Media path and transfer between servers

Asterisk Asterisk Support
1

I’m having trouble understanding how things work with the media path or if this is even possible. Consider the following scenario:

sip phone A -> server A -> IAX -> server B -> sip phone B

Now the person at sip phone B transfers to an extension on server A. I would expect server B to get out of the media path, but it won’t. I end up with the following:

sip phone A -> server A -> IAX -> server B -> IAX -> server A -> sip phone C

What I would like is that after sip phone B transfers back to server A, it become:

sip phone A -> server A -> sip phone C

I end up with messages like this on server B:

-- Attempting native bridge of IAX2/reading-28 and IAX2/reading-6
-- Attempting native bridge of IAX2/reading-28 and IAX2/reading-6
-- Attempting native bridge of IAX2/reading-28 and IAX2/reading-6
-- Channel 'IAX2/reading-28' unable to transfer
-- Channel 'IAX2/reading-6' unable to transfer
-- Channel 'IAX2/reading-28' unable to transfer
-- Channel 'IAX2/reading-6' unable to transfer
-- Channel 'IAX2/reading-28' unable to transfer
-- Channel 'IAX2/reading-6' unable to transfer

I don’t have notransfer=no set in iax.conf, when they transfer, it does a Dial command and does not use options t, T, h, H, w, W or L. Any idea what I could be doing wrong?

2

asterisk is not a sip proxy. it will stay in the media path at all times. possibly you may want to look @ SER (sip express router).

3

So what I describe above is not possible with asterisk?

I’m a little confused about SER, could you describe how SER would fit in to accomplish what I described?

4

you are looking for a sip proxy,
asterisk takes channels and bridges them. so no matter what happens asterisk will hang on a channel until hangup. it stays as the middle man.

NOW GENERALLY this is not too big of an issue but , obvious scalability problems result if you are running a distributed sort of network.

With sip express router, SER … its more like an oblivious router connecting endpoints and not dealing with any of the signalling.

from the website
In other words, SER is completely oblivious to the fact that calls are connected. So when SER connects a caller to a callee, SER forgets immediately about the call. By doing so SER can handle large volumes of subscribers. The bad news is that SER cannot disconnect a call or otherwise participate in the call.

you can combine ser and asterisk to get a system where the endpoints connect directly and the signalling goes through asterisk,

maybe this will help
voip-info.org/wiki/view/Realtime … th+OpenSER

also look at freeswitch. brian from molten gave a presentation at astricon about using freeswitch also.

asterisk is great but to a certain extent the sip capabilites of it are not the same as a product that is a TRUE sip proxy.

So what I describe above is not possible with asterisk?

I’m a little confused about SER, could you describe how SER would fit in to accomplish what I described?[/quote]