Cisco SPA112
TP-Link TD-W8960N
Technicolor DCM475
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to Webslinger
Re: [Cable] Teksavvy and FreePhoneLine.ca not working togethersaid by Webslinger:I use Freephoneline without issue with Teksavvy.
The problems discussed have nothing to do with Teksavvy.
They have everything to do with ATAs and routers not being setup properly.
1) A number of Linksys ATA devices require you to port forward RTP Port Min and RTP Port Max (typically listed in the "SIP" tab of your ATA device) as a UDP port range in your router to your ATA device.
2) Additionally, the SIP port listed in the "Line 1" tab of your ATA device must also be port forwarded (UDP) to your ATA device from your router.
3) RTP packet size needs to be set to 0.020 on the "SIP" tab of the ATA device.
4) These values may also need to be set in the ATA device:
a)Ring Tone Frequency: 52
b)Ring Tone Voltage: 90
Those experiencing issues would be far better served (unless your internet connection and speedtests are horrible) to be posting at freephoneline's (Fogo's) forum instead.
Even with SPA122 plugged straight on the modem, now it doesn't work. Only way I can make this work is SPA112 straight on the modem. I did of course program all the setup details as you said (same thing that freephoneline asks on their website) in SPA112 and SPA122. Again, teksavvy cable in Vidéotron land... |
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said by julienvf 
Even with SPA122 plugged straight on the modem, now it doesn't work.
Some modems have routers built in, and those modems require you to port forward too.quote:
I did of course program all the setup details as you said (same thing that freephoneline asks on their website) in SPA112 and SPA122. Again, teksavvy cable in Vidéotron land...
Freephoneline doesn't list configurations for specific ATA devices on their website, and in the configuration list they don't mention to port forward. In their forums, they have configs listed--but not on the website. If one ATA device is working for you and the other isn't that has nothing to do with Teksavvy. |
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Cisco SPA112
TP-Link TD-W8960N
Technicolor DCM475
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said by Webslinger:said by julienvf
Even with SPA122 plugged straight on the modem, now it doesn't work.
Some modems have routers built in, and those modems require you to port forward too.quote:
I did of course program all the setup details as you said (same thing that freephoneline asks on their website) in SPA112 and SPA122. Again, teksavvy cable in Vidéotron land...
Freephoneline doesn't list configurations for specific ATA devices on their website, and in the configuration list they don't mention to port forward. In their forums, they have configs listed--but not on the website. If one ATA device is working for you and the other isn't that has nothing to do with Teksavvy. Not to blame Tek for that... It used to work fine before the December outtage then on & off for the holidays and for about 2 weeks, when everyone got their issues solved, bam! No incoming calls... And Fongo/Freephoneline is now an "email only" support so my client took Tektalk premium. |
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noemails
Anon
2013-Jan-31 7:03 pm
this is where small isps lost the path they used to travel....how long does this post have to go on...jesus christ tek and freephonline you already have ridicoulous queues...assign an experienced agent to each other work with the firmware and software and make it work this dickering back and forth is stupid. between the two of you all the hardware should be covered off and certainly all the software...
