Para ajudar os clientes a escolherem onde hospedarem seus projetos e até mesmo desenvolvedores menos experientes, fiz esse top 5 provedores.
WE have an application similar to VOIP, does not tolerate delay. network: A---1GB---cisco 3560----40MB MPLS NTU---4x separate offices Goal is to shape outgoing traffic from A to NTU, not drop , like this: if dest_ip is 1.1.1 shape to 10MB(each office), buffer excess burst, do not drop packets. In summary, design QOS shape policy on 3560 to shape and buffer traffic to 10MB per office. Its simple network, will provide diagrams and configs
We are looking for an Asterisk or Elastix developer capable of customizing and integrating the soft PBX with a third party distribution. To those of you that claim to hold the skill requested we'll delivery proper documentation about the job. Thank you.
I want to add codec G729 to increase quality call on my system and make a modification on my server, check my setting to ensure i got them right. Currently cannot make international calls, suspect codec as my providers use G729, G711, G711u etc and not ulaw or alwaw which are default for a2billing. A few small bits here and there.
We are looking for an Engineer to be able to install the following equiptment in Gerogia USA (Columbus, GA 31907) Cisco ISR 4321 It's importat to mention that we just need te instalation and configuration of the equiptment with a duration of no longer of 4 hours. If you have any doubts please contact me. The job must be performed on site, we need to do the Unboxing, Racking and Configuration
i have hypermdia hg4000 series i need to configure hypermedia gsm gateway to work fro call termination. with the following features. the most important : 1-configure the gw to with my softswitch to reicieve calls ( need setup because internal router is reset) also i would prefer the following features 1-topup by sms 2-create costum ussd scritps to automate running on sim cards. 3-abilty to change multiple imeis at once. every new feature is welcomed.
We are a VoIP company using both MiRTA and FreePBX (eventually just MiRTA) and we are looking for support when needed. We have lots of provisioning requests and some (but few) complicated troubleshooting requests that we would need help with. We would prefer someon that speaks English and works during US working hours.
I am looking for a teacher to give CISCO Courses, English speakers, Cisco CCNA CCNP Juniper JNCIE JNCIP JNCIS JNCIA interested people send the course that are able to teach online.
Hello, I am looking for a GoAutodial expert with proven experience to do a few work for me on a monthly retainer basis. I already have the server running and your first job would be to secure it / put in place everything required to prevent BRUTEFORCE attack of my SIP, SSH and other usual attacks on those type of servers. Once this is done, I will pay you monthly to do maintenance on my server and from time to time a few adjustments. I am looking for serious Freelancers, who are very responsive and preferably living close the the GMT+4 regions. Thanks, Selven
Needed: Experienced PHP programmer for VoIP needed, knowledgeable in telecom programming particularly GoIP sim bank gateways, GoIP API programming, Asterisk A2Billing and FreePBX, and Cisco. Description: Create an API to control the actions of a GoIP sim bank gateway by Call Progress Tones. GoIP sim gateway is located in a third world country to which the selected contractor must program the proper Call Progress Tones. The GoIP sim bank gateway provides the termination into the country. A PHP program is then needed to conditionally control the Call Progress Tones that are sent back to the Asterisk on the origination side. The following acceptable Call Progress Tones (CPT), specific to the terminating country, should be selectably programmed to send back to the origination side: Ringing Tone Busy Tone Congestion Tone Dial Tone Number Unobtainable Tone Waiting Tone Holding Tone The GoIP sim bank gateways should conditionally act to capture any other Call Progress Tone, which essentially would represent an unwanted condition on the line, and redirect (hunt) the call to another available SIM across the GoIP gateways. This routine should continue until the GoIP can handle the call with the above acceptable Call Progress Tones. The API program should set a flag for the SIM that had the unwanted Call Progress Tone condition. The flag shall have an operator settable variable representing elapsed time prior to which another call may be sent to that particular SIM. For example, a set variable of 3 (three) would represent an elapsed time requirement of three minutes prior to the SIM card port being available to attempt another call. When this flag condition occurs, an SMS message should be sent to 4 phone numbers (set by data entry into variables) indicating the Sim on which this condition occurred. These numbers may be programmed in a separate data entry routine. If it is preferred to program this feature within the same CPT API routine it should have a time out feature (i.e. 1 min) so that the CPT API routine may continue unattended. This SMS send of the flag condition function should also be able to be turned off. The operator shall also identify the selected flag condition (one CPT in particularly) that this routine shall act on by use of the methods for diagnosing CPT described in the website below. A "CDR" reporting feature of unwanted CPT conditions is also needed. Sample Call Progress Tones for the subject country can be provided. Such information can be combined to provide a complete CPT solution. For example view: ftp://[url removed, login to view] The programmer shall be knowledgeable of the syntax to program Call Progress Tones within Asterisk, GoIP, etc. The presence of the Cisco in the solution is optional and removal from the solution will be considered. Programmer shall be familiar with the use of Audacity (or other program) to diagnose Call Progress Tones, as described at the following page: [url removed, login to view] Responders should provide a description of their experience. The system is equipped with the following: A2Billing version FreePBX version Running on Linux CentOS ver. 6.5 GoIP sim gateways with Operating System ( 2 units with 32 ports each ) connected to a single 128 port GoIP sim bank. A separate work station has the Session Controller operatiing with a MySQL database. Does each GoIP gateway have its own IP address or are they behind a NAT router? Ans: Each GoIP gateway has its own IP address and there is presently no router in front of the gateways.
Gooday, We are looking for someone in Pretoria (we are in Knysna) to setup a CISCO router (887VA-K9) on the OpenServe Fibre network (Axxess Service Provider, Fixed IP). We will provide a basic configuration but it is for an ADSL setup. Full diagram will be provided, the basics; Fa0 – WAN Port - NAT (Web & Mail Server to DMZ), QoS (Packet Queue - DMZ & LAN, WiFi Packet Drop on Max Upload Bandwdith reached), Fa1 – LAN – Dell Switch (DHCP) - VLAN, Fa2 – DMZ – Dell Switch - VLAN, Fa3 - WiFi - Zyxel Router (VLAN, DHCP, QoS, Max Badwidth, Drop Packets on Cisco) Also Install the 2x switches, 2x routers & Server Brackets into a New Cabinet and neatly cable tie cables inside cabinet according to provided diagram.