Hi, I have tried many times to set up Obi110. The SIP provider is Freephoneline. Below is the SIP information.
SIP Username: 177880242??
SIP Password: XXXXXXXX
Inbound/Outbound Proxy:voip.freephoneline.ca
Registration Expires: 180 to 3600
Use Preferred Codec: G711U or G729a
Use Preferred Codec Only:No
RTP Packet Size: 0.020
Ring Tone Frequency: 52
Ring Tone Voltage: 90
Recommended Linksys Dial String:(911|[2-9]xxxxxxxxx|1xxxxxxxxxx|011xxxxxxxxxxxx.|98*|[6-7]x*xxxxxxxxxxx.)
(please note that this is a Linksys dial string we use for our devices and may not work with third party, non-Linksys/Sipura devices)
I have entered these values into ProxyServer, AuthUserName and AuthPassword. Other items are the default. Then it can call out, but it doesn't ring. I have tried many times to set up it , it is only ringing 3 times.
What can I do now?
buhuo,
Hi, you can try this;
1) verify the System Status of the OBi unit to see if it says registered and it stays registered
2) Port forward the RTP ports to the OBi ITSP Profile A= OBI 16600-16798, and Profile B= 16800-16998
3) Check call histories in the OBi and in FPL
Then try again.
I have checked it out.
1) It stays registered. It shows Registered (server=208.65.240.142:5060; expire in 29s)
2) Yes, Port forward the RTP ports to the OBi ITSP Profile A= OBI 16600-16798, and Profile B= 16800-16998
3) In FPL, it shows all call in and call out history.In Obi, the call history is normal.
I means the Obi can call out, but when I call in, there is no ring in Obi, so I can't receive the calls.
If you call your FPL line and it should be ringing, can you pick up the phone and talk?
When your line should be ringing, what does the OBi's Phone and Line Status page show?
After a failed call, what appears, if anything, on the Call History page?
When I call this FPL line, it doesn't ring. There is no any tone in the phone even if I pick it up then.
Calling out is always OK.
In the Obi110, it shows 'Number of Active Calls: 0' when calling this FPL line, and there is no inbound record in call history.
In FPL call logs, it shows 'Disconnect reason' is 'Normal calling clearing' after calling this FPL number, but there is no ring.
buhuo,
Sign into your FPL acct and have a look in the "Settings" -> "Call follow me settings" to see how you setup the "Follow me settings".
1) What do you have setup in "Follow me status:"
2) What do you have setup in "Follow me mode:"
1)"Follow me status" is Disable.
2)"Follow me mode" is "Always call forward".
and "Voicemail to email" is "none"
"rings before voicemail" is 5.
I have tried another ATA, it can call out and call in. Everything is OK.
buhuo,
Verify the following setting by logging directly into the OBi:
Voice Services -> SPx -> X_InboundCallRoute : ph
where SPx is the FPL trunk (SP1 or SP2).
If you have something different there, please post the value.
Yes.
SP1 -> X_InboundCallRoute : ph
the FPL trunk is SP1
Try changing X_UserAgentPort from 5060 to 5070 or from 5061 to 5071. If no luck, try forwarding that UDP port (the new value) in your router to the OBi. If still no luck, post details of router and modem.
I have tried changing X_UserAgentPort from 5060 to 5070 or from 5061 to 5071. After I changed it, then in first calling in, it rings and can talk and can hear "tick" periodically 7s , but in second calling in, it rings and no voice in the phone, in third calling in, no ring.
Because I'm not very familiar with my router, I don't know how to forward this UDP port in my router to OBi. I'm learning it. I appreciate that you can tell me how to do it.
In Security log in my router, it shows a lot of errors. Do you think the router blocked this UDP port?
"Oct 19 14:36:20 2011 Firewall Error Firewall internal SIP ALG: Failed parsing. Error code: -1019 [repeated 2 times, last time on Oct 19 14:36:20 2011]
Oct 19 14:36:02 2011 Firewall Error Firewall internal SIP ALG: Failed parsing. Error code: -1020 [repeated 4 times, last time on Oct 19 14:36:18 2011]"
Quote from: buhuo on October 19, 2011, 02:42:59 PM
"Oct 19 14:36:20 2011 Firewall Error Firewall internal SIP ALG: Failed parsing. Error code: -1019
OBi recommends to disable SIP ALG. Try doing this in your router.
Thank you very much! I'm trying to find how to disable SIP ALG. My router model is Dlink BCM96358
Was there a resolution to this issue?
