Starting in recent days (Nov 6 to Nov 8, 2010), when I pickup the phone (connected with GrandStream HT-286 ATA) and make calls, I heard the dial tone before I key in the telephone numbers.
However, occasionally, it is totally dead air after I entered the telephone numbers. The other side phone was ringing. When pick up, no voice can be heard on both side.
If I tried to make call two or three times, some time it works and then for another while it happen again.
I tried to put the ATA both in front of the router directly to the cable modem and behind the router. Both get the same situation.
I purchase the configuration file two weeks ago and set up the ATA both before and behind the routers and it was working fine without any problem.
My router open the port 5060-6061 and 10000 to 12000 and should not be my router problem as I tried to put the ATA in front of the router directly on the cable modem (Shaw) and getting the same result.
I think FPL should look into their server performance and see if it is really too busy to handle call now as the customer base is growing. The server may make outgoing call based on sip information but is too busy to setup the RTP channels to provide ringtone and voice channel while calling the other side.
Can't get dial tone or ringing tone making outgoing calls
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- Active Poster
- Posts: 80
- Joined: 10/16/2010
- SIP Device Name: Softphone, GS ATA HT286
- Firmware Version: 3.0.1
- ISP Name: Shaw
- Computer OS: XP, Win7
- Router: DLink
- Smartphone Model: Iphone 4S
- iOS Version: 5.1.1
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- Site Moderator
- Posts: 2131
- Joined: 07/16/2009
- SIP Device Name: Grandstream 286 & 701
- ISP Name: Worldline.ca
- Computer OS: Windows 7 Ultimate / Mac OS X
- Router: TR1043ND w/ DD-WRT Mega
- Smartphone Model: Galaxy S3
- Android Version: 4.0.4
- Location: Cambridge
Re: Can't get dial tone or ringing tone making outgoing calls
We're barely using any capacity at all on the new server.tigerking wrote:I think FPL should look into their server performance and see if it is really too busy to handle call now as the customer base is growing. The server may make outgoing call based on sip information but is too busy to setup the RTP channels to provide ringtone and voice channel while calling the other side.
Are these trouble calls being dialed straight, or are you using our two-stage long distance package to call off-net areas that aren't in the usual free area?
Steve
Fongo
Development Support Specialist.
Fongo
Development Support Specialist.
-
- Active Poster
- Posts: 80
- Joined: 10/16/2010
- SIP Device Name: Softphone, GS ATA HT286
- Firmware Version: 3.0.1
- ISP Name: Shaw
- Computer OS: XP, Win7
- Router: DLink
- Smartphone Model: Iphone 4S
- iOS Version: 5.1.1
Re: Can't get dial tone or ringing tone making outgoing calls
Steve,FPL-steve wrote: We're barely using any capacity at all on the new server.
Are these trouble calls being dialed straight, or are you using our two-stage long distance package to call off-net areas that aren't in the usual free area?
The numbers I was calling are direct numbers in the free areas but on other carriers, Telus 604-678-xxxx, Bell Mobility 778-233-xxxx, AIC 604-275-xxxx and Magic Jack number 604-227-xxxx both are in the FPL Vancouver free areas. I didn't purchase LD voucher from FPL and should not be allowed for LD call to off-net area and If I do so, I should get the prompt "your plan is not covered to call this number", etc.
Anyhow, calling FPL numbers direct have no problem at all.
The problem happended occasionally and hard to catch a consistent pattern. Most of the time happended after the phone on hook for long time and the first attempt to make call got a dead air but the other side ring. Pick up no voice. The 2nd attempt in general OK but some time it will need to make the 3rd attempt. Sometime it happened between successful calls and then one or two occassions it failed. Some time you have to wait for 5 seconds to get the ring tone after dialed the number completely. (I set the no key detection to 2 seconds in my ATA).
I tried to program the ATA shorten the re-registration time from 3600 seconds (1 hour) down to 180 seconds (3 minutes) and the keep alive from 20 seconds to 10 seconds but still seeing the same problem.
That's why I suspect a processing power issue or the way the program handling call establishment need improvement.
Any clue?
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- Site Moderator
- Posts: 2131
- Joined: 07/16/2009
- SIP Device Name: Grandstream 286 & 701
- ISP Name: Worldline.ca
- Computer OS: Windows 7 Ultimate / Mac OS X
- Router: TR1043ND w/ DD-WRT Mega
- Smartphone Model: Galaxy S3
- Android Version: 4.0.4
- Location: Cambridge
Re: Can't get dial tone or ringing tone making outgoing calls
Would you mind PMing me a few of the full numbers in question? Also, if you have fresh tests to said numbers in the last 24-48hrs that would be excellent too.
