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07-26-2009, 06:24 PM | #1 |
Junior Member
Join Date: Jul 2009
Posts: 22
Thanks: 6 Thanked 3 Times in 2 Posts |
I made all the suggested changes above except I did not make any changes to line 1 as it's working fine directly connected to voip.ms.
I set NAT=no at voip.ms and it didn't make a difference. As a test, I configured my router to put my ATA on the DMZ. Surprisingly, that didn't work either and still only get audio one way. My thinking is that if the ATA can't even work on the DMZ with no firewall filtering, then I have config issues on the ATA itself and *maybe* voip.ms. Mind you, connecting directly to voip.ms on line 1 works fine. I've documented all of my changes and nothing is working. |
07-26-2009, 06:42 PM | #2 |
Senior Member
Join Date: Jul 2007
Location: Oregon, USA
Posts: 365
Thanks: 17 Thanked 77 Times in 64 Posts |
Try pointing your voip.ms SIP URI forwarding to your number at voxalot.com, not us.voxalot.com, 123456@voxalot.com, not 123456@us.voxalot.com.
I don't have a DID at my voip.ms account, but if there's a way I can test forwarding from voip.ms to my Voxalot account, let me know and I'll give it a try. |
07-26-2009, 06:46 PM | #3 | |
Junior Member
Join Date: Jul 2009
Posts: 22
Thanks: 6 Thanked 3 Times in 2 Posts |
Quote:
I'll also make the SIP URI changes that you are suggesting and test again. |
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07-26-2009, 09:25 PM | #4 |
Senior Member
Join Date: Jul 2007
Location: Oregon, USA
Posts: 365
Thanks: 17 Thanked 77 Times in 64 Posts |
The Wireshark packet capture points to a problem with voip.ms. The events happened as follows:
Call comes from User-Agent: VoIPMS/SERAST with Connection Information (c): IN IP4 24.102.60.67 I answered the call on a PAP2. About 0.11 sec. later VoIPMS/SERAST sent SIP re-invite with Connection Information (c): IN IP4 67.205.74.164. PAP2 begins sending RTP to 67.205.74.164 and receiving RTP from 24.102.60.67. For some reason the RTP outbound from PAP2 to 67.205.74.164 does not reach the party calling on the DID, therefore calling party hears nothing. You should contact support at voip.ms. I can send you the packet capture if needed. |
07-26-2009, 09:55 PM | #5 |
Junior Member
Join Date: Jul 2009
Posts: 22
Thanks: 6 Thanked 3 Times in 2 Posts |
boatman, I wish there was more than "Thank You" for the amount of time and effort you have put into trying to resolve my problem. Admittedly, I was prepared to give up a long time ago.
The good news is that we know what's occurring. The bad news is that the problem still exists and is largely out of our control. I hope that voip.ms is willing to help. I will likely need that packet capture when I do this all over again with voip.ms. |
07-27-2009, 01:36 AM | #6 |
Junior Member
Join Date: Jul 2009
Posts: 22
Thanks: 6 Thanked 3 Times in 2 Posts |
Great news!!!
After sending an email to voip.ms support about this issue, they just responded letting me know that they made a routing change on my account. I did a quick test and it worked. Audio is working both ways now. I will need to monitor for the next few days but it looks like I'm back in the mix with VoXalot. Considering the support that I've gotten here and at voip.ms (even over the weekend), I know that I've picked the right combination to provide me most of my phone needs. And it didn't cost me a red cent but my sincere gratitude. |
07-31-2009, 10:55 PM | #7 |
Junior Member
Join Date: Jul 2009
Posts: 22
Thanks: 6 Thanked 3 Times in 2 Posts |
As a follow-up, I asked voip.ms what they did to resolve my problem and their response was that they "...prevented the RTP from being re-invited."
While I don't know what that means, I'm hoping the info will be of benefit to boatman and any others going forward who have this problem. Needless to say, I've had no issues since the change was made by voip.ms a several days ago. |
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