No. Telus customer representative said they couldn't change the setting online. So I have no idea what to do now. I have tried this Obi110 in my friend's home, his provider is Shaw Cable, everything is OK.
Buy a new router or hack the existing one to disable ALG.
My issue is with a TPLink WR340G router -- and the fact that I'd eventually like to have more than one SIP provider. Doing permanent port forwards ruin that option.
I'm wondering if I can use a dd-wrt to run just a SIP bridge of somesort that maintains the sessions as a gateway proxy.
Quote from: onecanobi on November 02, 2011, 01:48:03 PM
My issue is with a TPLink WR340G router -- and the fact that I'd eventually like to have more than one SIP provider. Doing permanent port forwards ruin that option.
Why does doing permanent port forwards prevent you from having more than one SIP provider?
If you have a modem/router with an incompatible SIP ALG that can't be disabled, you can often work around the problem by setting X_UserAgentPort to a non-standard value (e.g. 5070 and 5071 instead of 5060 and 5061), and/or connecting to a non-standard port at the provider (if implemented on their server).
In some cases, forwarding the SIP local UDP ports (whatever values are used for X_UserAgentPort) is required, possibly in addition to the above. There should be no problem forwarding two ports, if you have two providers on the OBi, or forwarding many ports (two for each OBi), if you have several OBi devices.
Occasionally, none of the above will help. In such cases, it's usually possible to set the modem/router to be a simple bridge and use it with an external router. In very rare situations, that won't work, either, but it would usually be possible to use a commodity modem in place of the one supplied by the ISP.
Quote from: RonR on November 02, 2011, 02:00:46 PM
Quote from: onecanobi on November 02, 2011, 01:48:03 PM
My issue is with a TPLink WR340G router -- and the fact that I'd eventually like to have more than one SIP provider. Doing permanent port forwards ruin that option.
Why does doing permanent port forwards prevent you from having more than one SIP provider?
because most providers expect publicip:5060 to be forwarded to sip device, and i plan on having more than one physical device.
Quote from: Stewart on November 02, 2011, 02:34:11 PM
If you have a modem/router with an incompatible SIP ALG that can't be disabled, you can often work around the problem by setting X_UserAgentPort to a non-standard value (e.g. 5070 and 5071 instead of 5060 and 5061), and/or connecting to a non-standard port at the provider (if implemented on their server).
In some cases, forwarding the SIP local UDP ports (whatever values are used for X_UserAgentPort) is required, possibly in addition to the above. There should be no problem forwarding two ports, if you have two providers on the OBi, or forwarding many ports (two for each OBi), if you have several OBi devices.
Occasionally, none of the above will help. In such cases, it's usually possible to set the modem/router to be a simple bridge and use it with an external router. In very rare situations, that won't work, either, but it would usually be possible to use a commodity modem in place of the one supplied by the ISP.
Thank you for this, I will try this out and report back.
Quote from: onecanobi on November 02, 2011, 04:36:57 PM
because most providers expect publicip:5060 to be forwarded to sip device, and i plan on having more than one physical device.
Most providers support an array of SIP ports (5060, 5061, 5062, 5063, etc.). I currently use an OBi and PAP2 on the same LAN with ports 5060/5061 forwarded to the OBi and 5062/5063 forwarded to the PAP2.
Quote from: onecanobi on November 02, 2011, 04:36:57 PM
... because most providers expect publicip:5060 to be forwarded to sip device, and i plan on having more than one physical device.
IMO, that is not true. All SIP UDP packets have both a source and a destination port number. Can you name even one provider that requires port 5060 (or any specific port) on the user's end? I know that one can use any local port with all I have tried, including Callcentric, Anveo, VoIP.ms, Localphone, Phonepower, VOIPo, sipgate, Future-Nine, Vitelity, Voxbeam, VoxOx, and ippi.
In contrast, many providers accept registrations on server port 5060 only. Because that port is often blocked or specially (and incorrectly) treated, it's common for servers to also support at least one non-standard port. The specific values are provider dependent but do not relate to port forwarding.
As RonR pointed out, you can have e.g. device #1 with local ports A and B, and device #2 with local ports C and D. You forward ports A and B to the private IP address of device #1; ports C and D to device #2. This should work for any values of A, B, C, and D that are not in conflict with other devices or functions on your network.
