Number of Simultaneous Calls Error

[slinc]

I have started getting this error:

10:42:14.234 CallConf::Rejected [CM103005] Call(14) is rejected: Too many incoming calls
10:42:14.234 CallLegImpl:nIncoming [CM103000] Call(14): Number of simultaneous calls is limited by Administrator
10:41:33.062 CallConf::Rejected [CM103005] Call(13) is rejected: Too many incoming calls
10:41:33.062 CallLegImpl:nIncoming [CM103000] Call(13): Number of simultaneous calls is limited by Administrator
10:41:21.062 CallConf::Rejected [CM103005] Call(12) is rejected: Too many incoming calls
10:41:21.062 CallLegImpl:nIncoming [CM103000] Call(12): Number of simultaneous calls is limited by Administrator
10:40:39.062 CallConf::Rejected [CM103005] Call(11) is rejected: Too many incoming calls
10:40:39.062 CallLegImpl:nIncoming [CM103000] Call(11): Number of simultaneous calls is limited by Administrator
10:40:27.062 CallConf::Rejected [CM103005] Call(10) is rejected: Too many incoming calls
10:40:27.062 CallLegImpl:nIncoming [CM103000] Call(10): Number of simultaneous calls is limited by Administrator

I still have 3 out of 4 PSTN lines are free. The line will just ring and the auto attendant will not pickup. Is there a setting I should be looking at?

I am using GXW-4108 and SPA921 phones with 4 PSTN lines (NO VOIP). Any ideas?

 

[5qg4]

Please check "Maximum simultaneous calls" of virtual extension for GXW-4108:

By click Lines > Manage > Identification of GXW-4108 > Other Options

 

[slinc]

5gg4,
Thank you for your fast response. I do not see that option under the GXW-4108. Since it is a PSTN gateway and not a VOIP. I looked under other options and just don't see it.

The only thing i can find about Simultaneous calls is under "line Configuration" where you can specifiy a number value (ie. 1) but there is no "check".

Any further advice? Thanks again.

 

[Mirzab]

That option is there for PSTN or VoIP provider lines, just ensure you are selecting line by clicking link under "Identification" column and not under "Gateway/Provider Name"

 

[5qg4]

Just want to confirmed that your GXW4108's identifications (virtual extensions) setting.

Under "Line(s) Configuration" > other options. My GXW4104 setting as below for your reference. The setting of GXW4104 (4 FXO) and GXW4108 (8 FXO) should be almost the same.

Allow outbound calls on this line: ticked
Answer incoming calls on this line: ticked
Answer after (seconds): 0 (default)
Maximum simultaneous calls: 1 (default)

 

[slinc]

5gg4,

My settings are the same as yours under Other Options. Any other ideas?

We are also having a problem with transferred calls being disconnected after about 10-30 seconds. Not sure if it is related or not.

Do you know of anyone that has posted their "working great" GWX-4104 gateway settings? I have spent hours playing with different settings to find the right ones to fix my problems. But still with no luck. Thanks again.

 

[5qg4]

slinc,

No idea for your "Simultaneous Calls Error".

Which type of transfer, blind or attended transfer?

Our attended transfer also drop the call 18sec after the call transfered to the end point. However, the blind transfer seem works fine. Our hard phone is Polycom IP330.

Please try to temporary disable attended transfer at SPA 921 phone. I found this option at my SPA 942 under Advance Mode > Phone > Supplementary Services > Attn Transfer Serv: No

I'm not sure it can be help to fixed your issue or not.

 

[SY]

5gg4,

My settings are the same as yours under Other Options. Any other ideas?

We are also having a problem with transferred calls being disconnected after about 10-30 seconds. Not sure if it is related or not.

Do you know of anyone that has posted their "working great" GWX-4104 gateway settings? I have spent hours playing with different settings to find the right ones to fix my problems. But still with no luck. Thanks again.

Set "Maximum simultaneous calls" to desired maximum of simultaneous calls received on this line...

 

[slinc]

SY,
Since it is a PSTN line I assume it can be no higher than "1". Is that correct?

Any idea why 5gg4 and I have problems transfering calls and then having them disconnect after a few seconds? Would it be a 3CX, Gransstream or Hardphone problem most likely?

Thanks for your help.

 

[SY]

SY,
Since it is a PSTN line I assume it can be no higher than "1". Is that correct?

Any idea why 5gg4 and I have problems transfering calls and then having them disconnect after a few seconds? Would it be a 3CX, Gransstream or Hardphone problem most likely?

