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01-06-2008, 11:21 AM | #1 |
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Call drops after 21 seconds
Hi all,
I am a Premium subscriber and I am using Grandstream HT488 device with an FXS and another FXO account. Both accounts are configured to register with two different Voxalot accounts. FXS account uses 2060 port, whereas FXO account uses 3060 port. I have a VoipDiscount account configured on the FXS Voxalot account with Symmetric NAT Handling set as 'No' and Optimize Audio Path set as 'Yes'. The *600 echo test is successful. The same configuration worked properly before Voxalot launched VoxPremium. Then I shifted to some other VSP. Now I purchased VoxPremium and setup the same exact configuration. The problem is that calls going through VoipDiscount go silent after exactly 21 seconds and they are disconnected (as I see from my VoipDiscount call log, they get disconnected after exactly 21 seconds). I have performed the following steps: 1. It works properly if I connect to the Voxalot account from EyeBeam (X-Lite). 2. It also works properly if I directly configure the VoipDiscount account on my HT488 ATA. 3. I also tried setting the Symmetric NAT Handling to 'Yes' and Optimize Audio Path to 'No'. 4. *600 echo test lasts for more than 21 seconds and has no problems. 5. *1800 and other toll free numbers that go through Voxalot itself without using any VSP's also don't have any problems. 6. I tried setting the "Register" option for the VoipDiscount account to 'Yes' and 'No' both. But still the problem persists. 7. I changed the preferred cluster server to eu.voxalot.com and similarly changed the connection server on my ATA. The problem still persists. 8. I don't have any other VSP's configured in Voxalot in order to test calls going through them. Please let me know what else could be the problem. If it seems like a re-INVITE issue, then does VoipDiscount send a re-INVITE after 20 seconds? Also, it still seems like a mystery to me as the same exact configuration worked with Voxalot before the launch of VoxPremium. I can provide screenshots of my ATA configuration as well as Voxalot configuration settings if required. Regards, Pradeep |
01-06-2008, 06:07 PM | #2 |
Join Date: Sep 2006
Location: Toronto, Canada
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I did set up an HT-488 a few months ago, and it has been working reliably.
I have a Voxalot account on the FXS port using Poivy as an outgoing PSTN provider. The FXO SIP account is using another VSP. I do not have access to the settings as this ATA is currently 5000 miles away. Some suggestions : 1) Update to the latest firmware (The FXO port did not work properly with earlier firmware) 2) Use the default SIP ports 5060 and 5062 3) Use different RTP ports (Eg 5004, 5006) for the FXS and FXO ports 4) Use a STUN server (Eg. stun.voxalot.com.au:3478) 5) If you are behind a router, it might help to forward the SIP ports (5060-5063) and the RTP ports (5004-5007) in particular. You might want to try setting up the FXS port by itself first (with the FXO port disabled) until you trace the cause of the 21 second call-drop problem. |
01-07-2008, 04:09 AM | #3 |
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There is a known bug in asterisk that causes similar problems. However a message trace is necessary to confirm if this is the source of your problem. The bug is unusual in that some phones will work and others will not under the same conditions but the bug is not in the phone but rather in asterisk. I will not go into an explanation here as it is rather long and it is well documented elsewhere.
Refer to the following -- In sipbroker forum see Another misrouted ACK In bugs.digium.com see 0011230, 0011326 and 0011545 |
01-07-2008, 04:45 PM | #4 |
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I'm also a Premium subscriber and experiencing the same issues, but only for incoming calls from every SIP I have registered.
When someone calls, my TC-300 rings and the call goes on perfectly until it reaches 21 seconds, then the caller gets disconnected and my phone still shows the call as active until I have to hang up manually. |
01-08-2008, 01:36 AM | #5 |
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Problem still persists
I performed all the steps as Kurun said. Upgraded the firmware, used STUN server, setup port forwarding in my ISP's router and also disabled the FXO port & tested only with the FXS port configuration. Still the same problem persists. Though the only thing is that my ISP blocks regular SIP & RTP ports. So, I am not able to use any ports in the 5000 range. I have set RTP ports as 2004 (FXS) & 3008 (FXO).
The HT-488 works well if I set it up directly with any other VSP instead of Voxalot. But the surprising thing is that I used the same exact configuration with same HT-488 settings, same VSP's configured in Voxalot, everything same before Voxalot launched VoxPremium. Everything worked well then. I am not sure what that problem could be now. I am willing to give as much as possible to get this problem resolved. Any thoughts anyone? Martin? |
01-11-2008, 10:04 AM | #6 |
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ive got the same problem with inbound, the connection lasts 19 sec and then breaks up. echotest and calling out works as usual...
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01-11-2008, 10:28 AM | #7 |
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Hi victor & haifischjunge,
Can you please mention which hardware ATA adapter you are using? Also mention the VSPs your are registering against where you are facing these problems. I am facing problems with Betamax VSPs registered within Voxalot for outgoing calls. Regards, Pradeep |
01-11-2008, 11:04 AM | #8 | |
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Quote:
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07-02-2008, 11:36 PM | #9 | |
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Quote:
discovered it's not ATA, softphone, or IP phone, my guess routing is somewhere the blame, and this also limits the use of Voxalot, because you can't voip from the laptop this way, (WiFi) |
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07-05-2008, 11:04 AM | #10 |
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I now know it has nothing to do with the ATA, but somewhere in the routing, or protocol, because an older modem/router don't have the problem, but also less internet security..
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