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Asterisk & webrtc based project. - open to bidding

$30-250 USD

Cancelado
Publicado há quase 10 anos

$30-250 USD

Pago na entrega
We need support from an Asterisk expert on resolving issues. We are thinking on buy official diggium support, but first we want to try with an expert freelancer. Now the problem we have is a one way audio comunication in most of inbound calls. We'll pay for resolving this issue or for giving info on what's going on.
ID do Projeto: 5912220

Sobre o projeto

10 propostas
Projeto remoto
Ativo há 10 anos

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10 freelancers are bidding on average $257 USD for this job
Avatar do Usuário
Lets discuss to check this issue and fix it. One way audio are result of NAT, etc. So, lets check the setup and fix.
$244 USD em 3 dias
4,9 (43 avaliações)
5,7
5,7
Avatar do Usuário
Hello, I can do the job. If you want we can discuss the project details via PM. I also speak spanish. Be free to contact me if you have any questions or comments about it. Thanks in advance. -- Adolfo
$250 USD em 3 dias
5,0 (12 avaliações)
4,1
4,1
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A proposal has not yet been provided
$222 USD em 3 dias
5,0 (2 avaliações)
1,9
1,9
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I have a great deal of experience with Asterisk, Freeswitch nad Webrtc. Friend of mine acutally told me about this bid and that made me join freelancer. I would like to know if you are interested. I have no prior experience here in Freelance I should mention. By the way, Webrtc being an experimental product and not mature enough.I would recommend oversip as alternative. Thanks.
$155 USD em 3 dias
0,0 (0 avaliações)
0,0
0,0
Avatar do Usuário
I have a lot of skills with voip service like, SIP Servers, IP-PBX, UA and SoftSwitches. Please look my profile and you can see it.
$255 USD em 3 dias
0,0 (0 avaliações)
0,0
0,0
Avatar do Usuário
Hello, I've some years (+11 years) of experience on the Linux administration field, I'm new on Freelancer and maybe my profile isn't very complete... But returning to your issue, some times the voice in one way (instead of two) is caused by firewall or routing problems. Asterisk uses SIP with UDP (port 5060) transport for the session initiation and so on for the voice transport it uses RTP fom port 10000 to 20000 (in most cases). When a problem as yours occurs, the first reason can be because the above described port configuration isn't comunicating fine. Other possibility is the use of incorrect codec or voip client configuration in one of the ends. Analyzing the environment is possible find the gap and solve it. Have a nice day. Thank you, regards.
$277 USD em 3 dias
0,0 (0 avaliações)
0,0
0,0
Avatar do Usuário
This can be as simple as firewall issue of can be internal context, we can help u with this issue and we guarantee our work full money back guarantee if not happy.
$222 USD em 1 dia
0,0 (0 avaliações)
0,0
0,0
Avatar do Usuário
A proposal has not yet been provided
$222 USD em 1 dia
0,0 (0 avaliações)
0,0
0,0

Sobre o cliente

Bandeira do(a) SPAIN
Barcelona, Spain
0,0
0
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Membro desde mar. 31, 2012

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