Quote from: Stewart on November 02, 2011, 05:11:18 PM
Quote from: onecanobi on November 02, 2011, 04:36:57 PM
... because most providers expect publicip:5060 to be forwarded to sip device, and i plan on having more than one physical device.
IMO, that is not true. All SIP UDP packets have both a source and a destination port number. Can you name even one provider that requires port 5060 (or any specific port) on the user's end? I know that one can use any local port with all I have tried, including Callcentric, Anveo, VoIP.ms, Localphone, Phonepower, VOIPo, sipgate, Future-Nine, Vitelity, Voxbeam, VoxOx, and ippi.
In contrast, many providers accept registrations on server port 5060 only. Because that port is often blocked or specially (and incorrectly) treated, it's common for servers to also support at least one non-standard port. The specific values are provider dependent but do not relate to port forwarding.
As RonR pointed out, you can have e.g. device #1 with local ports A and B, and device #2 with local ports C and D. You forward ports A and B to the private IP address of device #1; ports C and D to device #2. This should work for any values of A, B, C, and D that are not in conflict with other devices or functions on your network.
The issue rises when you have older ATA hardware that isn't as flexible to co-exist with the OBI. :)
When I tried this method with a PAP2 and a Cisco 7940 in the past, it didnt work too well.
Anyway, thanks to your help above, i ended up doing a few things:
1) enable STUNEnable and set STUNServer to stun.softjoys.com port 3478
2) enable X_SymmetricRTPEnable [ seemed like a good thing to enable, not sure if it did anything ]
3) login to FreePhoneLine's webpage, and change a) Voicemail settings to ring 5 times prior to going to VM [ from a setting of 1 ] and b) ensure Follow Me Settings are disabled.
4) add X_UserAgentPort for port 6060 on the Obi for SP2
5) add a virtual server for port 6060 udp/tcp on the tplink router mapping to the OBI's internal IP.
After all of this, I am now able to receive incoming calls.
Without 4 and 5 completed, I would get a busy tone when dialing the number, and no ringing would happen.
I will in the very near future switch the cisco deskphone on to the same internal network and see if it can co-exist with a second FPL line and a different port forward.
Quote from: onecanobi on November 02, 2011, 07:59:21 PM
Quote from: Stewart on November 02, 2011, 05:11:18 PM
Quote from: onecanobi on November 02, 2011, 04:36:57 PM
... because most providers expect publicip:5060 to be forwarded to sip device, and i plan on having more than one physical device.
IMO, that is not true. All SIP UDP packets have both a source and a destination port number. Can you name even one provider that requires port 5060 (or any specific port) on the user's end? I know that one can use any local port with all I have tried, including Callcentric, Anveo, VoIP.ms, Localphone, Phonepower, VOIPo, sipgate, Future-Nine, Vitelity, Voxbeam, VoxOx, and ippi.
In contrast, many providers accept registrations on server port 5060 only. Because that port is often blocked or specially (and incorrectly) treated, it's common for servers to also support at least one non-standard port. The specific values are provider dependent but do not relate to port forwarding.
As RonR pointed out, you can have e.g. device #1 with local ports A and B, and device #2 with local ports C and D. You forward ports A and B to the private IP address of device #1; ports C and D to device #2. This should work for any values of A, B, C, and D that are not in conflict with other devices or functions on your network.
The issue rises when you have older ATA hardware that isn't as flexible to co-exist with the OBI. :)
When I tried this method with a PAP2 and a Cisco 7940 in the past, it didnt work too well.
Anyway, thanks to your help above, i ended up doing a few things:
1) enable STUNEnable and set STUNServer to stun.softjoys.com port 3478
2) enable X_SymmetricRTPEnable [ seemed like a good thing to enable, not sure if it did anything ]
3) login to FreePhoneLine's webpage, and change a) Voicemail settings to ring 5 times prior to going to VM [ from a setting of 1 ] and b) ensure Follow Me Settings are disabled.
4) add X_UserAgentPort for port 6060 on the Obi for SP2
5) add a virtual server for port 6060 udp/tcp on the tplink router mapping to the OBI's internal IP.
After all of this, I am now able to receive incoming calls.
Without 4 and 5 completed, I would get a busy tone when dialing the number, and no ringing would happen.
I will in the very near future switch the cisco deskphone on to the same internal network and see if it can co-exist with a second FPL line and a different port forward.
