no ACK from Voxalot

green · 11-29-2007, 03:51 PM

Hello
I have my incoming calls disconnected after a few seconds.
I checked my logs (Asterisk) - I'm sending 200 OK number of times and have no ACK back from voxalot.

Can anybody look into this problem? Thanks.

loraolo · 11-30-2007, 09:53 AM

Hi,

a similar behaviour happens to me also.
My voip scenario is Grandstream HT488 (ATA bridge) connected to two different voxalot account for fxs and fxo ports. The HT device has been configured with PSTN to VOIP forwarding calls by using voip number *010<my voxalot account>.

The result is that all the incoming PSTN calls are forwarded immediatly on voxalot termination at most without any problem but sometimes in the same day they the calls drops in a few seconds/minutes. After investigating I found that this problem is mainly related to internet jitter (variation on delay on internet communication) and codecs.

About Jitter, since I don't know if voxalot provides a tool to simulate SIP calls to monitor the communication quality over time, my actions were oriented on codecs. So, setting G729 as preferred codec on all my voip device - since I'm in Europe and use european VSP - the stability was significantly improved.

I hope this can help
cheers
Paolo

11-29-2007, 03:51 PM	#1
green Senior Member Join Date: Jul 2007 Posts: 143 Thanks: 13 Thanked 20 Times in 19 Posts	no ACK from Voxalot Hello I have my incoming calls disconnected after a few seconds. I checked my logs (Asterisk) - I'm sending 200 OK number of times and have no ACK back from voxalot. Can anybody look into this problem? Thanks.

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11-30-2007, 09:53 AM	#2
loraolo Member Join Date: Jul 2007 Posts: 36 Thanks: 0 Thanked 6 Times in 3 Posts	Hi, a similar behaviour happens to me also. My voip scenario is Grandstream HT488 (ATA bridge) connected to two different voxalot account for fxs and fxo ports. The HT device has been configured with PSTN to VOIP forwarding calls by using voip number *010<my voxalot account>. The result is that all the incoming PSTN calls are forwarded immediatly on voxalot termination at most without any problem but sometimes in the same day they the calls drops in a few seconds/minutes. After investigating I found that this problem is mainly related to internet jitter (variation on delay on internet communication) and codecs. About Jitter, since I don't know if voxalot provides a tool to simulate SIP calls to monitor the communication quality over time, my actions were oriented on codecs. So, setting G729 as preferred codec on all my voip device - since I'm in Europe and use european VSP - the stability was significantly improved. I hope this can help cheers Paolo