can't hear other side voice when make call or receive call

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Re: can't hear other side voice when make call or receive ca

Postby ckkatan » 11/29/2014

xebbmw wrote:Hi all,

I bought recently the VOIP key. After configuring my ATA SPA2102, I was not able to hear the other party voice although I was heard by them. This was happening for incoming or outgoing calls.

My ATA was behind a router (not in DMZ however) and the ports UDP 5060-5061 and 16384-16482 were already forwarded. I tested my VOIP key on a soft phone (3CX) and everything was ok, for outgoing and incoming calls I could hear the other party. After more search on the net I found the following settings which are required if ATA is behind a router:

Voice > SIP
NAT Support Parameters
Handle VIA received = yes
Handle VIA rport = yes
Substitute VIA Addr = yes
Send Resp to Src Port = yes

You may test it yourself to see if it helps. Remember that the problem was only for ATA, soft phone was working fine.


Thanks. I finally resolved my issue with forwarding both SIP and RTP ports without touching the NAT support parameters.

I also could resolve the same issue with connecting the ATA box on the DMZ port or ATA box on the modem with Router behinds the ATA box. But, I preferred the ATA on Router with port forwarding setting.
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Re: can't hear other side voice when make call or receive ca

Postby ckkatan » 11/29/2014

xebbmw wrote:Hi all,

I bought recently the VOIP key. After configuring my ATA SPA2102, I was not able to hear the other party voice although I was heard by them. This was happening for incoming or outgoing calls.

My ATA was behind a router (not in DMZ however) and the ports UDP 5060-5061 and 16384-16482 were already forwarded. I tested my VOIP key on a soft phone (3CX) and everything was ok, for outgoing and incoming calls I could hear the other party. After more search on the net I found the following settings which are required if ATA is behind a router:

Voice > SIP
NAT Support Parameters
Handle VIA received = yes
Handle VIA rport = yes
Substitute VIA Addr = yes
Send Resp to Src Port = yes

You may test it yourself to see if it helps. Remember that the problem was only for ATA, soft phone was working fine.


Thanks for the information.

I resolved my connection issue by forwarding both SIP and RTP ports on my Router.

I also could resolve the same issue by either
1. connecting the ATA box on a DMZ port, or
2. connecting in Router -> ATA box -> Modem configuration (in series)

But, I preferred the port forwarding method. The ATA box is now on one of the router network ports.
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Re: can't hear other side voice when make call or receive ca

Postby JREED » 12/02/2014

ckkatan wrote:
xebbmw wrote:Hi all,

I bought recently the VOIP key. After configuring my ATA SPA2102, I was not able to hear the other party voice although I was heard by them. This was happening for incoming or outgoing calls.

My ATA was behind a router (not in DMZ however) and the ports UDP 5060-5061 and 16384-16482 were already forwarded. I tested my VOIP key on a soft phone (3CX) and everything was ok, for outgoing and incoming calls I could hear the other party. After more search on the net I found the following settings which are required if ATA is behind a router:

Voice > SIP
NAT Support Parameters
Handle VIA received = yes
Handle VIA rport = yes
Substitute VIA Addr = yes
Send Resp to Src Port = yes

You may test it yourself to see if it helps. Remember that the problem was only for ATA, soft phone was working fine.


Thanks for the information.

I resolved my connection issue by forwarding both SIP and RTP ports on my Router.

I also could resolve the same issue by either
1. connecting the ATA box on a DMZ port, or
2. connecting in Router -> ATA box -> Modem configuration (in series)

But, I preferred the port forwarding method. The ATA box is now on one of the router network ports.
ckkatan wrote:
xebbmw wrote:Hi all,

I bought recently the VOIP key. After configuring my ATA SPA2102, I was not able to hear the other party voice although I was heard by them. This was happening for incoming or outgoing calls.

