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Beginner

On way audio on SIP phones on CME

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

Everyone's tags (3)
4 REPLIES 4

Re: On way audio on SIP phones on CME

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.

 

Source: https://www.slideshare.net/hacktivism/cisco-live-deploying-sip-trunks-with-cisco-unified-border-element-cubevcube-enterprise-2017

 

Rolando A. Valenzuela.

Beginner

Re: On way audio on SIP phones on CME

I am not using CUBE , this is Call manager express

Re: On way audio on SIP phones on CME

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.

 

Ref: https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-overview.html

 

Rolando A. Valenzuela

Beginner

Re: On way audio on SIP phones on CME

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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