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02-26-2009, 07:14 AM | #11 | |
Join Date: Jul 2007
Location: Toronto, Canada
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Quote:
Does the problem persist if instead of eu.voxalot.com you use proxy01.eu1.voxalot.com |
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02-26-2009, 01:07 PM | #12 | |
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changing the proxy from us.voxalot.com to eu.voxalot.com seems to work...
Quote:
I always used us.voxalot.com, as I'm in Canada. Should I only use eu.voxalot.com from now on? I will continue testing later today, but it looks, from the 10 calls I did after the change, that this change is the one that caused the one way audio in my case. Is that change required? Could it be rolled back to see whether everybody sees the problem fixed? Thanks, DoctorH |
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02-26-2009, 01:13 PM | #13 |
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us.voxalot.com load balances between 2 proxies. It is possible that one of these proxies had an RTP proxy problem (which would explain your 50% success/failure rate). As mentioned in another thread I have restarted all RTP proxies.
As such, could you please set your ATA back to us.voxalot.com and let me know if your problem still exists? Thanks
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Martin Please post support questions on the forum. Do not send PMs unless requested. |
02-26-2009, 01:39 PM | #14 | |
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Thanks for your help Martin. |
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03-19-2009, 05:47 PM | #15 |
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one way audio
hello, somebody pls help me..Here is the exact nature of the problem. I'm quite new to this voip so not sure how to fix it.
I have voxalot set up on my linksys pap2 with us.voxalot.com server. If i directly dial out from here, or if someone directly calls my sip URI, then the audio is two way and both parties can hear each other. However i have a IPKall number that forwards call to my voxalot account and when someone calls me on that number, (or another gizmo number that also forwards to my voxalot), then I cannot hear anything at all. However, they are able to hear me. Please help me. I rely a lot on this service and would definitely appreciate a resolution or a reason as to why this could be happening. It has been working perfectly until about 3 days ago. Pls let me know. Thank you so much in advance |
03-19-2009, 06:10 PM | #16 |
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Use of the following settings may solve the problem. I expect you already have most of these settings.
(under SIP tab) Handle_VIA_received: yes Handle_VIA_rport: yes Insert_VIA_received: yes Insert_VIA_rport: yes Substitute_VIA_Addr: yes Send_Resp_To_Src_Port: yes STUN_Enable: yes STUN_Test_Enable: yes STUN_Server: stun.voxalot.com:3478 (or any STUN server such as 'stun01.sipphone.com:3478' or 'stun.sipgate.net:10000') NAT_Keep_Alive_Intvl: 179 (The NAT_Keep_Alive_Intvl value 179 was chosen to be less than the NAT session expiry time of most routers.) (under Line_1 and Line_2 (or PSTN_Line) tabs) NAT_Mapping_Enable: yes NAT_Keep_Alive_Enable: yes NAT_Keep_Alive_Msg: (this should be blank in order to send dummy packets) NAT_Keep_Alive_Dest: $PROXY Register Expires: 3600 ---------- If you prefer not to use a STUN server then do the following: 1. Forward the SIP ports and the RTP port range. 2. Set STUN Enable: no 3. Set NAT_Keep_Alive_Enable: no 4. In order for the ATA/phone to know it's public IP address, make sure the ATA/phone is registered with at least one SIP registrar, or enter your public IP address into the appropriate setting in the ATA (in Linksys ATAs this setting is EXT_IP). ---------- Make sure that you have filled in 'Outbound Proxy:' the same as you have for 'Proxy:' including the proxy's port number. Also confirm that 'Use Outbound Proxy:' and 'Use OB Proxy In Dialog:' are both 'yes'. |
03-19-2009, 06:13 PM | #17 |
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Thank You very much, i will try this out right away and let you know. Just one question, my current settings were working perfectly until 3 days back. What could have changed, that has caused me to change my settings? Thanks so much for your help again
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03-19-2009, 06:39 PM | #18 |
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Location: Oregon, USA
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I don't know why your potentially mis-configured SIP phone worked some times and not other times. There are measures that can be taken on Voxalot's end that can compensate for some mis-configurations. Perhaps Voxalot was experimenting with some such configurations on the days your SIP unit worked/failed.
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03-19-2009, 11:16 PM | #19 |
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I configured as per instructions here Setting up Linksys PAP2 - Voxalot FAQ.
The NAT settings mentioned above are different from what you have mentioned in the post. Should i change and try?. Which is the right setting? Thanks so much for your help. Sorry i'm so new to this that i'm just overwhelmed with all the jargon etc. What i can't seem to understand is that if i directly dial out a number (like a gizmo 1747xxxxxxx) number from my voxalot phone, it works and i get 2-way audio. Similarly someone directly calling my SIP URI can hear me and i can hear them. Only when the call is directed thru IPKall, there is only one way audio, i.e the caller can hear me but i cannot hear them. Pls advise Thanks again. |
03-20-2009, 12:15 AM | #20 |
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Hi,
i tried everything u've mentioned, still same story. The caller can hear me but i can't hear a thing. However if they directly call me at myvoxalotid@us.voxalot.com, then we both can hear each other. BUt if they they dial my ipkall number, i cannot hear the caller. Please help me. I'm really in a rush to get this working. Thanks and i so appreciate your time and effort to help me. |
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