seriously if teksavvy doesnt have someone who knows how to make this work on the dsl/cable they offer something big is wrong..order the service and make it work....the fpl guys are gonna take clients away from you if you dont fix this soon and you know already your losing more then you know what to do with |
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to julienvf
said by julienvf:Not to blame Tek for that... It used to work fine before the December outtage then on & off for the holidays and for about 2 weeks, when everyone got their issues solved, bam! No incoming calls... VoIP makes use of random RTP ports. If the RTP range isn't forwarded (as UDP ports in the router to the ATA device) properly but a few are or only one out of the entire range is forwarded properly, then it's certainly plausible that the service may work fine on a specific ATA device at a given time--and then randomly not work at all. Consequently, I can't view what you're describing as being anything other than user error (or a faulty ATA device). It's not as though Teksavvy or Videotron is filtering UDP ports. If Videotron is filtering ports used by VOiP, then's that pretty brutal (but I sincerely doubt that's happening). Rogers certainly isn't. Everything in this thread points to improperly configured ATA devices and routers. |
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| Webslinger |
to noemails
said by noemails :this is where small isps lost the path they used to travel....how long does this post have to go on... It's a thread--not a post. jesus christ tek and freephonline you already have ridicoulous queues...assign an experienced agent to each other work with the firmware and software Teksavvy should not have to provide tech support for third party VOiP services, third party routers, and third party ATA devices. That's ridiculous. Freephoneline doesn't support any ATA device that's not bought directly through them (Fongo), which is somewhat unfair, I suppose (but understandable from a business perspective). seriously if teksavvy doesnt have someone who knows how to make this work on the dsl/cable they offer something big is wrong..order the service and make it work.... Unh, what? lol the fpl guys are gonna take clients away from you  if you dont fix this soon and you know already your losing more then you know what to do with
Can you speak English? Look, you either have internet service or you don't. Unless the ISP is filtering ports (and that's not happening here), the problem is on the other end. |
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Cisco SPA112
TP-Link TD-W8960N
Technicolor DCM475
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to Webslinger
said by Webslinger:said by julienvf:Not to blame Tek for that... It used to work fine before the December outtage then on & off for the holidays and for about 2 weeks, when everyone got their issues solved, bam! No incoming calls... VoIP makes use of random RTP ports. If the RTP range isn't forwarded (as UDP ports in the router to the ATA device) properly but a few are or only one out of the entire range is forwarded properly, then it's certainly plausible that the service may work fine on a specific ATA device at a given time--and then randomly not work at all. Consequently, I can't view what you're describing as being anything other than user error (or a faulty ATA device). It's not as though Teksavvy or Videotron is filtering UDP ports. If Videotron is filtering ports used by VOiP, then's that pretty brutal (but I sincerely doubt that's happening). Rogers certainly isn't. Everything in this thread points to improperly configured ATA devices and routers. That is of course if the ATA is behind a router which failed even if plugged straight to the modem in DHCP... It registered and I wasn't even able to receive calls when plugged straight to the modem without any router or NAT nor anything blocking traffic. Problem for sure was elsewhere rather than an improper port forwarding. I just find it amazing that the same ATA + router worked #1 on Distributel cable then when brought to my customer's house on Tek cable, had that problem. Traffic was indeed routed differently. |
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said by julienvf:That is of course if the ATA is behind a router which failed even if plugged straight to the modem in DHCP... It registered and I wasn't even able to receive calls when plugged straight to the modem without any router or NAT nor anything blocking traffic. Problem for sure was elsewhere rather than an improper port forwarding. I just find it amazing that the same ATA + router worked #1 on Distributel cable then when brought to my customer's house on Tek cable, had that problem. Traffic was indeed routed differently.
The problem can also occur if the ATA is behind a modem that has a built in router. Also, if I take my ATA over to someone else's house and plug straight into another modem, the ATA won't work unless I make changes to the WAN side of the ATA device (I often manually make changes). Also ensure that NAT mapping is enabled in addition to NAT Keep Alive in the ATA device. By the way, I just tried something, testing different codecs. When I use the G711u codec (use preferred codec only), my phone rings fine. When I use G711a (use preferred codec only), calls go straight to voice mail. Might be something for you to test (different codecs). This is also very important for calls: 1. RTP packet size needs to be set to 0.020 on the "SIP" tab of the ATA device. (affects phone call quality) 2. These values also need to be set in the ATA device: a)Ring Tone Frequency: 52 b)Ring Tone Voltage: 90 But you said you already did that. Without being there to look at your ATA settings, troubleshooting is pretty difficult. I'm positive Teksavvy doesn't filter traffic. If Videotron does, then that's pretty bad (I have no doubt it's possible, but I still have a hard time believing they are doing that on purpose to mess up VoIP services). Anyway, good luck. |
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Cisco SPA112
TP-Link TD-W8960N
Technicolor DCM475
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to marksy
The weirdest thing: as of today, the spa122 works fine with either freephoneline and tektalk. Must have been some router on the teksavvy side since I wasn't able to run tektalk properly even plugged straight on the modem but that's past story! It works! |
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said by julienvf:The weirdest thing: as of today, the spa122 works fine with either freephoneline and tektalk. Must have been some router on the teksavvy side since I wasn't able to run tektalk properly even plugged straight on the modem but that's past story! It works!
