SIP gateway select E1 line

bo liu · ‎09-10-2015

hello everyone

i want to known,can the gateway select e1 from 0/0/3 0/0/2 0/0/1 0/0/0 when the call outgoing to PSTN

and select e1 form 0/0/0 0/0/1 0/0/2 0/0/3 when the call from PSTN

and if the internal phone send invite to gateway ,but the gateway to E1 need about 20s ,how can i change parameter

sip-ua

retry invite 6?? default is 2...

Jaime Valencia · ‎09-10-2015

You control outbound via your config, MGCP or H.323, configure accordingly

You have NO control whatsoever over inbound, your telco controls that.

HTH

java

if this helps, please rate

Mohammed al Baqari · ‎09-10-2015

Hi,

I don't think even the second requirement is doable. You can't configure the gateway to hold processing INVITE message for 20 secs.

What you can do is to configure a dummy dialpeer with preference 0 (first selection) and make the session target point to unreachable addresses. Then have your pots dialpeer with preference 1. Finally, change the INVITE timeout to x secs. This will make the gateway wait for x seconds before sending the call to E1 which is second dialpeer.

However, 20 seconds is high value and the caller will timeout before dialpeer failover (unless you change too many parameters at the caller side as well).

In short, I don't recommend at all doing this unless there is a priority requirement.

bo liu · ‎09-10-2015

i ask this question is

i have a gateway, 10 E1 on it and a third party PBX connect with the gateway use SIP.

now i find that when the PBX send more call to the gateway,the gateway will send sip 500 error,

but the E1 just used 75%.

i use cucm to connect the gateway use SIP trunk,and i change the sip-ua config find when i dial a call after 20 secound i can hear the ringback tone.

and the E1 select is 0/0/0 - 0/1/1 is full ,and another line just use a little.

the CUP use 99%....

avinsrid89 · ‎09-10-2015

Inbound port selection cannot be controlled. Telco will send the call to the line they wish.

Outbound calls can be controlled to appropriate voice ports through dial peers.

Please rate useful posts and mark them resolved when needed.

~Avinash