05-22-2010 12:37 AM - edited 03-15-2019 10:53 PM
Hi all,
This my first attempt to integrate CUBE and Telco using SIP protocol.
I have CUCM >> H323 GW >> Telco.
now I will configure the GW to work as CUBE. I asked the Telco for the parameters and I have this issue:
they gave me two IPs for SIP one for Signaling and one for Media. How I can distinguish the traffic.
05-22-2010 02:39 AM
Hi
What you mean they gave you two ip address? Are those ip addresses to use on your gateway or you need to point to them?
If you need to point to them, then you need to use the Signalling only. The RTP address will be negotiated during the signalling. So you do not need to configure anything for it. But if you have any firewall or ACL, make sure you permit udp traffic from the media ip.
But if those IP address to use for your gateway. Then you can use sip bind command.
voice service voip
sip
bind {control | all} source- interface interface-id
!
Let me know if this helps
Thanks
- abu
05-22-2010 04:09 AM
Hi Abu Hadee,
I got two IP address for SIP setting and one for IP address for connectivity,
What I am going to do is to configure the gig interface for P2P connectivity.
but inside the dial-peer shall I use the signaling IP address in session target command.
Do you have sample of the configuration?
05-22-2010 06:39 PM
Hi,
I'm still confused if you have to use all these IP address for your gateway or one for your gateway, other two for remote
lets say you've received following IP Address.
1. A.A.A.A for your GW conectivity
2. B.B.B.B for SIP Signalling
3. C.C.C.C for meida
Here is scenario 1, say you've to use one IP for your GW and other two for remote
interface gig0/0
ip address A.A.A.A 255.255.255.0 !
!
voice service voip
sip
bind all gig0/0
!
dial-peer voice 1 voip
destination-patter 9T
incoming called-number .
session-target ipv4:B.B.B.B
codec g711ulaw
no vad
!
C.C.C.C you do not need to configure anything in the voip side. As this one will be media ip and will be negotiated autometically. All if you do permit any UDP sourced from that.
Option two: If you need to configure all three in your gw
interface gig0/0
ip address A.A.A.A 255.255.255.0 !
!
interfae loopback 1
ip address B.B.B.B 255.255.255.255 !
interfae loopback 1
ip address C.C.C.C 255.255.255.255 !
voice service voip
sip
bind control loopback1
bind media lookback2
!
dial-peer voice 1 voip
destination-patter 9T
incoming called-number .
session-target dns:
codec g711ulaw
no vad
05-22-2010 11:04 PM
Thank you too much Abu Hadee,
I will test it and if I have any problemI will return to you.
07-14-2010 12:08 AM
I tried the configuration above, I can now place calls but I have some issues:
1- When somebody call from outside to IP phone ( one of the DID extension) the calling party always hear ringing even if I pick up the phone and answer the call. I contacted the provider regarding this and they said we are not receiving any signals that some body answer.
2- It is a DTMF issue. if I called from IP phone to out-side and AA answers the AA does not recognize the DTMF. the provider inform me that they are using in-band signaling, so shall I delete the dtmf command from the dial-peer?
09-27-2010 10:26 PM
Hello,
I'm not sure if you ever figured this out, but the answer is the "dtmf-relay rtp-nte" command, which encapsulates the dtmf signals within RTP. It takes an MTP resource, generally speaking.
Ryan
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