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Unread 03-20-2009, 03:50 AM   #21
boatman
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Make sure your ipkall number is forwarded to voxalotid@voxalot.com, not voxalotid@us.voxalot.com. Strange things can happen when calls coming from outside the Voxalot network are addressed using us.voxalot.com or one of the other regional proxies.

If your public IP address doesn't change frequently you can forward ipkall directly to your PAP2 unit. You would just need to configure your router to forward the SIP ports and RTP port range to your PAP2, then go to your ipkall account and set "SIP Proxy:" to be your public IP address followed by the port number of the target line of your PAP2. Example; 98.99.100.101:5060
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Unread 03-20-2009, 04:12 AM   #22
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These are the only fields that IPKall has to fill in

Account type: SIP

SIP Phone Number: My 6digit voxalot id
SIP Proxy: us.voxalot.com (I tried changing this to just voxalot.com, still same story)
Email Address: my registered email id
Password: my passwd
Seconds to Ring before Hang Up: 300

Do you think it is an issue with IPKall? I tried forwarding from my gizmo account as well, same story. Infact when i forward thru gizmo there is no audio either way.

Thanks again for your time.
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Unread 03-20-2009, 05:24 AM   #23
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It could be ipkall or something else. More tests would help find the problem. If you call me at my SIP address (in that PM I sent you) I have Wireshark running and can check your SDP packet for possible errors.

In order to test your ipkall number you can temporarily forward it to a known working SIP address such as 411@proxy01.sipphone.com, then call your ipkall number and see if there is two way audio.
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Unread 03-20-2009, 07:55 PM   #24
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I tried forwarding to 411 and it seems like it's two way audio. Since i was able to navigate thru the 411 menu with voice. So IPKall seems to be working.
How do i call you at the SIP address? I'm actually operating my pap2 and IP phone from long distance.

Thakns for your help

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Originally Posted by boatman View Post
It could be ipkall or something else. More tests would help find the problem. If you call me at my SIP address (in that PM I sent you) I have Wireshark running and can check your SDP packet for possible errors.

In order to test your ipkall number you can temporarily forward it to a known working SIP address such as 411@proxy01.sipphone.com, then call your ipkall number and see if there is two way audio.
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Unread 03-20-2009, 08:07 PM   #25
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New thing. Now when i dial they sip URI myvoxalotid@voxalot.com or myvoxalotid@us.voxalot.com then again there is only one way audio. Could it be possible that the ISP in india is dropping incoming voice packets or something?
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Unread 03-20-2009, 08:31 PM   #26
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Quote:
Originally Posted by ilvox View Post
How do i call you at the SIP address? I'm actually operating my pap2 and IP phone from long distance.
Put the SIP address into one of the Speed Dial entries, I had suggested Speed Dial 2, then press 2# to make the call. The call can only be done from the phone connected to the PAP2.

You mentioned port 80 on the Voxalot end, what port are you using on the PAP2 end?

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Could it be possible that the ISP in India is dropping incoming voice packets or something?
Yes, if I were an ISP trying to block VoIP I would drop all identifiable VoIP packets.
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Unread 03-25-2009, 07:02 PM   #27
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Hi,

Right now here is the status: all voip-voip calls are 2 way audio. However if someone calls me from their landline on my ipkall or gizmo number (both forwarding to voxalot) then that is only one way audio. Similarly i can dial out other voip numbers and there is 2 way audio. But if i dial out a toll free or other US numbers (using grandcentral) then the audio is just one way i.e i cant hear anytg. Does this mean that it's not that my ISP is blocking voice but a problem at my end? If so, what else to try to resolve the issue?. I will call you on the number you mentioned..Only i'm working remotely on this phone/pap2 and that is making it rather hard. Can you please advise how to remedy the situation? Thanks so much!
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Unread 03-25-2009, 08:07 PM   #28
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If you have not already done so, try the recommendation of carlosalbffgomes given above. If it still doesn't work properly then I would recommend that you try pointing ipkall and Gizmo5 calls directly at your ATA's SIP address, bypassing Voxlot. By doing that you will loose Voxalot's voice mail and other features for inbound calls coming from Gizmo5 and ipkall, but if you gain two-way audio then you will be better off than you are now. If you decide to receive calls directly to the ATA's SIP address you will need to configure the router to forward the SIP ports and RTP port range to the ATA.
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Unread 03-25-2009, 09:27 PM   #29
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Quote:
Originally Posted by ilvox View Post
Hi,

Right now here is the status: all voip-voip calls are 2 way audio. However if someone calls me from their landline on my ipkall or gizmo number (both forwarding to voxalot) then that is only one way audio. Similarly i can dial out other voip numbers and there is 2 way audio. But if i dial out a toll free or other US numbers (using grandcentral) then the audio is just one way i.e i cant hear anytg. Does this mean that it's not that my ISP is blocking voice but a problem at my end? If so, what else to try to resolve the issue?. I will call you on the number you mentioned..Only i'm working remotely on this phone/pap2 and that is making it rather hard. Can you please advise how to remedy the situation? Thanks so much!
If two way audio is working in one situation but fails in an other, than you should be looking at the codecs. Seems like IPKall and or Gizmo5 are unable to establish proper connection due to voice codec misconfiguration. Perhaps you should tell us which voice codecs are enabled how are they configured.
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Unread 03-30-2009, 11:51 AM   #30
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Hi,

Today I made some calls from one voxalot account to another voxalot account via eu.voxalot.com.
I observed that the voxalot server at IP-address 85.17.19.194 not only sent the INVITE packet to the recipient (that's good), but also sent an INVITE packet to the sender (that's bad).
When I use X-Lite the call results in one way audio, when I use Ekiga the call works fine.

Regards,
zonljmv
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