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Asterisk PBX Calls cuts of Calls at 32nd second.

$30-250 USD

Terminado
Publicado hace más de 7 años

$30-250 USD

Pagado a la entrega
We have Class5 soft Switch (Voipswitch) And this switch send calls to asterisk and asterisk makes some changes to CLI prefix etc. and send it to Termination Partner. We can make calls from IOS app. and Access number. there is no issue. But if we call from Android app the call cuts of at 32nd second. There is no issue the calls from IOS and Access number. We need to fix this problem ASAP. Only VOIP and Asterisk experts please.
ID del proyecto: 11649385

Información sobre el proyecto

14 propuestas
Proyecto remoto
Activo hace 7 años

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$135 USD en 1 día
4,9 (42 comentarios)
5,6
5,6
14 freelancers están ofertando un promedio de $183 USD por este trabajo
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This problem seems to be a NAT issue, call is terminated at before 60 seconds, I not an Adroid developer but I cant check you Asterisk nat settings, also defining an stun server to your sip client could fix the issue
$250 USD en 3 días
5,0 (64 comentarios)
6,0
6,0
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Hello dear sir, asterisk server and android phone connecting in same local network or android\server behind NAT? Looks like it some network issue. Please contact me .Thanks.
$200 USD en 1 día
5,0 (27 comentarios)
4,9
4,9
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Hello, i am a voip engineer with more than 6 years of expertise in the field, i have worked with asterisk, freepbx, elastix, freeswitch and the most popular pbx/voip system in the market. i can help you with this task ASAP, if you want to talk regarding this just hit me on chat or add me on skype: montana3601
$100 USD en 1 día
4,8 (16 comentarios)
4,3
4,3
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Hi, i am php developer, with more the 8 years the experiencie with asterisk and voip, i have experiencie with nat problems .
$222 USD en 3 días
5,0 (2 comentarios)
2,5
2,5
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You may not want to hire me, but I would like to help you anyway. Please check these settings. If you have calls cutting off after 32 seconds, then you most likely have a SIP dialogue problem, where something hasn't been acknowledged properly for Android. There's a round trip timer timer called the T1 timer (normally 500ms) and the timeout is after 64 intervals. Each interval is 500ms, so 64 * 500ms = 32 seconds. Find that setting and ensure it is set properly. Hope this helps, Dan
$277 USD en 2 días
0,0 (0 comentarios)
0,0
0,0
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Grreting Codec, STUIN and external absence of domain can cause such problem. Im not are guru, but i hade similar issue
$155 USD en 3 días
0,0 (0 comentarios)
0,0
0,0
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A proposal has not yet been provided
$166 USD en 7 días
0,0 (0 comentarios)
0,0
0,0
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Hi, I'm Felipe Bonilla, I currently work in Elastix Mexico y Centro America, and I'm a expert in VoIP PBX based on asterisk. I know about this problems and, incredibly this is a simple and common problem, and of course, I can help you.
$70 USD en 2 días
0,0 (0 comentarios)
0,0
0,0
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We have lots of experience. I hope we will become your support team. We are a very cost effective provider. We are a Toronto base company with excellent reputation for VOIP support. Remus
$250 USD en 3 días
0,0 (0 comentarios)
0,0
0,0
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We are well experienced in setting up call center dialers and have successfully implemented over 300 setups ranging from 20 seats to 2000 seats. We can setup any type of dialers including inbound, outbound, blended, click to call types etc. We can also do branding and customise dialersas per the customer's requirement. We have in depth knowledge of call center hardware like telephony cards, voip gateways, gsm/cdma gateways, fxo/fxs cards and gateways etc. We also deal in IP Phones, Call Center Headsets and dial pads. In brief, we can do anything in setting up a call center.
$211 USD en 3 días
0,0 (0 comentarios)
0,0
0,0

Sobre este cliente

Bandera de CANADA
ancaster, Canada
5,0
12
Forma de pago verificada
Miembro desde nov 11, 2011

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