Solved Parking call via dial code not working

[Sunny_99]

Hello All,

i have a problem to park an ongoing call via dial code. During a call, i try to park the call by "<Transfer> *00", but it's not working. After a short while i got a voice warning "your call was forbidden by the administrator" ( I don't know the exact words in english, because my system is german).

The activity log shows:

12/07/2020 12:02:34 PM - [CM503008]: Call(C:673): Call is terminated
12/07/2020 12:02:34 PM - Leg L:673.3[EndCall:EndCall] is terminated: Cause: BYE from 127.0.0.1:5483
12/07/2020 12:02:31 PM - [CM503007]: Call(C:673): EndCall:EndCall has joined, contact <sip:[email protected]:5483/UDP>
12/07/2020 12:02:31 PM - [CM503007]: Call(C:673): Extn:19 has joined, contact <sip:[email protected]:5060/udp>
12/07/2020 12:02:31 PM - L:673.3[EndCall:EndCall] has joined to L:673.1[Extn:19]
12/07/2020 12:02:31 PM - [CM503002]: Call(C:673): Alerting EndCall:EndCall by contact <sip:[email protected]:5483/UDP>
12/07/2020 12:02:31 PM - [CM503025]: Call(C:673): Calling T:EndCall:EndCall@[Dev:sip:[email protected]:5483;rinstance=84962b04f5a8650e] for L:673.1[Extn:19]
12/07/2020 12:02:31 PM - L:673.1[Extn:19] failed to reach ParkOrb:*0, reason Forbidden
12/07/2020 12:02:31 PM - Leg L:673.2[ParkOrb:*0] is terminated: Cause: 403 Forbidden/INVITE from 127.0.0.1:5483
12/07/2020 12:02:31 PM - Call to T:ParkOrb:*0@[Dev:sip:*[email protected]:5483;rinstance=ba92bf2d696f112f] from L:673.1[Extn:19] failed, cause: Cause: 403 Forbidden/INVITE from 127.0.0.1:5483
12/07/2020 12:02:31 PM - [CM503003]: Call(C:673): Call to <sip:*[email protected]:5060> has failed; Cause: 403 Forbidden/INVITE from 127.0.0.1:5483
12/07/2020 12:02:31 PM - [CM505001]: Endpoint ParkOrb:*0: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Parking Orbit] PBX contact: [sip:*[email protected]:5060]
12/07/2020 12:02:31 PM - [CM503002]: Call(C:673): Alerting ParkOrb:*0 by contact <sip:*[email protected]:5483/UDP>
12/07/2020 12:02:30 PM - [CM503025]: Call(C:673): Calling T:ParkOrb:*0@[Dev:sip:*[email protected]:5483;rinstance=ba92bf2d696f112f] for L:673.1[Extn:19]
12/07/2020 12:02:30 PM - [CM503027]: Call(C:673): From: Extn:19 ("A+D" <sip:[email protected]:5060>)  to  T:ParkOrb:*0@[Dev:sip:*[email protected]:5483;rinstance=ba92bf2d696f112f]
12/07/2020 12:02:30 PM - [CM503004]: Call(C:673): Route 1: from L:673.1[Extn:19] to T:ParkOrb:*0@[Dev:sip:*[email protected]:5483;rinstance=ba92bf2d696f112f]
12/07/2020 12:02:30 PM - [Flow] Call(C:673): has built target endpoint: ParkOrb:*0 for call from L:673.1[Extn:19]
12/07/2020 12:02:30 PM - [Flow] Target endpoint for *00 is ParkOrb:*0
12/07/2020 12:02:30 PM - [CM503010]: Call(C:673): Making route(s) from Extn:19 to <sip:*[email protected]:5060/UDP>

What happens: I call from 12 to 19, pick-up the call and hit "<Transfer> *00". The call itself got registered, but then got denied with the error message "[Extn:19] failed to reach ParkOrb:*0, reason Forbidden".

