sip trunks in cli
Hi,
I have a TB 2.4 where i setup a lot of sip trunks and sip extensions. With CustomContexts a restrict for every extension to use just one sip trunk (for example, extension 101 use for callout just one trunk, say 823101). For inbound same as for out, is used one extension for one trunk. When i make in cli the commad "sip show channels" i see something like:
for out call:
xx.xx.56.12 079992282 690c8dcb301 00103/00000 0x380100 (g729| No Tx: INVITE
10.10.1.105 101 29889194111 00101/00002 0x100 (g729) No Rx: INVITE
and for incoming call:
10.10.1.105 101 7fd376ab335 00102/00000 0x380100 (g729| No Tx: ACK
xx.xx.56.12 823129 51D00F69-D8 00101/00101 0x100 (g729) No Rx: ACK
xx.xx.56.12 is my sip gateway;
823129 is the last trunk in the trunk list.
And for ALL incoming calls cli's "sip show channels" say that they come with 823129. Anyway, the call is incoming via the restricted trunk, and all work very fine.
I tried to disable CustomContext, but effect is the same, so this is not a bug of CustomContexts.
The question is what this mean? is a bug or missunderstood? Can i configurate to see the real coresponding trunk? Maybe i need more RTFM, then please say what to read.
Tnx.
P.S: Sorry for my poor english.
P.P.S: If more configuration detail or cli show needed, please say and i will post here.
Anyone?
So, is this a normal stuff in the asterisk, and/or can be what re-configured so when I make an cli's "sip show channels" to see the real trunk used?
Yesterday I make some tests, and I saw what all inbound calls comes in via one single sip trunk (the last in the list of trunks). So, again the same question, is this ok for asterisk/freepbx and can i do anything to make the inbound calls to come in via the coresponding trunk(calls for extension 101 to be received via trunk xxx101 etc)?

Member Since:
2007-09-10