr=0.0.0.0 bindport=5060 useragent=Zoiper ra47a93e 1111111111111 register => 1111111111111:pass@k1111111111111/1111111111111 [k1111111111111] type=peer secret=pass insecure=invite use

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    ... The main purpose for it is that the ISPs here in my country blocked the SIP protocol with all of its related ports so we have tried to change the default ports from 5060 to any other with changing the RTP but did not workout, tried to use the VPN solutions, it's works but it's unstable and leads to a noticeable voice quality issues. So

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    I have a python script that needs to be reworked. It is a short script that listens on port 5060 for a sip subscribe message and it responds back with information about where a provisioning server is located. I will provide the script to be changed and also a wireshark capture to show what the sip subscribe message will be like. The original script

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    Sip Proxy Server Đã kết thúc left

    I need to bypass my ISP as they are blocking port 5060, and other port for SIP. I need to know if I can bypass using proxy to connect my PBX.. Based on my research I want SIP Proxy and RTP Proxy installed on my ubuntu which will be used as proxy server to bypass there are software like reSIProcate and rtpproxy . I want some one to install and

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    ...servers.. The main purpose for it is that the ISPs here in my country blocked the SIP protocol with all of its related ports so we have tried to change the default ports from 5060 to any other with changing the RTP but did not workout, tried to use the VPN solutions, it's works but it's unstable and leads to a noticeable voice quality issues. So I

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    ...servers.. The main purpose for it is that the ISPs here in my country blocked the SIP protocol with all of its related ports so we have tried to change the default ports from 5060 to any other with changing the RTP but did not workout, tried to use the VPN solutions, it's works but it's unstable and leads to a noticeable voice quality issues. So I

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    ...servers.. The main purpose for it is that the ISPs here in my country blocked the SIP protocol with all of its related ports so we have tried to change the default ports from 5060 to any other with changing the RTP but did not workout, tried to use the VPN solutions, it's works but it's unstable and leads to a noticeable voice quality issues. So I

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    Outbound Proxy Server "For SIP" Đã kết thúc left

    ...servers.. The main purpose for it is that the ISPs here in my country blocked the SIP protocol with all of its related ports so we have tried to change the default ports from 5060 to any other with changing the RTP but did not workout, tried to use the VPN solutions, it's works but it's unstable and leads to a noticeable voice quality issues. So I

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    ...who want to try to solve (only needs a softphone and SSH access). When I turn on iptables, my Asterisk serve cant send invite (using UDP 5060) to my another asterisk but he can register normally (using UDP 5060 too) . All servers have public IP, and are not under NAT. With tcpdump I see the package being sent on the source server but not caught

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    When I turn on iptables, my Asterisk serve cant send invite (using UDP 5060) to my another asterisk but he can register normally (using UDP 5060 too) . All servers have public IP, and are not under NAT. With tcpdump I see the package being sent on the source server but not caught anything in the destination, if I turn off the firewall, the problem

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    When I turn on iptables, my Asterisk serve cant send invite (using UDP 5060) to my another asterisk but he can register normally (using UDP 5060 too) . All servers have public IP, and are not under NAT. With tcpdump I see the package being sent on the source server but not caught anything in the destination.

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    ...Under the command console. (CLI) This requirement is only the client application but in IOS-APPLE. The server application, is ready to test your development. [[đăng nhập để xem URL]:5060, UDP packages only] This is an example, therefore the UI should be the minimum necessary to write the message chat. I repeat, "the minimum necessary to write the message"

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    ...need request this service of development of this software, to one programmer experienced for develop it in Smartphone IOS / Apple, which send chat messages over UDP, use port 5060, and generate UDP traffic until server. One example is available on the web and really we want that same example but en IOS/APPLE. Example: [[đăng nhập để xem URL]

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    ...port E3-0/3 Global Configuration Add 1. Add Banner to configuration file 2. Disable SIP-ALG (Application Layer Gateway) 3. Add no ip nat service sip udp port 5060 4. Optimise router for voice usage 5. Allow external ping requests 6. Add PAT's...

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    ...followed by number. i.e. 441554788822 The configuration file will contain the following data Proxy server: For example [đăng nhập để xem URL] Outbound Proxy port: For example 5060 Outbound Proxy server: For example [đăng nhập để xem URL] Registration Period: For example 600 SIP-ID: For example 2319801e0 SIP Password: For example ucccccQdgA2 Local

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    FREE PBX environment Đã kết thúc left

    ...access to any LINUX version you want. 2. You need to secure the FreePbx environment and Ubuntu Linux. 3. You need to close all default SIP 5060 ports. we need to run the free pbx on custom ports not the default 5060 ports. ...

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    Project for DreamersLTD -- 3 Đã kết thúc left

    Hi DreamersLTD, we would like to offer you a new project. Translation of a repair manual to French translation (total 5060 words) for the amount of $200.

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    I have a AAstra 6753i phone behind a NAT and a PBX hosted on a public IP. I am able to register another SIP device but not this phone. Need help to setup phone and exten...IP. I am able to register another SIP device but not this phone. Need help to setup phone and extension properly so that port registration occurs on a dynamic port and not 5060.

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    Help configure Asterisk Đã kết thúc left

    I'm trying to connect to a certain carrier (this carrier has ...it's based by IP Authentication) yet is not working correctly. here is the current [đăng nhập để xem URL]: [kwak] canreinvite=no context=kwak qualify=yes host=[đăng nhập để xem URL] port=5060 type=peer disallow=all allow=ulaw trustrpid=yes sendrpid=yes insecure=port dtmfmode=rfc2833 nat=no

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    Configure a Sip Proxy Server Đã kết thúc left

    We Need a SIP expert to Install + Setup outbound sip proxy server using Ubuntu or CentOS. The aim to run sip communication on ports other then 5060 which is blocked by certain ISp's. Payment will be release when the project is done with all install and configure documentation.

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