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Topics - ky4k0b

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1
Other Yate server issues / yate doesn't proxy rtp between legs
« on: April 05, 2018, 10:53:33 AM »
Hi, I'm facing pretty same issue as described here
https://forum.yate.ro/index.php?topic=1201

I have the following setup with yate binded to two interfaces - wan and vpn


unfortunately yate doesn't pass rtp streams between legs so it causes dead air.

regexroute is very basic - send all calls to Bob(VPN)
.*$=sip/sip:${called}@172.21.42.2:5063;callername=${caller};

If you can help me that would be great. Full debug is here:
http://paste.org.ru/?5xtq93

2
Yate users hangout place / yate basic configuration
« on: May 15, 2014, 06:13:13 PM »
Hello yate gurus!

I just installed yate pbx on debian and need to implement simple scheme with routing based on prefixes.
         GW1 (1.1.1.1)
            ↓  (no registration, auth by ip-address only)
          yate
   ↓                 ↓ (yate registers on both GW2 and GW3)
GW2           GW3

What is done so far:
Code: (accfile.conf) [Select]
[test1]
enabled=yes
protocol=sip
username=user1
password=pw1
registrar=2.2.2.2

[test2]
enabled=yes
protocol=sip
username=user2
password=pw2
registrar=3.3.3.3

I see in /var/log/yate
Code: [Select]
20140515233120.052067 <sip:CALL> SIP line 'test1' logon success to 2.2.2.2:5060hence I suppose that I've been registered successfully.

Next in
Code: (regexroute.conf) [Select]
[default]
^99991001$=tone/dial
^99991002$=tone/busy
^99991003$=tone/ring
^99991004$=tone/specdial
^99991005$=tone/congestion
^99991006$=tone/outoforder
^99991007$=tone/milliwatt
^99991008$=tone/info

^001\(.*\)$=sip/sip:\1;line=test1
^002\(.*\)$=sip/sip:\1;line=test2
[check_addr_auth]
${address}^1\.1\.1\.1:=return
.*=-;error=forbidden;reason=fuckyou
But somehow yate keeps returning 404 for all calls.
Here're my questions:
1) Correct me pls if I made mistakes above.
2) How do I check sip call flow? Basically how can I see it online or enable traces in conf files?
3) How do you apply new configuration? For example after I edit .conf files do I need to restart yate or put any commands? Or is everything applies on fly?
4) How do I enable full non-proxy media for all calls? I need RTP packets go directly from GW1 to GW2

That's all yet. Looking forward to your answers!

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