SIP canreinvite=yes via multible Asterisk servers

Asterisk Asterisk Support
1

Hi,

I am trying to configure a direct rtp media stream between SIP devices using canreinvite=yes and it is working fine when both SIP devices are connected to the same Asterisk server.
The problem is that we have several Asterisk server in our branch offices and if I try make a call between SIP devices that are connected to different Asterisk servers then the rtp media stream is always via the Asterisk servers.
The Asterisk servers are connected via IAX2 trunks.
Is it possible to implement direct rtp media stream between SIP devices that are connected to different Asterisk servers ?

Best,
Gardar Georg Nielsen.

2

Yes. Use SIP trunks.

3

I have tried to route between the two asterisk via SIP trunk instead of IAX2 trunk but the result is the same. If both SIP devices (Snom phones) are connected to the same server I get direct rtp media stream between the phones.
If I move one of the phones to another server and route between the servers using SIP trunk then all the traffic is routed via the Asterisk server.
Are there any special options required in the SIP trunk between the servers ?

Best,
Gardar Georg Nielsen.

4

Which version of Asterisk? There have been bugs in this area.

5

One server is “Asterisk 1.6.0.22-samy-r60 (Trixbox 2.8.0.3)” and the other is “Asterisk 1.6.0.26-FONCORE-r78 (Trixbox 2.8.0.4)”.
I could try this with both server using Asterisk 1.6.0.26-FONCORE-r78 (Trixbox 2.8.0.4)" ?

Best,
Gardar Georg Nielsen.

6

Both of those versions are “end of life” and neither has had any normal maintenance performed for over a year.

7

If we decide to change to different Asterisk distribution, is AsteriskNOW recomended ? Will it give us up to date Asterisk version ?
Or do you recomend to build our own setup, install Linux, Asterisk 1.8, etc. ?
We are company with almost 1000 users distributed around the world, connecting all branch offices via private MPLS network with each office having their own Asterisk server and PRI ISDN lines.

Best,
Gardar Georg Nielsen.

8

Hello Goggi,

I have problem now which the direct RTP between Polycom Soundstation IP doesn’t works, if i disconnected the LAN cable from Asterisk (i am using Elastix 1.6) then the conversation/RTP stopped. When i plugged again the LAN cable then automatically the voice/conversation continued.
I just read your post and you said it was works. Do you minad to share what was makw it works?

Thanks.