My ATA was behind a router (not in DMZ however) and the ports UDP 5060-5061 and 16384-16482 were already forwarded. I tested my VOIP key on a soft phone (3CX) and everything was ok, for outgoing and incoming calls I could hear the other party. After more search on the net I found the following settings which are required if ATA is behind a router:

Voice > SIP
NAT Support Parameters
Handle VIA received = yes
Handle VIA rport = yes
Substitute VIA Addr = yes
Send Resp to Src Port = yes

You may test it yourself to see if it helps. Remember that the problem was only for ATA, soft phone was working fine.


Thanks for the information.

I resolved my connection issue by forwarding both SIP and RTP ports on my Router.

I also could resolve the same issue by either
1. connecting the ATA box on a DMZ port, or
2. connecting in Router -> ATA box -> Modem configuration (in series)

But, I preferred the port forwarding method. The ATA box is now on one of the router network ports.


Hello,

Please explain in details how to forward port or explain in details about your 1. step and 2. step.

Kind regards,
JR
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Re: can't hear other side voice when make call or receive ca

Postby Mango3 » 12/03/2014

Hi JR,

What is the model number of ATA and router that you are using?
What symptoms are you having that you intend to correct?

Just a note with regards to the post you quoted. DMZ should never be used as it presents a serious security risk. Also, the SIP Port should not be forwarded unless things won't work any other way. In that case, a high SIP number should be used.

Let us know some more specifics about your problem and we'll try to make some suggestions.
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Re: can't hear other side voice when make call or receive ca

Postby ckkatan » 12/04/2014

JREED wrote:
ckkatan wrote:
xebbmw wrote:Hi all,

I bought recently the VOIP key. After configuring my ATA SPA2102, I was not able to hear the other party voice although I was heard by them. This was happening for incoming or outgoing calls.

My ATA was behind a router (not in DMZ however) and the ports UDP 5060-5061 and 16384-16482 were already forwarded. I tested my VOIP key on a soft phone (3CX) and everything was ok, for outgoing and incoming calls I could hear the other party. After more search on the net I found the following settings which are required if ATA is behind a router:

Voice > SIP
NAT Support Parameters
Handle VIA received = yes
Handle VIA rport = yes
Substitute VIA Addr = yes
Send Resp to Src Port = yes

You may test it yourself to see if it helps. Remember that the problem was only for ATA, soft phone was working fine.


Thanks for the information.

I resolved my connection issue by forwarding both SIP and RTP ports on my Router.

I also could resolve the same issue by either
1. connecting the ATA box on a DMZ port, or
2. connecting in Router -> ATA box -> Modem configuration (in series)

But, I preferred the port forwarding method. The ATA box is now on one of the router network ports.


Hello,

Please explain in details how to forward port or explain in details about your 1. step and 2. step.

Kind regards,
JR


Hi JR,

For 2. "step" (not a step): It is a configuration setup. I connect the "Router" to the "ATA" box and the "ATA" box to the Modem. Nothing else needed (other than programming the ATA box with FPL parameters) for this setup.

For 1. step", you will need to set up a DMZ port on the Router and connect the "ATA" box on the DMZ port. As mentioned by many, it is not a very good idea to connect the ATA box to a DMZ port due to the security issue.

As for port forwarding, please consult your router user manual. Every router is slightly different.
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Re: can't hear other side voice when make call or receive ca

Postby jtrider » 12/12/2014

Hello.

