I have a CM6.3 and SMGR/SM 6.3. I created a SIP user in SMGR 1234@avaya.com and assigned a CM profile to it as an xmobile type. I then launched my One-X Mobile SIP client and registered with no problem. I know there's some new features with dual registration. Is it possible to have an h.323 deskphone with this same SIP extension so I can get calls on either the deskphone or the mobile client?
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One-X Mobile SIP CM6.3
- Thread starter wpetilli
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Bear with me here to give a quick summary of how I’m testing this.
I have 1 SMGR that has entity links to 3 Session Managers and another 3 CM’s. The CM’s are separate evolution servers and are linked to each other via h.323 trunks for 4 digit dialing. Our dial plan is in CM and not SM. The dial patterns for each extension range are setup in SM and pointing to the correct CM entity links, but we send no traffic up to SM. We keep everything within CM / h.323. Each CM does has a SIP trunk to each Session Manager. The CM I’m testing this one-x mobile SIP/dual reg feature is a 6.3.5. This is the only CM I have synchronized to SMGR, since it’s a lab.
I did create the station in SMGR as 96xxSIP, was able to login to the One-X Mobile SIP client and was able to simultaneously log into a 9630 h.323 deskphone. From my production CM deskphone I dialed the 4 digit extension created on the lab system. I have an aar entry for this extension to route over the trunk to SM. In SM I have a dial pattern for this extension pointing to the lab CM as a destination. The call rang fine on the mobile client, but not on the deskphone. If I log out of the mobile client and repeat this test, I get fast busy. The traceSM comes back with unregistered user. I must be close here, but am somehow missing a steps. I thought both phones are supposed to ring through. And at the very least if it cannot find the registered user in SM, it should have followed the dial pattern down to the CM, which in theory would have found the h.323 station.
I have 1 SMGR that has entity links to 3 Session Managers and another 3 CM’s. The CM’s are separate evolution servers and are linked to each other via h.323 trunks for 4 digit dialing. Our dial plan is in CM and not SM. The dial patterns for each extension range are setup in SM and pointing to the correct CM entity links, but we send no traffic up to SM. We keep everything within CM / h.323. Each CM does has a SIP trunk to each Session Manager. The CM I’m testing this one-x mobile SIP/dual reg feature is a 6.3.5. This is the only CM I have synchronized to SMGR, since it’s a lab.
I did create the station in SMGR as 96xxSIP, was able to login to the One-X Mobile SIP client and was able to simultaneously log into a 9630 h.323 deskphone. From my production CM deskphone I dialed the 4 digit extension created on the lab system. I have an aar entry for this extension to route over the trunk to SM. In SM I have a dial pattern for this extension pointing to the lab CM as a destination. The call rang fine on the mobile client, but not on the deskphone. If I log out of the mobile client and repeat this test, I get fast busy. The traceSM comes back with unregistered user. I must be close here, but am somehow missing a steps. I thought both phones are supposed to ring through. And at the very least if it cannot find the registered user in SM, it should have followed the dial pattern down to the CM, which in theory would have found the h.323 station.
OK... couple of things
You do not need dial patterns to route calls to stations registered to SM.
To say, I have SIP 1234 registered to SM. Station 1234 in CM has off-pbx-station mapping of 1234 to AAR to a route to a trunk to SM.
Anyone calling 1234 goes thru CM first. CM will ring the H323 station and the off-pbx station - as if it were EC500. CM will ring SM about it and the SIP phone will then ring.
All of the above is accomplished exclusively through application sequences in SM - not dial patterns.
So, if I'm understanding you correctly when you say...
and
Then I think what you're saying is that when the test station is dialed in CM from an analog phone for example, that you are shipping that call via AAR to SM and not processing the call through that station in CM first and using off-pbx station-mapping to get the first invite up to SM.
I hope that helps more than confuses
You do not need dial patterns to route calls to stations registered to SM.
To say, I have SIP 1234 registered to SM. Station 1234 in CM has off-pbx-station mapping of 1234 to AAR to a route to a trunk to SM.
Anyone calling 1234 goes thru CM first. CM will ring the H323 station and the off-pbx station - as if it were EC500. CM will ring SM about it and the SIP phone will then ring.
All of the above is accomplished exclusively through application sequences in SM - not dial patterns.
So, if I'm understanding you correctly when you say...
and
Then I think what you're saying is that when the test station is dialed in CM from an analog phone for example, that you are shipping that call via AAR to SM and not processing the call through that station in CM first and using off-pbx station-mapping to get the first invite up to SM.
I hope that helps more than confuses
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- #6
So essentially what you're saying is when the CM-B station wants to call this CM-A station (that has a logged in h.323 phone and SIP Mobile at the same time) that the AAR entry in CM-B should not go via the SIP TG to SM and instead go via an H.323 trunk directly to CM-A, with the off-pbx entry in CM-B going up to SM 2nd?
OK, you should be able to send the calls between CMs via SIP.
You just confused me with your statement...
