Issue with SIP audio calls in SIP as RTMP .

[Karan Bhansali]

hi,
Previously audio calling was working fine with DTMF code but now DTMF is not accepting . Please consider the issue as highest priority. Below are the files attached of property and log

 

[Karan Bhansali]

Any update. It is important, please close it asap

 

[Max]

hello
try to remove 'telephone-event' from the exclude statement

codecs_exclude_sip_rtmp  =opus,g729,g722,mpeg4-generic,vp8,mpv,telephone-event

It should be

codecs_exclude_sip_rtmp  =opus,g729,g722,mpeg4-generic,vp8,mpv

telephone-event is DTMF

 

[Karan Bhansali]

Hi
Still the issue is not solved. Before everything was working fine. The server is also updated to latest version. Please need to resolve by today itself.

 

[Max]

Hello
Please provide:
1. tcpdump log captured during your own test:

tcpdump udp -s 4096 -w log.pcap

2. SIP call details
1) login
2) authentication name
3) password
4) outbound proxy
5) port
6) callee number (what number we should call to hear reply)
7) DTMF number (what DTMF number we should send to get reaction)

3. SSH access to your server.

You can send logs and credentials to support@flashphoner.com
Please note. We need all listed information to proceed.

 

[Karan Bhansali]

Hi,

I need to book conference for that. You need to call between the specified time period, so that you can dial in to the conference

The conference will start at 8pm IST and the end time of the conference will be 11:00 IST.

All the details are mailed to the support@flashphoner.com

 

[Karan Bhansali]

Any updates? When I can use the server again. ? It is production server

 

[Max]

Hello

We have checked your server according received SSH access.
You have version WCS 5.0.x
On the current support level (Simple License, Forum support) we do not provide any scheduled support sessions based on a time window.

Please
1. Install latest 5.1.x version.
https://docs.flashphoner.com/display/WCS5RU/Release+notes
2. Test and reproduce your issue.
3. Arrange testing access to your environment 24/7. Then we will be able to check.

If you can't provide unlimited access, please gather debug logs according documentation and send to support@flashphoner.com for investigation.
https://docs.flashphoner.com/display/WCS5EN/Preparing+an+error+report

When I can use the server again. ? It is production server

We do not manage your servers.
If testing affects your production servers you may better spin up a new server for testing.

 

[Karan Bhansali]

Hi,
So have you checked using the credentials provided ? Does the DTMF work

 

[Karan Bhansali]

Hi,
Now we have two servers with wcs 5.0 and 5.1 latest version .

Using wcs 5.0 i am able to do normal sip call and but when i try with dtmf i have issues - it has elastic ip
Using wcs 5.1 i am not even able to call using sip - it has elastic ip

Below i have attached flashphoner.properties files of both servers

I can give you access of the audio bridge on Monday.

 

[Max]

Hello
If you are testing with plivo, you can create a simple IVR menu, not a conference bridge.
Then we will be able to test DTMF with the menu.

 

[Karan Bhansali]

Hi,
We don't use IVR menu in production, so using audio bridge test DTMF, let me know what time you can test both the instance , so accordingly i can give you access to audio bridge.

 

[Max]

Hello
As we mentioned above we do not schedule time-frames on the current support level. Please send all necessary instructions and accesses which allow us to test this any time.
I.e. you can setup a test plivo account and provide access and instructions for this stage account.

 

[Karan Bhansali]

Hi,

All the details are given to the email id mentioned

 

[Max]

Hello.
We reproduced the issue with Plivo.com and raised internal case WCS-1672. We investigate it and let you know when there will be any results.

 

[Karan Bhansali]

Hi,

Any updates??

 

[Max]

Hello. We will inform you through this forum thread once we have any updates.

 

[Max]

Hello.
We fixed SIP call establishing issue in build 5.1.3695, you welcome to test it.
As for DTMF signal sending, your conference just plays a music no matter PIN entered or not. When call is established from SIP softphone (MicroSIP or Bria for example), the result is the same - just playing a music. So you should arrange IVR menu for reaction to be more straightforward while testing.

 

[Karan Bhansali]

Hello,
We don't know anything about IVR since we don't use it , is there some other way to test DTMF?

 

[Max]

If you can not arrange IVR menu, please try to call via Plivo.com to any answering machine with IVR (bank, call center etc) and send DTMF on its request.