I have trouble for incoming calls. It works fine for outbound calls but no inbound. It rings several time but no answer. The display on the phone shows coming from 100.
I use with obi110 devise and freephoneline. Any suggests is appreciated.
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Re: can't hear other side voice when make call or receive ca

Postby dlau88 » 12/17/2014

Hi,

I am having the same issue, I used a friend's working FPL setting on my SPA2102 and still not able to hear the other party. When I open a ticket, Fongo's sends me an e-mail stating they don't provide technical support and closed the ticket. Just wondering what "Issue Type" should I put down so tech. support will look into it.
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Re: can't hear other side voice when make call or receive ca

Postby redzed1990 » 12/18/2014

Thanks for the suggestions re changes to the SIP settings on the ATA which were specified as follows:

Voice > SIP
NAT Support Parameters
Handle VIA received = yes
Handle VIA rport = yes
Substitute VIA Addr = yes
Send Resp to Src Port = yes

I can confirm this solved the lack of incoming audio on my SPA112 in combination with a Rogers cable modem and a Linksys WRT54GS upgraded to DD-WRT v24-sp2 firmware.

I found the lack of incoming audio with FPL to be a strange problem because I am using this ATA and router combination with Callcentric for my VoIP service in Florida, and it worked fine from the get-go when I brought the SPA112 home and plugged it into this router. I purchased the SIP from FPL to use on the second line of this ATA, and no incoming audio. This post solved my problem, and I am appreciative of people who understand the technology and also take the time to post solutions for those of us less knowledgeable.
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No ring tone and Can't hear other side voice when make call

Postby amy » 12/19/2014

Hello,

I bought the second VIP key today and try to set it up with my new ATA PAT2T. When I make a landline call, I cannot hear the ring tone and other party phone did not ring. But, if other party pick up the phone then other party can hear me and I cannot hear anything. Note: There is no problem on receiving call and both parties can hear with no issue.

Note: I have the old existing ATA with my old VIP key and it is working fine for more than 2 years. Hence, I put my old VIP key to replace the new VIP key on to the new ATA PAT2T to test it. And it works with no problem. Hence, I don't think that there is an issue on my both ATAs and my old VIP key.

Would you please help me out on this.

thanks,
Amy
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Re: No ring tone and Can't hear other side voice when make c

Postby Mango3 » 12/19/2014

redzed1990 wrote: with a Rogers cable modem and a Linksys WRT54GS upgraded to DD-WRT v24-sp2 firmware.

Is your DD-WRT router receiving a public IP address from Rogers (i.e. modem is in bridge mode)? And if not, are you using the special Rogers proxy?

amy wrote:I bought the second VIP key today and try to set it up with my new ATA PAT2T. When I make a landline call, I cannot hear the ring tone and other party phone did not ring. But, if other party pick up the phone then other party can hear me and I cannot hear anything. Note: There is no problem on receiving call and both parties can hear with no issue

Please set the following on your new ATA with the new VoIP key:

SIP tab
Handle VIA received: yes
Handle VIA rport: yes
Substitute VIA Addr: yes

These are the settings that redzed1990 reported are working (minus Send Resp to Src Port which may be coincidental). Let us know if this solves the problem.
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Re: HT701 firmware 1.0.6.1

Postby juched » 12/21/2014

Having a problem where I can hear the caller, but they cannot hear me. Also, when calling out I don't hear ringing, but the other side can pickup. Odd. I am running dd-wrt on my router, and I have tried both UPNP and Keep Alive for NAT Traversal.

Ideas? Port forwarding and fixed RTP and SIP ports don't work either.
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Re: can't hear other side voice when make call or receive ca

Postby sulagok » 12/30/2014

I used Linksys PaP2, flawlessly, for over a year now until it got bad recently. I now have Grandstream HT701 and after following the configuration provided religiously, I still cannot hear an inbound caller while outband calls are nice and smooth.

I can see from this forum that some people have also complained about this but no cogent resolution.

