mh1983 wrote:Hi,
I have FPL with the unlock key and has been working great since we purchased. I'm using the HT701 ATA, configured with the settings in the guide in these forums, and with the latest firmware. I'm eventually going to port a number from MagicJack, but for now am forwarding the calls from that number to the FPL number.
Today (and probably a handful of times in the past), I joined a conference call which dialed me at the MJ number, which then forwarded to my FPL. The call was going smoothly but dropped at the 90 min mark.
When I used MJ as the primary service, I noted drops around a similar timeframe. Is it related? I'm curious if I had the conference call the FPL number directly if the call would've continued uninterrupted.
Still, I'm wondering if there's anything I should verify on the ATA or router side. I don't have any port forwarding set up nor DMZ.
Thanks in advance!
mh1983 wrote:Hi,
...
Today (and probably a handful of times in the past), I joined a conference call which dialed me at the MJ number, which then forwarded to my FPL. The call was going smoothly but dropped at the 90 min mark.
When I used MJ as the primary service, I noted drops around a similar timeframe. Is it related? I'm curious if I had the conference call the FPL number directly if the call would've continued uninterrupted.
Still, I'm wondering if there's anything I should verify on the ATA or router side. I don't have any port forwarding set up nor DMZ.
Thanks in advance!
Bing Kol wrote:mh1983 wrote:Hi,
...
Today (and probably a handful of times in the past), I joined a conference call which dialed me at the MJ number, which then forwarded to my FPL. The call was going smoothly but dropped at the 90 min mark.
When I used MJ as the primary service, I noted drops around a similar timeframe. Is it related? I'm curious if I had the conference call the FPL number directly if the call would've continued uninterrupted.
Still, I'm wondering if there's anything I should verify on the ATA or router side. I don't have any port forwarding set up nor DMZ.
Thanks in advance!
I can tell you that when I was actively using MJ, every call would be terminated after 90 minutes.
I can also tell you that we can talk for more than 2 hours uninterrupted using my FPL line .
Mango wrote:T
Please try logging out and closing the desktop application.
Verify that your Register Expiration is 60 and let us know what it was if it was not.
Finally, try to change your Primary SIP Server from voip.freephoneline.ca to voip2.freephoneline.ca, or vice versa.
Let us know if that works.
f_laplante wrote:Mango wrote:Since my ATA is before my router, i guest there should not be problem with the router...
other idea?
Mango wrote:Hello; thank you for the PM. I'm sorry I don't know what's wrong. We can try some general troubleshooting.
Earlier I told you to use voip2. Now please set your Primary SIP Server to voip4.freephoneline.ca:6060 X
Please set Local SIP Port to 26913 X
Please set Use Random Port to No, if it is not that way already. OK
Theoretically, a firmware upgrade to your cable modem, applied by your ISP, could have caused this problem. If the above works, it will support that theory.
Please double check the following settings are correct:
Account Active: Yes OK
SIP Transport: UDP OK
NAT Traversal: Keep-Alive OK
DNS Mode: A Record OK
SIP Registration: Yes OK
If you still have the problem, try to set your Primary SIP Server to 162.213.111.21:6060 so that you may test for DNS-related issues.
If you still have problems after trying all this, please check your Grandstream ATA's Status tab and post the WAN IP Address here. You can remove the last octet to protect your privacy.
wolaiye wrote:1. incoming call is fine when we just change our internet provider from Rogers to Bell fiber (CIK FNNP PROVIDER)
2. But I can't make a call outbound, I try to call my cellphone ,cellphone is ringing but can't hear from the VOIP phone .Switching from voip2.freephoneline.ca to voip.freephoneline.ca makes no difference.
3.Sometimes it is going fine. when I tried outgoing calls, it took about 1 minute until the other side can receive my call,or can't get through (took 1 minute until I canl hear the ring tone for the otherside as well ). I've tried to reset the phone adapter also the router, but nothing worked. no problems for incoming calls.
4.We followed CISCO-SPA112 Setup guide,nothing changed.
5.voip ADAPTER: Cisco SPA112; network in Toronto:CIK (bell)FTTN; MODERM: CIK TELECOM 's DSL MORDEM: HG-A800 which is multiuse for ROUTER,VOIP ATA
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