Thanks for your help.

About "Maximum..."
I don't know is that correct or not, (it is completely fair!). I think it depends on your line (device, provider etc.) :?:

About "transfer":
Sorry, but I'm also don't know where from and where to you transfer a call. Who initiate it, who receive it... as well as where to it was transferred, what device transferred it, what device received it... :?:

"In general":
Log is not only the "suspected message", there is some context and information... Please check PBX documentation, it describes PBX log messages in detail.

Conclusion: All listed problems are related to following entities - "3CX", "GrandStream", or "Hardphone". Exact reason depends on situation.

Did I help you?

Thanks

 

[slinc]

SY,

Thanks for your response. I think it comes down to the fact that there are so many variables that anything could be the problem. For the most part I think 3CX is typically not the problem but how it is configured with the gateway and phones.

For people like me that don't have a near by reseller professional to do the installation it would be cool if there was a place people could post there "working good" configurations. For example, I am sure others are using GrandStream GXW-4104 and SPA941 phones with great success. I just don't know all the settings they used to make it work good. Therefore, I have to spend a lot of time doing trial and error.
Just an idea.

I will continue to research my problem and post any new information that I have that may assist in fixing these issues. Thanks again.

 

[SY]

slinc,

Problem described in first post of this thread - PBX didn't finish call, but new call is coming. Check "Line status" page. It always show some information about PBX based status of line.

Regards

 

[archie]

It is posiible, that your PSTN gateway originates all calls with the same line-id, so for PBX it is the same line every time. Just to check: set maximum calls to 4 for each line of the gateway.

 

[slinc]

Good Idea Archie I will try that. Thanks for your help.

 

[Psychlist1972]

FWIW, I had this same problem last night. It turns out that one of my calls got "stuck" in 3CX. The line was on hook (a single PSTN line) and usable, but 3CX still showed an active call with the gateway and client extenion both in use.

I had to bounce the service to clear it up. I couldn't find any other way to address it. Even rebooting the gateway had no effect.

Pete

 

[slinc]

Psychlist1972,

I had to restart 3CX also to get it cleared up. We have been running all day and so far not had the problem again except for one short period and then it cleared up on it's own.

 

[AlecM]

Hi.

I am using a GXW-4104. I have only 3 PSTN lines, so FXO line 4 on the GXW-4104 is not connected or mapped to any SIP channel.

I am also seeing the problem where the GXW-4104 seems to send a call through to a 3CX SIP “channel” that is already in use on another PSTN call.

The 3CX server log shows this by a pair of entries phrased as:

14:02:57.834 CallConf::Rejected [CM103005] Call(841) is rejected: Too many incoming calls
14:02:57.834 CallLegImpl:nIncoming [CM103000] Call(841): Number of simultaneous calls is limited by Administrator

This results in the caller getting a continual ringing instead of our 3CX digital receptionist, or the engaged tone.

I had thought this might be the GXW-4104, so tried to configure the “Unconditional Call forward to VOIP” value to specify which PSTN line goes to which 3CX SIP channel. However, when I attempt to enter a specified mapping, the GXW-4104 changes them.

For example, I entered the following mapping:
ch1:10000;ch2:10001;ch3:10002;ch4:;

Which gets changed on update + reboot of the GXW-4104 to:
ch1-3:10000+;ch4:;

This would seem to indicate that the 4104 doesn’t care which SIP channel it uses (though I guess there must be some method of deciding which is used next) – am I correct in thinking this?

I then came across this message thread, so will try and monitor whether the 3CX server is not releasing the virtual extensions quickly enough.

FYI, I am running 3CX on a twin-CPU system:
650MHz p-III (2)
512Mb RAM
Win XP Pro SP2

Plus:
14 internal extensions, all Grandstream BT200's.
Plus the 4104 FXO gateway.


I might hazard a guess that some calls rely on tones (DTMF) for hanging up correctly? In which case there may be some input from the Grandstream gateway or handsets here, where they are not helping the 3CX system recognise when the call has finished? Or am I completely wrong in thinking that?

Is there perhaps a (user configurable) setting in 3CX for the timer duration before releasing the PSTN virtual extension?

Kind regards,

Alec

 

[Ryandc]

SY,
Since it is a PSTN line I assume it can be no higher than "1". Is that correct?

Any idea why 5gg4 and I have problems transfering calls and then having them disconnect after a few seconds? Would it be a 3CX, Gransstream or Hardphone problem most likely?