If i do the same thing via the SwitchBoard, everything is running ok:

12/07/2020 12:29:53 PM - Leg L:675.2[Extn:19] is terminated: Cause: BYE from local
12/07/2020 12:29:53 PM - [CM503007]: Call(C:675): ParkOrb:SP0 has joined, contact <sip:[email protected]:5483/UDP>
12/07/2020 12:29:53 PM - L:675.4[ParkOrb:SP0] has joined to L:675.1[Extn:12]
12/07/2020 12:29:53 PM - [CM505001]: Endpoint ParkOrb:SP0: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Parking Orbit] PBX contact: [sip:[email protected]:5060]
12/07/2020 12:29:53 PM - [CM503002]: Call(C:675): Alerting ParkOrb:SP0 by contact <sip:[email protected]:5483/UDP>
12/07/2020 12:29:53 PM - [CM503025]: Call(C:675): Calling T:ParkOrb:SP0@[Dev:sip:[email protected]:5483;rinstance=4eb11c3ce5ead64c] for L:675.1[Extn:12]
12/07/2020 12:29:52 PM - [CM503027]: Call(C:675): From: Extn:12 (<sip:[email protected]:5060>)  to  T:ParkOrb:SP0@[Dev:sip:[email protected]:5483;rinstance=4eb11c3ce5ead64c]
12/07/2020 12:29:52 PM - [CM503004]: Call(C:675): Route 1: from L:675.1[Extn:12] to T:ParkOrb:SP0@[Dev:sip:[email protected]:5483;rinstance=4eb11c3ce5ead64c]
12/07/2020 12:29:52 PM - [Flow] Call(C:675): has built target endpoint: ParkOrb:SP0 for call from L:675.1[Extn:12]
12/07/2020 12:29:52 PM - [Flow] Target endpoint for SP0 is ParkOrb:SP0
12/07/2020 12:29:52 PM - [CM503010]: Call(C:675): Making route(s) from Extn:12 to <sip:[email protected]:5060/UDP>

Any ideas what goes wrong?


All the best

Jens

 

[Sunny_99]

Hello JohnS,

unfortunately, it's not working.

In an ongoing call, as soon as i press the hook the call is terminated completely.

If i press the flash key, the caller is thrown out, while the callee (me) get a new dialtone. Then i can dial whatever i want, the connection to the caller is lost.

I can see that in the SwitchBoard also, once i press the Flash key, the call disappears from the display.

Is the "*87" a dial code from the 3CX or from the Grandstream?

The options in the Grandstram HT818 related to Flash:


and


80ms is the standard flash time used by ISDN-pbx and analog phones in germany.


All the best

Jens

 

[leejor]

It has been a while since I have played around with the settings on a Grandstream ATA, but, I'm wondering if the Send Hook Flash Event should be set to NO. Setting to YES will (as it says) sends DTMF, which is usually used in an attempt to trigger a flash on an FXO port device, which you are not trying to do.

Most ATAs have a list of all vertical service (star) codes that are accessible, internally, after a flash.

 

[JohnS_3CX]

*87 is a grandstream code

You may have to hold the hook down a bit shorter (the goal is to imitate a flash). This will give you a dial tone and the other end will hear music on hold. then you can dial the aforementioned code.

Grab the HT818 manual from their website, it includes all this information(that where I read it and tested it)

 

[Sunny_99]

*87 is a grandstream code

Hello JohnS, hello leejor,

thanks for staying with me and this Grandstream problem!

I checked the HT818 admin dokumentation and found some explanation for the settings mentioned earlier:

Send Hook Flash Event: If set to yes, flash will be sent as DTMF event. Default is No. I assume, if set to No a Flash event will be send as an SIP event, but that's not clear. I set it to NO.

Flash Digit Control: When it set to YES it allows the user to perform some call setting when both channels are used while pressing "Flash + [1-4]". I set it to NO. What does it means by "when both channels are used..."?

Callee Flash to 3WC: I set it to NO (default).

Enable Call Feutures: Enables do not disturb, call forward and other call features via the local feature codes on the base. Otherwise, ITSP feature codes can be used. Default is Yes. I set it to YES.

Enable Hook Flash: Enables the FLASH button to be used for terminating calls. Default is Yes. I think this is the setup if the HT818 could recognises the Flash from the phone, but i am not sure. I don't want the call to be terminated. I set it to NO.

I applied the new settings and rebooted the HT818, but to my surprise nothing changed. An ongoing call will still be terminated if pressing the flash key and hen i got a new dialtone on the phone. Now i am i bit out of options...

All the best

Jens

 

[leejor]

Not having used 818, I can't confirm, however, the user guide http://www.grandstream.com/sites/default/files/Resources/ht818_user_guide.pdf would seem to indicate that after a flash you get solid dialtone, where as some ATAs provide stutter dialtone. After a (proper) flash, the connected call should not be dropping.