I find that unlikely. I just read on the Fongo forums that some of their servers were blocking accounts based on registration timeout settings. IPs were being temp blocked based on multiple registration attempts within a short timeframe (considered abuse to them), and calls were going straight to voicemail (despite the ATA indicating being registered). Registration Expires: 180 to 3600 They're now saying that setting should really be 3600. If you're unplugging and reconnecting the ATA for testing, the reattempts could have caused a temp IP block. Someone from Fongo contacted someone I know to tell him server-side changes were being looked into. Regardless, I'm glad it's working for you. |
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fefrie
join:2012-08-17
Vancouver, BC |
to marksy
I followed ALL of webslineracs suggestions as well as this link » voipfan.net/ATAs/PAP2_fr ··· line.phpand the combination of the two seemed to work. I did a DMZ and port forwaded everything and it seems to work properly now. I do have issues when calling my V-MS number from my FPL number. It seems like IP phones don't like to talk to each other. |
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People on freephoneline should also note that your required SIP settings may have changed recently: » support.freephoneline.ca ··· dentials |
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| Webslinger |
to fefrie
You shouldn't need to. That link you posted is outdated. Registry expires needs to be 3600, or you face having your account temporarily locked out. » support.freephoneline.ca ··· dentials |
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fefrie
join:2012-08-17
Vancouver, BC |
fefrie
Member
2013-Apr-20 9:25 pm
Hi Dan, PM Sent
you can call me ANYTIME.
I'm in vancouver.
One weird thing I have so far is that when I call in with my cell phone, my cell phone goes straight to voicemail. Which is really weird.
EDIT: You MUST Port Range forward all 3 5060 6060 13000 ports. I just added those and now it seems to accept cell phone and regular phone calls. before I port ranged these ones, cell phone calls went straight to voicemail. Now it dials normally |
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Technicolor TC4350
Asus RT-AC56
Grandstream HandyTone 702/704
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to Xuhum
said by Xuhum:I have a PAP2T-NA in-front of my router, used to have it behind and it worked fine too but I figured in-front would be better with it's own IP.
This guide looks good »voipfan.net/ATAs/PAP2_fr ··· line.php
- NAT Mapping Enable: yes
- NAT Keep Alive Enable: yes
being the most important, didn't need to port forward when I was behind but you can try forwarding port 5060.
To put it in-front I'm just using a switch.
My modem is SB6120.
A switch is a really good idea. You do NOT want your ATA to bottleneck the traffic going to your router. |
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to marksy
I have FPL on TSI works fine, but the best installation method is ROUTER > ATA in bridged mode > MODEM. this ensures ports are not blocked and telephony gets proper QOS.
if the ATA has no built in router and is designed to be a stand alone network device.
ATA > Router > Modem.
BUT, you must forward a bunch of port. These ports are:UDP 5060-6061, UDP 10000-20000. Also set the ATA to a static IP map in the DHCP of router. so that the network mapping rules don't change.
open config on ATA and make sure it's using sip ports in 5060-6061 and RTP inside 10000-20000.
and should work fine. then run a SHIELD UP! scan and ensure the ports are not CLOSED. |
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I have to disagree unless that's a highly unusual VoIP provider. No ATA needs open ports or port forwards. Even a PBX with lots of internal phones doesn't need it unless it specifically supports SIP devices from the internet and can't use VPN or known IPs. |
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said by vikingisson:unless that's a highly unusual VoIP provider. It used to be (or can be). It was really fun when FPL used to not broadcast its external IP:sip port, but that was a long time ago. Currently, if you're properly configured you don't need to port forward with FPL. If you're not, you will need a workaround, and one common workaround is port forwarding. When FPL/Fongo fiddles with its network, sometimes it forces people to port forward (but you really shouldn't have to anymore). By the way, I feel using an ATA to run traffic through it in bridge mode from the modem to your router is not advisable since many ATAs will just produce bottleneck doing this (and most aren't good enough to do it properly). Also, FPL only uses sip port 5060 and a (random port from) RTP range of 16384 (min) to 16482 (max). So if you're going to port forward, you just need to port forward 5060 and 16384-16482. And these are the only ports that matter for the ATA device as well (simply have these specified: 5060 for the sip port; 16384 for RTP port min and 16482 for RTP port max). Port forwarding that ginormous range BTC Kevin listed is like breaking down a wall of your house just to let a little air in. |
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