I'll see if the logs show anything that may help.
I'll see if the logs show anything that may help.
Steve
Fongo
Development Support Specialist.
Fongo
Development Support Specialist.
-
- Active Poster
- Posts: 80
- Joined: 10/16/2010
- SIP Device Name: Softphone, GS ATA HT286
- Firmware Version: 3.0.1
- ISP Name: Shaw
- Computer OS: XP, Win7
- Router: DLink
- Smartphone Model: Iphone 4S
- iOS Version: 5.1.1
Re: Can't get dial tone or ringing tone making outgoing calls
I just figure out why I always got a silence when making outgoing call. It would be with codec selection.
I learned from other post that FPL use G729A codec for outgoing and 711u for incoming call?
That may be the reason why I have no problem receiving call but could not make outgoing call consistently as I was setting "PCMU" to the 1st choice of the codec.
During the call setup, it may negotiate on PCMU codec for outgoing call and I guess it mismatch with the VoIP server and not able to set up the RTP probably and getting occasion silence. (But the call log show it was successfully connected)
After I changed the 1st choice back to G729A/B, I could successfully make some calls so far so good.
HOWEVER, it will go back to square ONE regarding the DTMF code sending issue.
I originally followed FPL's recommendation to set G729A as first choice codec but I discovered the DTMF code send to the other side via PSTN is very bad and my company e-mail IVR is not always recognize the key strokes. I therefore changed the 1st choice codec to PCMU (711u) and the DTMF quality improved a lot and both FPL *98 and my company voice mail IVR can recognized the key strokes.
I set the ATA to send DTMF via RTP RFC2833.
Now I have my voice channel fixed by changing the 1st choice codec back to G729A when making outgoing calls BUT I have the DTMF problem occasionally.
Unfortunately, my ATA only allow one choice of DTMF insertion method but not both "in-audio" and "RTP".
If I set the ATA to send DTMF via "in-audio".
It solve my company voice mail access problem BUT I COULD NOT retrieve FPL voice mail using the phone to call *98 or remote access line. I believe, FPL use out-of-band DTMF for internal call (i.e. use my phone attached to the ATA call the FPL#).
The best way is FPL do some investigations see if it would be able to make sure using phone attached to ATA connected to FPL when calling *98 and remote access number to retrieve voice-mail also recognize the "in-audio" DTMF signal on top of DTMF via RTP.
OR
Do some researches to improve the DTMF qaulity they send over the PSTN if user choose to send DTMF via RTP!!
I think the former solution is easier for FPL to implement.
I learned from other post that FPL use G729A codec for outgoing and 711u for incoming call?
That may be the reason why I have no problem receiving call but could not make outgoing call consistently as I was setting "PCMU" to the 1st choice of the codec.
During the call setup, it may negotiate on PCMU codec for outgoing call and I guess it mismatch with the VoIP server and not able to set up the RTP probably and getting occasion silence. (But the call log show it was successfully connected)
After I changed the 1st choice back to G729A/B, I could successfully make some calls so far so good.
HOWEVER, it will go back to square ONE regarding the DTMF code sending issue.
I originally followed FPL's recommendation to set G729A as first choice codec but I discovered the DTMF code send to the other side via PSTN is very bad and my company e-mail IVR is not always recognize the key strokes. I therefore changed the 1st choice codec to PCMU (711u) and the DTMF quality improved a lot and both FPL *98 and my company voice mail IVR can recognized the key strokes.
I set the ATA to send DTMF via RTP RFC2833.
Now I have my voice channel fixed by changing the 1st choice codec back to G729A when making outgoing calls BUT I have the DTMF problem occasionally.
Unfortunately, my ATA only allow one choice of DTMF insertion method but not both "in-audio" and "RTP".
If I set the ATA to send DTMF via "in-audio".
It solve my company voice mail access problem BUT I COULD NOT retrieve FPL voice mail using the phone to call *98 or remote access line. I believe, FPL use out-of-band DTMF for internal call (i.e. use my phone attached to the ATA call the FPL#).
The best way is FPL do some investigations see if it would be able to make sure using phone attached to ATA connected to FPL when calling *98 and remote access number to retrieve voice-mail also recognize the "in-audio" DTMF signal on top of DTMF via RTP.
OR
Do some researches to improve the DTMF qaulity they send over the PSTN if user choose to send DTMF via RTP!!
I think the former solution is easier for FPL to implement.