03-07-2018 01:54 PM - edited 03-14-2019 06:01 PM
First off, thanks to the community!
Is it possible to add a second ITSP using SIP (not backup) to a Cisco PCCE environment?
My Network:
What needs to be done:
Previous state:
Current state:
What needs to be done:
What should I look to do next? Are there topics I should read up on? Does anyone know of a guide to add a 2nd ITSP (not backup) for use in an environment?
Thank you,
-Daniel
03-07-2018 01:58 PM
03-08-2018 12:37 PM
Thank you for responding.
I believe our VXML routers are the CUBE routers. They have the DIAL PEER configurations in the configs.
CUCM has the VXML/CUBE routers in separate route groups. I believe this means they are NOT active/active.
I'll post part of my response to another responder in this thread:
"I've made a few changes in CUCM:
Created a new device pool under the existing VXML01 trunk
Created a new Route Group under the VXML01 trunk
Put a physical phone profile into that device pool. THe phone boots, registers. However, it calls out, and receives calls, from the old pool.
Where should I be looking?"
Thank you,
-Daniel
03-07-2018 05:05 PM
If you have your SIP trunks coming in via CUBEs then its easy to route to whatever via dialpeers.
RIght now I have two different PCCE environments running which means two different versions of CVP. All easy to account for via dialpeers on CUBE.
At a higher level I have two different SIP trunks hitting my CUBEs with one set heading to PCCE and one to a different CUCM cluster.
The key thing is as long as you hit CUBE first what you want to do is not hard at all.
03-08-2018 12:33 PM - edited 03-08-2018 12:41 PM
I believe the VXML routers (Cisco 3900 series devices running IOS 15.4) are the CUBE routers. They have the dial peers in the configuration.
I've attached redacted versions of the configuration files. The redactions are an attempt to hide SSH keys/password hashes, and to obfuscate the identity of my employer.
Changes I've made (Note, these are the changes and/or additions).
Note: The ITSP provider is doing both signaling and stream via one channel. the IP is 68.68.117.116
They have provided a test number that starts with a different area code than the blocks I control. I control two four-digit blocks, and several toll free numbers.
class-map match-any VoIP-Ctrl+DNS+SIP match access-group name ITSP2-DNS match access-group name ITSP2-SIP class-map match-any VoIP-RTP match protocol rtp audio ! dial-peer voice 10001 voip description ***Inbound from ITSP2 PSTN SIP*** incoming called-number 720....... no vad ! dial-peer voice 11001 voip description ***Outbound to ITSP2 PSTN SIP; US/N. America Call*** session target ipv4:68.68.117.116 sip-ua sip-server ipv4:68.68.117.116 dial-peer voice 11003 voip description ***Outbound to ITSP1 PSTN SIP; 911 Call*** session target ipv4:68.68.117.116 dial-peer voice 11004 voip description ***Outbound to ITSP1 PSTN SIP; 911 Call*** session target ipv4:68.68.117.116
I've made a few changes in CUCM:
Created a new device pool under the existing VXML01 trunk
Created a new Route Group under the VXML01 trunk
Put a physical phone profile into that device pool. THe phone boots, registers. However, it calls out, and receives calls, from the old pool.
Where should I be looking?
03-08-2018 01:15 PM
03-08-2018 02:52 PM - edited 03-08-2018 02:53 PM
When I say "Old Pool", I mean that when I make a call from the IP phone, calling a 3rd party phone number (Verizon cell), there are no active calls listed on VXML1, as indicated by 0 results on VXML1 using the "show call active voice compact" command.
I cannot see whether my individual call was going out through VXML2, as there are always calls going through on VXML2.
Looking at my CUCM configuration, it doesn't appear that we are using SLRG.
Attached are screencaps from my CUCM configuration. I have made attempts to pipe the new SIP from ITSP2 to the IP phone.
Thank you,
-Daniel Clawson
03-08-2018 03:05 PM
03-19-2018 04:45 PM
Thank you for responding. I've been OOO for several days.
Assumptions from your last post:
RP = Route Pattern
RL = Route List
RG = Route Group
We only have one RP for normal 10-digit dialing. It points to one Partition, and one route list.
The DNA analysis shows:
Does this provide the information you wanted?
Thank you,
-Daniel Clawson
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