Thanks for your help.

First up, the Gradnstream 4104 and 4108 are atrocious, at least in my environment. Upgrading to the latest firmware helped some. But they're still bad. So expect a certain level of badness to be relegated to the Gateways.

Next up, to get transfers to work properly, you have to bind the Gateway to the Media Server. Make sure that box is checked. Make sure that your phones are NOT bound to the Gateway.

You probably have a "wrong leg has arrived" message somewhere in your logs. I had the exact same config (well SPA922's rather than 921's, but identical otherwise) and it drove me nuts trying to get it to work.

If this doesn't work, let me know. I'll send you a transcript of my Gateway settings for your reference.

 

[AlecM]

Further to my experience of this problem, I have set my virtual extensions that map to my PSTN lines to permit 3 simultaneous calls. This was actually in response to trying to correct aproblem with attended transfers not operating.

It seems counter-intuitive to add more "simultaneous calls" to an analogue PSTN line than 1, but it does seem to work. I have 3 PSTN lines and corresponding 3CX virtual extensions.

I was concerned that internal extensions trying to make a 4th call out would still get a dial attempt. So I did a test of my system by trying to make 4 calls across my three PSTN lines, both by calling in and by calling out. In both scenarios the respective systems correctly responded with "busy/engaged" tones to the handset making the 4th simultaneous call attempt.

So far I haven't seen the problem of "Too many incoming calls" since doing this last night, where we had been experiencing it once per day. One day is too early to be absolutely sure yet, but I'm hopeful this has helped.

I did also log a support request with Grandstream, as I wasn't sure if it was the 4104 not hanging up the virtual extension properly, but I now think this was a "red herring". They have responded on this, and I'll follow it up anyway, just to be sure.

Alec

 

[LLangston]

Alec,
You may want to try setting the Round-Robin and/or Flexible setting under Channels on the Gateway to rr:1-3 since you only have 3 PSTN lines.

Also may want to look at the FXO Termination settings under FXO Lines and try setting it to 1. Enable Current Disconnect - Yes; with a threshold of 100, 2. Enable Tone Disconnect - No, 3. Enable Polarity Reversal - No, 4. AC Termination Impedance - 600 Ohm(North America), 5. Silence Timeout(X1s) - ch1-4:60

As I have mentioned before, we had the Grandstream 4104 and the BT-200 with it, we got everything working on it, our problem with them was audio quality not being high enough for our clients.

Hi.

I am using a GXW-4104. I have only 3 PSTN lines, so FXO line 4 on the GXW-4104 is not connected or mapped to any SIP channel.

I am also seeing the problem where the GXW-4104 seems to send a call through to a 3CX SIP “channel” that is already in use on another PSTN call.

The 3CX server log shows this by a pair of entries phrased as:

14:02:57.834 CallConf::Rejected [CM103005] Call(841) is rejected: Too many incoming calls
14:02:57.834 CallLegImpl:nIncoming [CM103000] Call(841): Number of simultaneous calls is limited by Administrator

This results in the caller getting a continual ringing instead of our 3CX digital receptionist, or the engaged tone.

I had thought this might be the GXW-4104, so tried to configure the “Unconditional Call forward to VOIP” value to specify which PSTN line goes to which 3CX SIP channel. However, when I attempt to enter a specified mapping, the GXW-4104 changes them.

For example, I entered the following mapping:
ch1:10000;ch2:10001;ch3:10002;ch4:;

Which gets changed on update + reboot of the GXW-4104 to:
ch1-3:10000+;ch4:;

This would seem to indicate that the 4104 doesn’t care which SIP channel it uses (though I guess there must be some method of deciding which is used next) – am I correct in thinking this?

I then came across this message thread, so will try and monitor whether the 3CX server is not releasing the virtual extensions quickly enough.

FYI, I am running 3CX on a twin-CPU system:
650MHz p-III (2)
512Mb RAM
Win XP Pro SP2

Plus:
14 internal extensions, all Grandstream BT200's.
Plus the 4104 FXO gateway.


I might hazard a guess that some calls rely on tones (DTMF) for hanging up correctly? In which case there may be some input from the Grandstream gateway or handsets here, where they are not helping the 3CX system recognise when the call has finished? Or am I completely wrong in thinking that?

Is there perhaps a (user configurable) setting in 3CX for the timer duration before releasing the PSTN virtual extension?

Kind regards,

Alec