VoipSwitch Support Voip Gateway configuration configure termination


Hello, I have a problem with VoipSwitch from [url removed, login to view] or [url removed, login to view]

We need someone who has experience with Voipswitch for support and configuration.

Must have skill in managing Voipswitch

Please explain your experience with Voipswitch in PM.

I have a Voipswitch version [url removed, login to view]

I have 2 terminations in Venezuela, one with the Gateway Grandstream GXW4108, and other with a Zycoo Zx50 - 4FXo ports. Both have the same problem.

I added them as gateway and as gatekeeper respectively.

Im using this router: Tplink TL-ER5120, and also a TPlink TL-WR642G

I can provide you remote access to everything.

If a person connected to my Voipswitch direct SIP with a Voip Phone dials, it works perfect,

but, when a person calls using the VoipBox, it have a problem: you can hear the other party, but the other party can't hear you.

Why is this happening? I tried many many times with many many options and nothing works.

I have this ports forwarding configuration:

1 10000-10200 [url removed, login to view] ALL Enabled

2 9999 [url removed, login to view] ALL Enabled

3 9000-9049 [url removed, login to view] UDP Enabled

4 5060 [url removed, login to view] ALL Enabled

Terminations are well configured, because when connecting with SIP trunks it make calls perfectly. The problem is when going thru the voipBox. It happens with both gateways.

Please write me if could solve this problem, and how much will cost

Please do not post if you dont have experience in voipswith, voip gateways configuration.

Thank you

Skills: Asterisk PBX, Linux, VoIP

See more: gxw4108, voipswitch version, remote support linux, 192 168.1.1, voipswitch, voip termination, voip support, VoIP PBX, sip router, router configure, remote support, remote support it, grandstream pbx, asterisk configuration, need someone configure router, can hear, linux udp configure, linux sip voip, tplink, sip phone configuration, trunks, voip phone pbx, asterisk problem, pbx asterisk support, direct phone calls

Project ID: #4911180

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This task is simple and will take only a while. I have long experience with VoipSwitch software configuration.

$111 USD in 3 days
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6 freelancers are bidding on average $156 for this job


please see PM

$166 USD in 3 days
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Hi, I have experience with VoIPSwitch and SIP troubleshooting. I previously owned one of the first USA SIP carriers starting in 2004 and did a lot of development work on Asterisk and FreeSWITCH. I have recently config More

$150 USD in 2 days
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Hi we are freelance software developers. If you contact us, we can give a quote and we can discuss further details of the project. w w w . so l ve r . i o

$155 USD in 3 days
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hi i can solve your problem asap to check my ability check my profile

$155 USD in 3 days
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Hello. I can do all of the above at the specified time. I am a professional system administrator and my experience is more than 10 years of practice.

$200 USD in 3 days
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