No voice between two VICIdial servers

Asterisk Asterisk Support
1

Hi,
i have created a trunk between two servers in remote locations so i can transfer calls. The calls, voice, everything was working perfectly until two days ago. No changes i have made to the servers or whatsoever, but suddenly the calls can still transfer but there is no voice at all (!). I rechecked again the trunks on both sides and i checked also the NAT rules between the two IP’s, but no luck. I also contacted both ISP’s to make sure that they have not blocked any port. It is the weirdest problem i have faced until now. For your information i am sending you the configurations on both servers.

Server 1 - Sending Calls Trunk Configuration

[transferbr]
disallow=all
allow=ulaw,alaw
type=friend
host=
dtmfmode=auto
insecure=port,invite
canreinvite=no
nat=yes
context=trunkinbound

Global String: transfer1 = SIP/transferbr

Dialplan Entry:
exten => _778,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _778,2,Dial(SIP/transferbr/778,tTo)
exten => _778,3,Hangup

Server 2 - Receiving Calls Trunk Configuration

[receivesip]
disallow=all
allow=ulaw,alaw
type=peer
host=
dtmfmode=auto
canreinvite=no
insecure=port,invite
nat=yes
context=default

Dialplan Entry:
exten => _778,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _778,2,Dial(SIP/cc101,tTo)
exten => _778,3,Hangup

I can see the trunk listed in “sip show peers” in both servers and i can also transfer the call but neither of both parts can hear any voice.

Any suggestions of how to determine what the problem is or what is causing it?

PS: on both routers i have NATed port range 5000-20000 for the link between the two servers.

Thanks in Advance!

2

We would need to see the general section on each side to see if they’ve been configured with localnet and externip, as well as the SIP traffic itself (sip set debug on) of a call attempt as well as “rtp set debug on”.

3

Hi, @jcolp and thanks for answering. With sip set debug on entered on both servers, from the “sending” Server 1 when i dial 778 (the extension for the transferring) i get the following:

PS: Because of the long lines of output i am sending you the links of the PDF uploaded to my Dropbox.

Thanks in Advance

Server who sends calls: https://www.dropbox.com/s/b0rlmhyx77uymwp/LOG%20of%20tranferring%20Server.pdf?dl=0

Server who receives calls: https://www.dropbox.com/s/zdofbixl7avomuj/LOGS%20of%20Server%20who%20receives%20calls.pdf?dl=0

4

The logs did not include many of the SIP messages, you limited them too much and as a result it did not catch them.

It looks as though you don’t have it configured to work behind NAT though. What is the general section of sip.conf on each side?

5

@jcolp all the configuration shown in the first post were made only within VICIdial… so my sip.conf contains only a reference for sip-vicidial.conf… and there is nothing there except the extensions i have create again from within the VICIdial for the operators. Same is on both sides. :frowning:

What makes me crazy is that configuration i showed in the first post was working just fine… until two days ago.

Do i need to create manually the 778 extention which tranfer the calls between the servers and add a registration string in the trunk? Or it is enough with type=peer and type=friend?

6

Something must have changed somewhere. And no, your SIP traffic appears to be going fine. Without a general section with externip and localnet, though, Asterisk will not place a public IP address in the SDP. This will result in media not flowing if both sides are behind NAT or do not have the correct IP address in the SDP.

1 Like
7

And how can i fix that and try it?

8

The sample configuration file[1] has a section detailing how to configure chan_sip for use behind NAT.

[1] https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L869

9

Hi @jcolp! I know it may look stupid but the problem was a faulty Mikrotik Router. As it seems it was a bug or maybe not enough performance from the router that caused the problem. I replaced with a new router with better specs and the voice came “magically”! It is the weirdest problem ever!!

Thanks so much for helping me!

Regards,

10

Most likely it was the Mikrotik sip helper messing you up.

In Winbox go to IP > Firewall > Service Ports and disable it.