Digital Recepcionist with no call audio

[dlrmartin]

Hello,
I'm really excited with this product, it's the best!

But... I'm experiencing a trouble.
I have all incoming calls working ok, but after setup a digital recepcionist:
Call is attended by DR, tranfered to extension, but when extension pick up the phone no audio is transmited or received. I meant, PSTN caller and extension hasn't audio...
Then, I remove DR and all works fine again...

I'm confused, please help.
Regards,
Martin

 

[Costas3CX]

Hello Martin
I would like some more info please
The call is from PSTN line to DR , then you press option #? and transfer to extension , and then no audio?
What is gateway model? what is phone model?
When you phone from PSTN to extension directly, audio is ok?
How about transfers. Do transfers work normally? Blind and attended work?
Just some question that will help with troubleshooting...

 

[dlrmartin]

Thanks for your answer.
I'm using a gateway linksys SPA400, X-lite and eyebeam softphones.

I've tested with all calls from PSTN.
At DR I tested with following configs in diferent times:
Key 0: connecto to extension
Timeout : connecto to extension

If PSTN line is routed to an extension or ring group, no problem.
If PSTN line is routed to DR, the call is attended by DR and transfered to extension defined on DR, the extension rings and when person pick up no audio is listened by PSTN caller or extension person.
When phone from PSTN to extension directly, audio is ok. Transfers are ok.

Let me know if I could give you more information.

 

[dlrmartin]

any updates?
please

 

[SY]

any updates?
please

It will be easier to investigate problems if you provide logs.

Thanks,
Stepan

 

[dlrmartin]

hello,
I just upgraded to build 2434
And here I post logs generated when I make an incoming call answered by DR and tranfered to an extension, without audio. not audio at all.
Please help me to find the trouble.
I'm copying 2 tries with same result, no audio.
192.168.7.1 is the server
192.168.7.102 is the ext.
192.168.7.12 is the gateway

16:10:12.406 StratLink:nHangUp [CM104001] Call(2): Ext.124 hung up call; cause: BYE; reason: SIP;description="User Hung Up"
16:09:48.656 CallLegImpl:nConnected [CM103001] Call(2): Created audio channel for Ext.124 (192.168.7.102:51880) with third party (192.168.7.1:7066)
16:09:48.656 StratInOut:nConnected [CM104005] Call(2): Setup completed for call from Ln:10010@spa400 to Ext.124
16:09:45.500 CallConf:nProvisional [CM103003] Call(2): Ext.124 is ringing
16:09:34.015 CallLegImpl:nConnected [CM103001] Call(2): Created audio channel for Ln:10010@spa400 (192.168.7.12:10000) with Media Server (192.168.7.1:7064)
16:09:34.000 CallConf:nIncoming [CM103002] Call(2): Incoming call from anonymous (Ln:10010@spa400) to sip:[email protected]
16:09:22.671 ServRegs:nAdd [CM113002] Registered: Ext.124
16:08:59.093 ServRegs:nRemove [CM113003] Unregistered: Ext.124
16:08:26.218 StratLink:nHangUp [CM104001] Call(1): Ext.123 hung up call; cause: BYE; reason: SIP;description="User Hung Up"
16:08:20.953 CallLegImpl:nConnected [CM103001] Call(1): Created audio channel for Ext.123 (192.168.7.1:56512) with third party (192.168.7.1:7060)
16:08:20.953 StratInOut:nConnected [CM104005] Call(1): Setup completed for call from Ln:10010@spa400 to Ext.123
16:08:08.390 CallConf:nProvisional [CM103003] Call(1): Ext.123 is ringing
16:07:57.062 CallLegImpl:nConnected [CM103001] Call(1): Created audio channel for Ln:10010@spa400 (192.168.7.12:10000) with Media Server (192.168.7.1:7056)
16:07:56.953 CallConf:nIncoming [CM103002] Call(1): Incoming call from anonymous (Ln:10010@spa400) to sip:[email protected]

 

[SY]

hello,
I just upgraded to build 2434
And here I post logs generated when I make an incoming call answered by DR and tranfered to an extension, without audio. not audio at all.
Please help me to find the trouble.
I'm copying 2 tries with same result, no audio.
192.168.7.1 is the server
192.168.7.102 is the ext.
192.168.7.12 is the gateway

16:10:12.406 StratLink:nHangUp [CM104001] Call(2): Ext.124 hung up call; cause: BYE; reason: SIP;description="User Hung Up"
16:09:48.656 CallLegImpl:nConnected [CM103001] Call(2): Created audio channel for Ext.124 (192.168.7.102:51880) with third party (192.168.7.1:7066)
16:09:48.656 StratInOut:nConnected [CM104005] Call(2): Setup completed for call from Ln:10010@spa400 to Ext.124
16:09:45.500 CallConf:nProvisional [CM103003] Call(2): Ext.124 is ringing
16:09:34.015 CallLegImpl:nConnected [CM103001] Call(2): Created audio channel for Ln:10010@spa400 (192.168.7.12:10000) with Media Server (192.168.7.1:7064)
16:09:34.000 CallConf:nIncoming [CM103002] Call(2): Incoming call from anonymous (Ln:10010@spa400) to sip:[email protected]
16:09:22.671 ServRegs:nAdd [CM113002] Registered: Ext.124
16:08:59.093 ServRegs:nRemove [CM113003] Unregistered: Ext.124
16:08:26.218 StratLink:nHangUp [CM104001] Call(1): Ext.123 hung up call; cause: BYE; reason: SIP;description="User Hung Up"
16:08:20.953 CallLegImpl:nConnected [CM103001] Call(1): Created audio channel for Ext.123 (192.168.7.1:56512) with third party (192.168.7.1:7060)
16:08:20.953 StratInOut:nConnected [CM104005] Call(1): Setup completed for call from Ln:10010@spa400 to Ext.123
16:08:08.390 CallConf:nProvisional [CM103003] Call(1): Ext.123 is ringing
16:07:57.062 CallLegImpl:nConnected [CM103001] Call(1): Created audio channel for Ln:10010@spa400 (192.168.7.12:10000) with Media Server (192.168.7.1:7056)
16:07:56.953 CallConf:nIncoming [CM103002] Call(1): Incoming call from anonymous (Ln:10010@spa400) to sip:[email protected]

What is "Ext.124"?

Thanks,
Stepan

 

[dlrmartin]

the extension 124 is where DR forward the call.
So,
if I call, DR pick up and offer to dial 0 to transfer to extension 124.
when 124 pick up the phone, no audio in neither way.

 

[dlrmartin]

any help. please?

 

[SY]

bind your gateway to Media server(spa400). Does it help?

 

[dlrmartin]

Excellent SY!
it works.
Thank you.
Martin

 

[SY]

Excellent SY!
it works.
Thank you.
Martin

I'm glad to hear it but please specify your gateway model.

Thanks,
Stepan

 

[dlrmartin]

sure.
I'm using linksys SPA400

thanks

 

[SY]

sure.
I'm using linksys SPA400

thanks

I could be more smart if catch it two posts before! :lol:

I need just kindly ask you to check how it works after binding on media server. Just pay attention on progressive delay during conversation or other audio artefacts

Thanks,
Stepan

 

[dlrmartin]

hi,
yes, we are experiencing long delays on communications...
we are evaluating desactivate DR due to this trouble.
do you have any fix?

thanks
Martin

 

[dlrmartin]

Finally, due to hi delays we decided disble DR....
Please found a solution to this issue