07-17-2018 10:50 AM - edited 03-17-2019 01:13 PM
Hello
I'm in a router 3925 with version : c3900-universalk9-mz.SPA.152-4.M4.bin.
Cisco IOS Software, C3900 Software (C3900-UNIVERSALK9-M), Version 15.2(4)M4, RELEASE SOFTWARE (fc2)
I have configurated a dial peer with max-con 120, but the sip trunk is for 180 channels.
dial-peer voice 230 voip
description Incoming-Teleconsultas
max-conn 120
session protocol sipv2
incoming called-number NUMBER
dtmf-relay rtp-nte
codec g711alaw
no vad
When the calls excedd the 120 the nexts calls respond with "503 Service Unavailable" But i need to answer with a "486 Busy Here"
I try to use: error-code-override max-conn but dont have this option.
AS-3925(conf-serv-sip)#error-code-override ?
cac-bandwidth Status code to be sent for max-bandwidth CAC
call Configure call parameters
options-keepalive Status code to be sent for options keepalive
AS-3925(conf-serv-sip)#error-code-override max-conn ?
% Unrecognized command
AS-3925(conf-serv-sip)#error-code-override
Can anyone helpme?
thanks
Solved! Go to Solution.
07-24-2018 12:20 PM
According to the documentation, it looks like you'll need to upgrade to 15.4
Feature Name |
Releases |
Feature Information |
---|---|---|
Configurable SIP Error Codes |
15.4(1)T |
The Configurable SIP Error Codes feature describes how Cisco Unified Border Element provides support for configurable SIP Error codes to override or modify Session Initiation Protocol (SIP) error response codes. The following commands were introduced or modified: sip-header SIP-StatusLine |
07-24-2018 12:15 PM
does anyone had the same problem?
07-24-2018 12:20 PM
According to the documentation, it looks like you'll need to upgrade to 15.4
Feature Name |
Releases |
Feature Information |
---|---|---|
Configurable SIP Error Codes |
15.4(1)T |
The Configurable SIP Error Codes feature describes how Cisco Unified Border Element provides support for configurable SIP Error codes to override or modify Session Initiation Protocol (SIP) error response codes. The following commands were introduced or modified: sip-header SIP-StatusLine |
07-27-2018 08:34 AM
Hello! Thanks so much for your answer, i haven't seen it.
Ill try it first!!
Let you know!!
Have a nice day
07-26-2018 04:29 PM
This can get a little squirrely, but you can try using a sip profile. I have used this to change messaging from one leg to another (changing a response to a message that came in on provider side, went out to cucm, response came back from cucm, this profile translated the response going out to provider).
I applied it on the incoming dial peer from the provider. These are all of the random things I was successful in changing in my lab. I am not sure how this would behave on messages generated by the router itself, you could try it first on the inbound peer, then apply globally if that didn't do it. Both worked for messages THROUGH the router, and inbound dial peer is less intrusive.
voice class sip-profiles 444
response 100 method INVITE sip-header SIP-StatusLine modify "Trying" "Tryingtest"
response 200 method BYE sip-header SIP-StatusLine modify "OK" "BYEOK"
response 200 method BYE sip-header Reason modify "cause=16" "cause=99"
response 404 sip-header SIP-StatusLine modify "404" "400"
response 404 sip-header Reason remove
07-27-2018 07:51 AM
Thanks for your answer, i'll try it and let you know.
Have a nice day
07-27-2018 08:07 AM
07-27-2018 11:10 AM - edited 07-27-2018 11:29 AM
Her, but you are right ; )
I should have signed my name on my post!
Mary Beth
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