IOS-XE SIP Dial-Peer configuration on CUBE for Call Routing to ITSP
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- Опубликовано: 7 сен 2024
- #DialPeer
In this Video, we are going to learn about the SIP Call Routing IOS Configuration using SIP Dial-Peers. We will see what are the LAN Dial-Peers and what are WAN Dial-Peers. We will also talk about the new SIP Dial-peer features that could optimize the Call Routing to the ITSP.
We will be testing the Translation Profile and see how can we Troubleshoot basic Dial-Peer issues.
Thanks for sharing this valuable information.
Thank you for nice class bro
Hello TrainCollab Team see your videos on cisco phones they are very useful and clearly and in detail explained but if you answer me one more question I have received 4 numbers from the provider and I do not know how to toggle the setup to cisco Cube even more briefly please if you give an indication! with you was the Sunnat of Uzbekistan
Hi Amit, There are few mistakes. In beginning of the video you said LAN inc DP is 1000(from CUBE to CUCM) & outgoing DP is 1001(From CUCM to CUBE) & WAN Incoming DP is 2000 & WAN outgoing DP is 2001. But when you configured those DP in the router i think you have interchanged it. And also in the end for the "WAN OUTGOING DP from CUBE to ITSP" you have bind the wrong interface Gi 0/0/0 instead of Gi 0/0/1. Hope you can put some comments to correct this.
yes there was a confusion while recording. but it is just the tag so it’s unimportant.. just to clarify.. LAN outgoing dial-peer is CUBE-to-CUCM. LAN INCOMING DIAL-Peer is CUCM TO CUBE. 1000 is outbound DP. 1001 is Inbound DP. 2001 Inbound WAN DP and 2000 WAN Outbound DP. Tag as I said can be anything. But stay tuned for tomorrows video you have all inside them. ;) About the binding you are bang on!! It was on purpose.. I kept it for tshoot.. You will see it in the next video ;)
Ok I see your 4 videos. I have two question ... 1) The last dial-peer 2000 the source int should be gi 0/0/1 instead 0/0/0 right? and 2) That interface gi 0/0/1 is the one you call WAN and in that physical port is where you are going to connect the Ethernet Port that your provider give you?
I think both your points are correct
In part 2 you enabled SIP OPTION Pings in the profile for the trunk and at timestamp 33:30 you indicated that the trunk is still down because the CUBE has not yet been configured.
But here I missed the corresponding configuration in your dial-peer: "voice-class sip options-keepalive"
That’s a Good question. I will try to explain it in short. So options ping on CUCM is a mechanism by which you can avoid the issue related to 3- way handshakes mechanism needed by cucm to send a call to the destination IP mentioned in the SIP Trunk and incase the first IP isn’t reachable it can use the alternate Ip/trunk. OP on CUCM also enables the SIP Trunk statistics like Trunk is up or down and the Time since when the Trunk is up. Had I configured the dial-peer in my part 2, that points to CUCM IP address, my Trunk would have just registered without OOD SIP Options keepalive config on the dial-peer. So in short OOD Keep alive is not needed on the Dial-peer for, CUCM SIP Trunk registration. I hope that helps.
how to i configure my voip gateway to support IVR for my CALL Manager for numbers originating from the PSTN?
Hi....What could be the cause of the trunk on CUCM side not coming up?
Can you make a video with the more recent Cube Version covering Multiple Pattern Support on a Voice Dial Peer
Yes, it is on the pipeline. Especially just the dial-peer section.
some sip providers offer authentication by the IP your coming from. is it possible for cucm to authenticate with those sip providers without CUBE using direct cucm
Yes it is possible via a SIP realm. However that is not a recommended way to setup your SIP Trunk with the Service Provider as you expose you VoIP network and additionally you would lose the SIP features that might be needed by the Enterprise and it is not natively provided by CUCM.
Sir i have a question. How can I specify the source SIP trunk port on the voice gateway. For example, I want to force the gateway to always send the SIP invite using 5080 port.
JAQIE, there are 3 ways that might help you. First, for a global change you can use connection-reuse command under sip-ua. At a tenant level you can specify it under voice class tenant. At a Dial-Peer level you can specify it using the SIP Profiles by modifying all the requests.
Hope this helps.
@@TrainCollab Thank you so much. it would be so nice of you if you please tell me what specific commands I can use at Dial peer level to specify the originating fixed SIP port for SIP/SDP invites?
"destination-pattern DEF " what DEF stands for here? We have CUBEs which sends call to contact centre have dial-peer with this command. Thank you.
This is just a place holder for the dialpeer to be active for call routing.
@@TrainCollab Thank you :)
hi , can you please guide me on the configuration if I have two different cucm clusters and a cube in my lab and I want to route calls between the cucm's using the CUBE ..can you please help me with the config..
Create respective dial-peers and create SIP Trunks from CUCM’s to the CUBE. We do have some specific lab scenarios on www.traincollab.com lab rentals.
Actually before the dial peers I want to know about the basic config like do i use single interface on cube for both cucms? Or one interface for one cucm and one for other?
@@prabhatbhalla doesn’t matter as far as you have routing from the single interface of the cube to both the cucm’s you can just use single interface. Else 2 interfaces.
Sir your voice is feable , facing problem to understand
Very Poor Config , specially in Dial-Peer , he did it in long way around , aka , he didnt segregated the Dial-peer , seems due to lack of to his poor Experience so that's why i dislike the video
Regards
Ali
I am happy with your feedback. Would be great to let me know the improvements so that the other viewers could also learn. Did you mean aggregate the DP's? I understand with the new Dial-Peer features you could aggregate the DP's from 4 to 2. However this series is about helping people (especially beginners) understand the call flow and then at a later stage explain the advanced config. You will have to come back to watch the advanced CUBE/Dial-peer config and features. I hope this helps. ;)
Too much aggregation can be more worst. I think a separation of incoming and outgoing dial-peers gives a better overview and troubleshooting.
Clearly the video follows the line of basic settings for beginners. To say that it is poor and not to see the context in which the author has created the video, your opinion is valid for a video where we expect optimizations and advanced settings.
Excellent video.
Ali thanks for your feedback, but TrainCollab did his best but where is your advance configuration?