Fix a problem with Asterisk/Freepbx and an Axis IP Speaker C3003-E.

$30-250 CAD

Закрито

Опублікований

over 7 years ago

$30-250 CAD

Оплачується при отриманні

We have a PBX running on Asterisk and we have around 60 extensions configured. We use this in order to do remote voice alerts in softphones running on servers and some Axis IP speakers using SIP. We have many Axis speakers working and for this we had forwarded the necessary ports for them to work and we use the public IP address. All of the extensions are outside of the network our server is running. For the softphone we use Jitsi and we have preamplified speakers connected in the speaker plug of our sound card. So when from our office we dial an extension, we have Jitsi auto answer the call and then we speak and call talk and the sound is heard. When we use Jitsi, we use a VPN connexion (OpenVPN) so we use the VPN (local) ip address. But when we work 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 works :(

Asterisk PBX

Network Administration

ID проекту: 12283871

Про проект

6 пропозицій(-ї)

Дистанційний проект

Активність 7 yrs ago

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@ngxperts

hi, no sound is almost always because of NAT issues, kindly let me know if you can give me access to the asterisk server and also control the NAT/Firewall settings at that particular client . regards,

$300 CAD за 5 дні(-в)

5,0

(35 відгуки(-ів))

5,7

@ast3r1sk

Hello dear sir, i can try to fix your issue with no sound on Axis Speaker. Usually it some network misconfiguration. Please contact me if you interested in. Thanks.

$222 CAD за 3 дні(-в)

5,0

(27 відгуки(-ів))

4,9

@vladimirumnov

Hi. You need forward diffreent rtp range, not 10-20k. This range from asterisk side. From axis side you should take a range from axis configuration page or from manual. Also you need to disable sip alg and enable nat on extension on asterisk. Sometimes some dumb devices not support nat at all. If so - you will have to use router with openvpn like for jitsi.

$100 CAD за 0 день

5,0

(27 відгуки(-ів))

4,9

@gilbertarias

Hello, i am a voip engineer with more than 7 years of expertise in the field, i work on a daily basis with asterisk and freeswitch based systems, i can help you to troubleshoot this issue and solve it. if you interested to talk regarding this just hit me on chat or add me on skype: montana3601.

$150 CAD за 5 дні(-в)