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Avaya Client SDK - General
» SIP DTMF during Other Phone mode, 21/01/2022 08:04:56
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Hello,
we are using the 'User' property 'OtherPhoneService' to connect our client to a mobile phone. While in this mode we noticed that dtmf sounds aren't working (Command: Call.SendDtmf(DtmfTone))
Do you know if we're missing any settings of if this is an issue within the sdk?
Thanks for help and best Regards,
Viktor Thierbach
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Avaya Client SDK - General
» Lost Audio after network reconnect., 18/06/2020 06:04:47
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Yes i am using a desktop application and i am connected via Ethernet
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Avaya Client SDK - General
» Lost Audio after network reconnect., 17/06/2020 08:13:11
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So, what i don't understand here is why the registration couldn't succeed.
Because it seems to be fine. Every other new call incomming or outgoing call is fine an can be handled.
Just the one call that was active while the disconned happened seems to be in an invalid state.
Currently it would be enough for me if there was a way to just end this call and call the customer again by creating a new call. But it doesn't react to any command.
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Avaya Client SDK - General
» Lost Audio after network reconnect., 16/06/2020 09:32:44
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Hi,
thanks for the help.
1)When the registration failure occurs the current call is getting closed (I get the CallRemoved Event), then after the reconnect I get this call again with the CallCreated event. But now I can't hear or talk to the other person.
I also can't end the the call by using the sdk (tried the call.end() command), but the station I called can end it and the sdk will notice it.
(Btw. I noticed today that this call is getting closed automatically after about 1 min)
2)I am working on a desktop client (using the ix sdk) which is connecting to AURA. To test this network issue i just unplugged the cable for a few seconds.
The requested log is attached
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Avaya Client SDK - General
» Lost Audio after network reconnect., 15/06/2020 10:49:31
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Hi,
I'm having the issue that I loose all audio data when a network problem occures.
On getting the RegistartionFailed event I try reconnecting to the server and this works as long as I don't have a call while the disconnect happens. After reconnecting I get the callservice event "CallCreated" and everything looks fine, except that I can't talk to to or hear the other person on the line.
Everything is fine when I drop the call and create a new one. Is there something I can do to restore the audio after the reconnect?
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Avaya Client SDK - General
» IX Call Forwarding and enhanced Call Forwarding, 18/03/2020 08:00:57
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Yes, the station I am using is of type 9611SIPCC.
Do you maybe have a list for not supported features in CC?
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Avaya Client SDK - General
» IX Call Forwarding and enhanced Call Forwarding, 17/03/2020 04:38:58
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I submitted my last reply too early and editing doesn't seem to work so here ist the missing text.
Hi,
here is a screenshot of all my buttons.
I removed the buttons agnt-login, auto-in and manual in but the error is still the same.
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Avaya Client SDK - General
» IX Call Forwarding and enhanced Call Forwarding, 17/03/2020 04:32:09
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Hi,
here is a screenshot of all my buttons.
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Avaya Client SDK - General
» IX Call Forwarding and enhanced Call Forwarding, 16/03/2020 11:02:37
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Thank you, our initial configuration didn't seem to work here.
Now I'm setting ppm enabled manually right before creating the user, but I'm still getting the same error.
I attached a new Log file. I coud see PPM enabled : 'true' in there now.
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Avaya Client SDK - General
» IX Call Forwarding and enhanced Call Forwarding, 11/03/2020 12:19:29
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Hello Pallavi and thanks for your reply.
1) Yes call forwarding is configured. Enhanced call forwarding ist not configured anymore.
2) Yes the call forwarding button is visible on my CM station and it is working fine as long as I don't use the sdk.
3) Yes PPM is enabled
I attached my debug log.
Thanks,
Viktor Thierbach
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Avaya Client SDK - General
» IX Call Forwarding and enhanced Call Forwarding, 05/03/2020 07:26:34
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Hello,
I'm trying to forward all incoming calls to another station - by using the IX sdk.
Threrefore I wanted to activate the features 'Callforwarding' and 'EnhancedCallForwarding'. But the Capability already shows me that it is not allowed, with denial reason 'invalid state'.
When I try to use the enable method 'EnableCallForwarding' or 'SetEnhancedCallForwardingStatus' I'm getting the exception "Unsupported - Feature invocation failed".
I'm not sure if I'm missing something to enable these features or if there is something wrong with my station.
Regards
Viktor
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Avaya Client SDK - General
» SIP Invalid VDN, 28/11/2019 03:21:01
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Hi,
you're right. I'm sorry I mixed this up with another issue when we were using DMCC.
So in case of the latest scenario the vdn 5070 is the correct one. I can see in this number in the log file as History-Info.
Do you know how I can find this data in the sdk. Right now I'm listening to the CallService IncomingCallReceived Event but I can't find this information there.
Thanks a lot
Viktor Thierbach
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Avaya Client SDK - General
» SIP Invalid VDN, 27/11/2019 04:23:25
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Hi,
thank you. This should be the correct log now.
Setup: Agent 5343 is logged in at station 732
VDN 5071
VDN 5070
Szenario: Phone 702 calls vdn 5071
5071 transfers the call to vdn 5070
The agent has the skill for 5070 so he will get the call now.
But I only get the information for vdn 5070 not that the call came from vdn 5071 at first.
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Avaya Client SDK - General
» SIP Invalid VDN, 22/11/2019 08:11:23
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Hi,
sorry for the late response.
I attached the log from the sdk. Is this the one you wanted?
732 is my station
702 is the calling station
Thanks
Viktor Thierbach
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Avaya Client SDK - General
» SIP Invalid VDN, 04/11/2019 06:53:59
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Hello,
I have a problem getting the correct vdn number when using SIP. (Fine with a H.323 station). I'm using two vdn numbers 5436 and 5437. 5436 is configured by vector to transfer the call to 5437.
Now, when using a SIP phone the 'Called Device Id' will still be 5436 but I want the last vdn number it got transferred to.
When using H.323 the 'Called Device Id' will correctly display 5437.
Already tried VDN Override but there was no difference on SIP.
Thanks for any help.
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