 

[JohnS_3CX]

I think maybe you are wasting too much time on something very outdated?

I would ask myself if this is worth the trouble VS getting supported voip phones.

And if cost is an issue, our apps are free and don't require any special setup. So try to weigh that against your needs before you invest too much time on an analog adapter which is just a necessary evil just so you can have analog phones.

 

[Sunny_99]

After a (proper) flash, the connected call should not be dropping.

Hello leejor,

that's the point. The problem is: Is dropping the line something the HT818 is doing wrong or is that a 3CX problem?

I will try to get some information by joining the Grandstream forum, maybe i can make things more clear.

If you know of an ATA that works properly in terms of flash control, please let me know.

For now, thanks for your help!

All the best

Jens

 

[Sunny_99]

I think maybe you are wasting too much time on something very outdated?

Hello JohnS,

i am asking the other way around: Why is it necessary to waste so much time for something like that?

VoIP/SIP exists for a long time already, and there are still so many problems (technical / interoperability / company decisions) that block the easy use of the components.

You asked about switching to VoIP phones. I had a ISDN-pbx (Agfeo AS40) with 16 analog phones, 15 of them DECT stations (base/handset). So i tried to find a nice DECT solution for VoIP. But for me as an end user, this was a nightmare. In the good-old analog times you can connect any phone you want to an analog line, the pbx handles the rest. Even DECT handsets could be hooked to any DECT base, maybe with the lack of some features (GAP). On my desk, i have a Gigaset 2020 analog phone with an internal DECT base and a Panasonic DECT handset hooked on. No problem.

If i choose a DECT provider for a VoIP-DECT-Base right now, i am married to that company for the rest of my life. That goes for the base AND the handsets. Example: I tried the Gigaset N670 DECT-Base because they said you can hook any DECT handset. But that is wrong, in fact there are 3 (!) compatible handsets available, starting at EUR 80,- each. You can also pay EUR 150,- without a problem. I looked at SNOM, Yealink etc., it doesn't matter. They are all the same.

So i want to stick at my working analog installation. Analog is not dead... ;-)

Costs are an important issue also. I went for the 3CX-pbx because of that also. The cost effectivness of 3CX is unbeaten. In terms of installation support, it's awesome. I got the pbx cloud-hosted and the internal sbc running in no time without a hassle.

And they provide a supported forum with you guys to help me with my problems. Great job, thank you very much!!!

But if it comes to phone support or software CTI, 3CX is far from being perfect. I will spare that for later discussions.

All the best

Jens

 

[JohnS_3CX]

VoIP/SIP exists for a long time already, and there are still so many problems (technical / interoperability / company decisions) that block the easy use of the components.

SIP doesn't magically go away with your current setup - the gateway/FXS is a SIP converter at the end of the day so you can skip all the additional redundant parts of converting analog to SIP and use SIP straight up.

You asked about switching to VoIP phones. I had a ISDN-pbx (Agfeo AS40) with 16 analog phones, 15 of them DECT stations (base/handset). So i tried to find a nice DECT solution for VoIP. But for me as an end user, this was a nightmare.

I can't say about your specific environment of course, but I must disagree that supported VoIP DECT devices are a nightmare. They provision fast and easy, and the phones can also be paired quickly (all of which needs to be done once and you are now set). We also support devices like the W80 and W90 that do multicell.

If i choose a DECT provider for a VoIP-DECT-Base right now, i am married to that company for the rest of my life. That goes for the base AND the handsets. Example: I tried the Gigaset N670 DECT-Base because they said you can hook any DECT handset. But that is wrong, in fact there are 3 (!) compatible handsets available, starting at EUR 80,- each. You can also pay EUR 150,- without a problem. I looked at SNOM, Yealink etc., it doesn't matter. They are all the same.

You are also married to a supported phone, with supported and tested templates and firmware for the entire lifetime of the product. Remember, a DECT provider needs to commit to ensuring compatibility before they can be part of our supported devices.


However, it's not my place to convince you - the above are just points for consideration and we will still of course help with any questions!

 

[JohnS_3CX]

As per this thread, we can mark the issue as solved

https://www.3cx.com/community/threads/looking-for-a-sip-ata-with-proper-flash-support.78086/