02-26-2019 03:31 AM
Hi Guys
I have problem with one way audio with SIP phones to particularly dial-peer.
Calls from SIP to SCCP - works fine
Calls from SIP to PSTN and from PSTN to SIP - works fine
This is part of configuration that is related to SIP phone registration
voice service voip
ip address trusted list
ipv4 192.168.60.2
ipv4 10.36.0.30
ipv4 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol pass-through g711alaw
h323
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server expires max 1200 min 300
This is Dial-peer that is one way audio problem in outgoing direction
dial-peer voice 402 voip
description ### MUP MVPEI ###
translation-profile outgoing MVPEI
destination-pattern 98.....
session protocol sipv2
session target ipv4:10.36.0.30
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec g711alaw
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
When i remove these two of command from dialpeer , no SIP and no SCCP call can not be made to that peer
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
I attachment sip debug
Please Help
02-26-2019 10:02 AM
On way audio usually means that there is a network problem.
Depending on your setup the remote phone needs to be able to communicate directly with your router's IP and/or IP phone IP.
Rolando A. Valenzuela.
02-26-2019 11:29 PM
I am not using CUBE , this is Call manager express
02-27-2019 07:42 AM
Yes you are, just by adding "allow-connections from-type to to-type" you are enabling the traverse features on the router.
So please double check the routing settings to confirm reachability between both phones.
Rolando A. Valenzuela
02-27-2019 11:23 PM
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
I have these lines in configuration and problem is still here
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