My router is Dlink DVA-G3810BN/TL. I have been following this instruction, but it doesn't work now.
were you able to setup the virtual server, or port forward?
you could try maybe 5061 or 6061 instead of 6060 that I tried.
but I think the key is the 5 rings prior to vm setting in FPL's configuration page, along with proper port forwarding/mapping.
is your obitalk on dhcp or static ip?
SAME PROBLEM HERE :(
Hello yesterday i purchased a configuration file and today i have been trying all day to set it up in my OBIHAI 110 ATA, i have already read freephoneline forums as well as OBITALK forums but i cant make it ring when someone calls.
Router: linksys wireless-N home router WRT120N (No SIP ALG i have looked so many times and havent found it)
ATA: ObiHai 110 with firmware up to date
I have test the outbound call and it works fine, quality is good too only problem is the inbound calls ringing that doesn't ring at all. PLEASE HELP!
Quote from: Veronica on December 16, 2011, 06:01:09 PM
Router: linksys wireless-N home router WRT120N
Veronica,
Try temporarily connecting the OBi directly to your modem, eliminating the WRT120N. Power-cycle the modem and the OBi after making the connection. It's important to determine if your router is part of the problem and the best way to do that is to take it out of the picture.
Quote from: RonR on December 16, 2011, 06:06:42 PM
Quote from: Veronica on December 16, 2011, 06:01:09 PM
Router: linksys wireless-N home router WRT120N
Veronica,
Try temporarily connecting the OBi directly to your modem, eliminating the WRT120N. Power-cycle the modem and the OBi after making the connection. It's important to determine if your router is part of the problem and the best way to do that is to take it out of the picture.
thank you for reply, i will try that but what you mean with Power-cycle? could you explain?. thanks
Disconnect and reconnect the power from both the modem and the OBi after making the direct connection.
thank you will try that then and report back , but can you help me with something else how do i disable provisioning on device?
Log into the OBi at the IP address returned by dialing ***1 and set:
System Management -> Auto Provisioning -> ITSP Provisioning -> Method : Disabled
System Management -> Auto Provisioning -> OBiTALK Provisioning -> Method : Disabled
The default username/password is admin/admin.
Quote from: RonR on December 16, 2011, 06:18:53 PM
Disconnect and reconnect the power from both the modem and the OBi after making the direct connection.
OK so i connected the obi to the modem directly, then while both connected i disconnected the router and the obi from power waited few minutes and made a test my boyfriend called to my frephoneline number and it RANG! yeiii BUT then i connected the router back to the modem and the obi to the router waited few minutes, made a test DID NOT RING! :'(, im frustated now what else can i do? PLEASE HELP!
Make a port forwarding entry in your router, forwarding ports 5060-5061 to the IP address of the OBi.
Quote from: RonR on December 16, 2011, 10:47:22 PM
Make a port forwarding entry in your router, forwarding ports 5060-5061 to the IP address of the OBi.
i did that i even changed X_UserAgentPort for port 5070 , in freephoneline i have voicemail to wait 5 rings to get in and follow me is disaled aswell. im going to try with port 6060 as a user said and report back.
Also i have port fordward the RTP ports to OBI.
My suggestion to forward ports 5060-5061 would only possibly help if you haven't changed the OBi or Freephoneline from the default of 5060.
Quote from: RonR on December 16, 2011, 11:02:07 PM
My suggestion to forward ports 5060-5061 would only possibly help if you haven't changed the OBi or Freephoneline from the default of 5060.
i havent change the default 5060 just changed the X_UserAgentPort to 5070 which didnt work too
alright i changed X_UserAgentPort to 6060 i made a port forwarding to 6060 aswell (apart of the port 5060 that i port forwarded too), made test and did not ring, so went back went to Aplications & Games>>QoS and enabled priority to ports: 5060 to 5061 and 6060 to 6061 in Medium(recommended) priority, then just like that decided to clean my temp files, cache and more in my computer with Ccleaner made a test again and VOILA IT RANG :o just to be sure i made another test and it rang again ;D ;D ;D so problem solved for now, i'll play with the codects tomorrow tho what you would recommend to stay in G729 as first priority or change to G711U in Canada.
Thanks for the support i will report back if i have any problems thanks @RonR.
hello can anyone using freephoneline post some tips and tricks/setups, a friend from equador called my SP1 and SP2 lines using her Google Voice with a softphone in her android smartphone using headphones and she said is hearing eco and cuts a bit , what could cause this, tho calling here in canada so far no problem.
Im using codec G729 and dont know if G711U is better, also i cant find the Use Preferred Codec Only: No option in the obi web interface
Veronica