How to suppress echo on Asterisk?

Question

Asterisk

Good day!
I use a virtual PBX AskoziaPBX 4.0.3 I
connected 3 providers TTK, SIPNET and Megafon
1) TTK works flawlessly
2) There are a lot of problems with sipnet, and the main one is ECHO when calling landlines (if you call mobile phones, then there is no echo). There is no echo on the TTK and megaphone. Perhaps this is due to the fact that the sipnet server is at a distance of > 8000 km and the ping is stable at 104 ms.
A question about the sipnet settings: are there mechanisms in the asterisk to combat ping? It is also unclear why there is no echo when calling to mobile or other sip phones.
Internet channel 10 Mbit.
3) There are also enough problems with a megaphone. For example, if I connect a megaphone to a phone located in Vladivostok, the phone starts ringing after 15 seconds. If I connect a megaphone in Khabarovsk (where the asterisk server is located), then the call goes through instantly.
When calling a megaphone, an error always appears in the logs:
2016-09-01 11:45:40 asterisk[12184]: WARNING[9758]: pbx.c:1563 in pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Set(INCOMINGPROVIDERNAME=Megafon
Dialplan copied from TTK, which works perfectly.
In general, the main problem with ECHO is with the sipnet. I have a "bottle" for someone who can help figure it out.
Update 09/20/2016
There is new information on the problem with the megaphone:
Firstly, the error that the PBX generated could be solved simply by removing the | in the name of the provider.
Secondly, we managed to find out that the problem at all branches with a megaphone is not related to the provider.
I logged into the TTK SIP account and placed an incoming call on the branch phone, which is located at a distance of 900 km.
What happened: The incoming call screen lights up on the branch phone, but the phone itself does not make any sounds. And now, after about 15 seconds, it suddenly starts ringing. Moreover, an employee at the branch can pick up the phone at the moment when the screen lights up and start a conversation.
(with multiphone, identical story)
Question: why does the call sound start to go after ~ 15 seconds?
During tests in most cases 90%, this time was 15 seconds. There were calls after 20-30 seconds. And also sometimes the phone at the very beginning of the call makes a ringtone sound for a split second and immediately falls silent, displays information about the incoming call on the screen and starts ringing after 15-20 seconds.
The asterisk and phones on branches are behind NAT
voxlink.ru/kb/asterisk-configuration/asterisk-nat-...
3rd scheme. The ports are forwarded, the connection quality is amazing. The routers at the branches are different. The problem is the same.
Update 09/28/2016
1) I assembled a separate server, connected it via a dedicated channel with a white IP
2) I rolled up an asterisk on it and created a provider account + branch phone
3) Next, I registered a new server in the branch phone
4) I called
All of a sudden, the phone at the branch office instantly beeped an incoming call!
At the same time, an impeccable connection, in short, everything worked as it should.
There is a version that, in addition to the signal port 5060 and voice RTP 10000-20000, you need to register some other one on the firewall in order for the phone to ring right away.
I can't find which port to forward for Yealink. For example, the cisco website has a table with all the ports needed for telephony. www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/p...
I didn't find it according to Yelink.
Has anyone encountered such a problem? The truth is already out there somewhere!

Answer 1

The echo occurs at the junction of a digital with an analog, so it is necessary to configure exactly the equipment that is at this junction. so kick the provider. And then, you won’t get rid of the echo completely, you can only reduce it to the threshold of audibility, while the quality of the connection may suffer.
Advice - buy a city Sipovsky number from Rostelecom.

Answer 2

I managed to suppress the echo by turning off the echo canceller. Those. in a bunch of sip - analog (PSTN) there were 2 suppressors, and they interfered with each other constantly changing characteristics.

Answer 3

For the echo issue, try setting jbenable=yes globally in sip.conf.
Because problem with specific providers, then it would be nice to discuss this problem with the technical support of the provider

Answer 4

Echo - these are problems on the side of the one who hears this echo. But! In most cases, the problem is at the junction of the SIP analog. There should be no echo in a pure sipe. Write to the technical support of the sypnet.
And about the fight against ping - this is five :)))
Regarding the errors - he told you in pure English that you have a vertical line instead of commas - you forgot to convert the dialplan from the old version of the asterisk

Answer 5

I've been in dialogue with Sipnet's technical support for a long time. What was advised to do:
1) Advised to turn on the recording of conversations Turned on
. On the recording, an echo is heard even at the stage of the welcome menu (when the employee has not yet picked up the phone). By the way, Yealink T19E2 IP phones.
2) Call not from Yealink.
Well, here, firstly, I don’t have other phones, and secondly, the first paragraph eliminates the need.
3) Connect Yealink bypassing Asterisk (In the account settings of the phone itself, I registered all the parameters)
I connected a dedicated channel, bypassing the asterisk and the firewall. I called a landline phone (analogue) from TTK, in response I heard a weak echo with a delay of about a second. Then I called the same phone, but the provider is Rostelecom. I heard a strong echo of myself on yealink with the same delay of about a second.
Then he called his mobile, there was no echo. Also, the echo is not heard if you call through the sipnet to another IP phone.
From point 3, it seems to become clear that the problem is at the junction of the SIP analogue.
Threw off in support of a sipnet the link to this subject.
4) They said to connect the sipnet to the X-lite or Zoiper softphone. Haven't tested this option yet.
Now I will build a separate server for the asterisk. I want to check the connection bypassing Hyper-V