I will appreciate any assistance in this regard.
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Re: SPA2102 (Firmware 5.2.13.004)

Postby luntan » 01/23/2015

Thank you, Jake and Mango, for your posts. Very helpful. I have two lines. Both work except one problem: incoming call to Line 1 rings only once (not even complete, while my FPL setting is 5 rings before voicemail) and then goes right to voicemail before I have the time to pick up the phone. When I was with my previous service provider, I used the same model with no problem (of course they had already configured the adapter and made sure all was ok), so I am sure there must be some place I could adjust, but I just can't figure out where. Any idea? There's another reason I prefer the incoming call rings longer, because eventually, we want to use our own answering machine for Line 1, just like before, let the answering machine kicks in before FPL voicemail so that we can check messages without calling FPL voicemail. Thanks.
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Re: SPA2102 (Firmware 5.2.13.004)

Postby Mango » 01/23/2015

That is an interesting problem. Maybe FPL is not receiving the response your device sends. I wonder why that could be? You may wish to test for SIP ALG and NAT issues:

Try changing your SIP Port on the problem line to a random number between 20000 and 65535. If both lines are with FPL, try setting your Proxy to voip.freephoneline.ca for one line and voip2.freephoneline.ca for the other.

If my first guess is wrong, perhaps there is an electrical problem. To test for this, unplug the telephone cable from your ATA, and use your cell phone make a test call to that line with your cell phone. If via your cell phone you hear the requisite five rings before your FPL voicemail, try to reduce your ring voltage to 70.

I'm pretty sure if DND or call forwarding was accidentally enabled, there would be no ring at all, but you may want to check the User 1 tab just in case.

Let us know how things go.
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Re: SPA2102 (Firmware 5.2.13.004)

Postby luntan » 01/23/2015

Mango, your suggestion of checking User tab eventually led it to work! Thank you! It's not the DND or call forwarding. Both are fine. It's the ring setting that does the trick. I checked both User 1 tab and User 2 tab. I first changed default ring from "1" to "5" in User 2 tab (LIne 2, the good line), Line 2 started to behave like Line 1, the problem line. So I changed the default ring back to "1" in User 2 tab and went to changed the default ring in User 1 tab from "5" to "1", and Line 1 starts to ring 5 times before going to FPL voicemail, and of course when I turn on the answering machine, the message now goes there exactly as I wanted. What I don't understand though is why the default ring has to be "1" in order for it to ring 5 times, but it really doesn't matter now if it works.

One minor issue and it happens to Line 2. The incoming call ring sounds normal from the caller's end, but it varies at my end. Sometimes it sounds 5 normal rings, sometimes the ring is delayed after 2 rings, sometimes it rings twice, and silent for the length of 1-2 rings and then another ring. If you have any suggestion to improve, I'd appreciate it.
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Re: SPA2102 (Firmware 5.2.13.004)

Postby Mango » 01/24/2015

I'm glad I was able to help, even if indirectly!

As for your Line 2 ring problem, you may want to verify that the following is set on the Regional tab:

Ring Waveform: Sinusoid
Ring Voltage: (Try 70 or 90)
Ring Frequency: (Try 25 if none of the above solved the problem.)

This affects both lines, but different phones will respond in different ways to these settings.
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Re: can't hear other side voice when make call or receive ca

Postby Mango » 02/04/2015

Updated configuration for HT701: viewtopic.php?f=15&t=16292
Updated configuration for PAP2(T): viewtopic.php?f=15&t=16294
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dropped off after 32 seconds

Postby csdeer » 02/14/2015

I am using voip3.freephoneline.ca as SIP proxy. which solves my problem calling cell phone.
Now, another issue comes. I make a call and it will drop off after 32 seconds. If I change to voip2.freephoneline.ca, it will be longer than 32 seconds. But, with vopi2, I can not call cells.
I ma using WRT54G and rogers cable. please help.
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Re: can't hear other side voice when make call or receive ca

Postby Mango » 02/16/2015

What ATA are you using?
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Re: can't hear other side voice when make call or receive ca

Postby T2WIN » 02/19/2015

Hello,
All was well for over 14 months and yesterday incoming calls routed direct to voicemail and outgoing calls now get busy signal.
Rebooted modem, router and OBI100.
OBI dashboard shows SP1 FPl is registered. Test call to OBI is successful.
FPL SIP settings Status = connected (with latest firmware).
Switched from voip.feephoneline.ca, 5060 to voip4.freephoneline.ca, 6060 and back again. No change.

Appreciate your help
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Re: can't hear other side voice when make call or receive ca

Postby Mango » 02/19/2015

Your problem seems to be different than the others who posted in this thread.

I find these symptoms somewhat curious.

- We know the device is registered.
- DND would not cause outgoing calls to stop working.
- Physical connection problem would not permit registration, or test calls to OBiTALK.
- Changing servers eliminates the possibility of a corrupted NAT connection or accidental ban.

As always, be sure your ATA is configured as per the official guide: viewtopic.php?f=15&t=16090

Do you have other VoIP hardware or software on your network? If so, set your X_UserAgentPort to a random number between 20000 and 65535.

Is your router's SIP ALG enabled? If so, disable it. SIP ALG should always be disabled, unless you have a specific reason to use it.

If you have your router's SPI Firewall, DoS protection, or UDP-Flood Attack filtering enabled, you may want to disable those for testing purposes. I have historically seen malfunctioning attack detection cause problems with VoIP. If that does not solve the problem, change it back. These security settings should only be disabled if things won't work any other way.

If my suggestions do not solve the problem, you may want to try swapping routers if you have a spare one, or try to connect your OBi to a friend's network, with a different router. That will tell us where to investigate further.
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Re: can't hear other side voice when make call or receive ca

Postby T2WIN » 02/19/2015

Thanks Mango, I will try and get back
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Re: can't hear other side voice when make call or receive ca

Postby T2WIN » 02/19/2015

Halfway there. Receiving calls ok.
But dialing out gives me a fast busy tone when pressing for dial tone followed by the dial tone. Pressing first digit dial tone stops. But after the 3rd digit I get a regular busy tone. I tried a few different area codes.

I'm on Rogers Cable. First I followed the official guide you posted (viewtopic.php?f=15&t=16090). But no change. So then I followed the OBI guide for Rogers cable users. Basically voip4.
I did have to reboot a few times to get the dial string in the digitmap to stick. A few x's missing (strange). But that is lined up now.

Getting closer. Do you suggest anything else/different?
TIA
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Re: can't hear other side voice when make call or receive ca

Postby Mango » 02/19/2015

T2WIN wrote:I did have to reboot a few times to get the dial string in the digitmap to stick.


That is a little concerning. From within System Management >> Auto Provisioning, be sure that ITSP Provisioning and OBiTalk Provisioning both have the method "Disabled". You may also want to disable Voice Services >> OBiTALK Service, at least until you get the problem solved. Then you can re-enable it.

After doing this, reboot, and double check that your configuration matches the Rogers guide.

If you still have problems, check Physical Interfaces >> PHONE Port. Be sure everything is set to its default.

If you still have problems, inspect Call History and report what it says for the test calls you've made.
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Re: can't hear other side voice when make call or receive ca

Postby T2WIN » 02/19/2015

Situation unchanged.
Incoming calls operational.
Outgoing dial tone starts with a fast busy signal (about 6-8), followed by a regular dial tone. First digit stops dial tone, second digit is good. But the third digit triggers a very fast busy signal for several seconds.

If you still have problems, check Physical Interfaces >> PHONE Port. Be sure everything is set to its default.

OutBoundCallRoute was not Default. Changed to Default to test. Only difference was {911:sp1}. But again no difference.

If you still have problems, inspect Call History and report what it says for the test calls you've made.

Call logs show incoming calls but nothing outgoing.
I am able to dial OBI test number beginning with **9. Test works.

Went back to FPL SIP Settings.SIP Status: connected. Inbound/Outbound Proxy: voip.freephoneline.ca. This does not appear to change regardless of ProxyServer input in OBI (e. voip, voip2, voip3, voip4).
The DIgitMap entry has been stable throughout. Maybe my error initially??
Thanks
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