I understood that the SIP trunks in CM/SM were for SIP stations to register to and get off-pbx station calls out of CM to SM.
Have you set up many SIP stations before on SM? Typically there's something called "application sequencing" that's the thing that makes SM get a call for 1234 from another CM or PSTN SIP trunks or whatever, SM sees 1234 and instead of shipping it to station 1234 registered to it, it will see that in the Session Manager profile for that user that there is an "application sequence" that goes to CM1. SM would then send the call to CM1 to process through your station - like if you were cfwd'd to voicemail or something - and then CM uses off PBX station mapping to send 1234 back to AAR up to SM, but in the SIP message this time there's something to indicate its the "terminating application sequence" and then your phone rings.
All that to say when your H323 phone is ringing, it should go thru off-pbx station mapping to ring your SIP phone, and your SIP phone should
You should see this in the invite of a 1234 going off hook: avaya-cm-fnu=off-hook - and it should be 1234 sending an invite to 1234@yourcompany.com to CM to originate its application sequence.
Calls terminating to your SIP phone should have a SIP header like this:
Route: <sip:10.10.10.10;transport=tcp;lr;phase=internal_terminating;seq=explicit;remote-ure-route=true>
You just confused me with your statement...
I understood that the SIP trunks in CM/SM were for SIP stations to register to and get off-pbx station calls out of CM to SM.
Have you set up many SIP stations before on SM? Typically there's something called "application sequencing" that's the thing that makes SM get a call for 1234 from another CM or PSTN SIP trunks or whatever, SM sees 1234 and instead of shipping it to station 1234 registered to it, it will see that in the Session Manager profile for that user that there is an "application sequence" that goes to CM1. SM would then send the call to CM1 to process through your station - like if you were cfwd'd to voicemail or something - and then CM uses off PBX station mapping to send 1234 back to AAR up to SM, but in the SIP message this time there's something to indicate its the "terminating application sequence" and then your phone rings.
All that to say when your H323 phone is ringing, it should go thru off-pbx station mapping to ring your SIP phone, and your SIP phone should
You should see this in the invite of a 1234 going off hook: avaya-cm-fnu=off-hook - and it should be 1234 sending an invite to 1234@yourcompany.com to CM to originate its application sequence.
Calls terminating to your SIP phone should have a SIP header like this:
Route: <sip:10.10.10.10;transport=tcp;lr;phase=internal_terminating;seq=explicit;remote-ure-route=true>
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- #8
We are 100% H.323 here, so this is the first SIP station. The only thing we use SM for is VM. CM pilot number routes via SIP Trunks to SM with the destination entity as the VM. The dial patterns were put in place so VM has the destination CM for MWI. All other call traffic between offices routes via h.323 trunks. We also still use ISDN PRI for PSTN, so no SIP trunks externally.
I'll putz around with the app sequencing and apply to this station. Thanks for the guidance.
I'll putz around with the app sequencing and apply to this station. Thanks for the guidance.
Fair enough. So, you want to look in "Session Manager", in "Applications" and define the CM and SIP entity to that CM as an "application" and build an "application sequence" that includes that "application" you built.
After, in the "User" "Session Manager Profile" you assign that app sequence as originating and terminating.
Your call flow would be...
CMB-->INVITE:1234 to SM -->CMA -->ring station H323 (including off pbx station mapping) -->SM as termination sequence (INVITE:1234@SM) -->ring SIP set
The difference between the 1st and 2nd invite to SM for 1234 is that the first one is from CMB and does not have "phase=internal_terminating" in it - so SM looks at dial patterns to route down to CMA.
The 2nd invite has that "terminating" message in it, so SM does NOT look at dial patterns but instead looks at SIP phones registered to it.
Have fun!
After, in the "User" "Session Manager Profile" you assign that app sequence as originating and terminating.
Your call flow would be...
CMB-->INVITE:1234 to SM -->CMA -->ring station H323 (including off pbx station mapping) -->SM as termination sequence (INVITE:1234@SM) -->ring SIP set
The difference between the 1st and 2nd invite to SM for 1234 is that the first one is from CMB and does not have "phase=internal_terminating" in it - so SM looks at dial patterns to route down to CMA.
The 2nd invite has that "terminating" message in it, so SM does NOT look at dial patterns but instead looks at SIP phones registered to it.
Have fun!
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- #10
This exercise is probably going to fail for me because I'm doing this testing all internally, so if the call hits the Off-pbx step it won't be able to dial outbound to an external number. I guess I can create a 2nd SIP extension w/o app sequencing applied and do the off-pbx via aar to the SM.
Nono, it should work for you.
"change off-pbx station mapping" and for extension 1234, you enter "application" as "OPS" for off-premise station.
Trunk? is AAR, and 1234 in AAR picks a trunk to SM
That's how CM knows to route calls for SIP stations through Session Manager.
You don't need PSTN or outside world trunks to make that work - just a single trunk between CM and a SM is all that's required.
"change off-pbx station mapping" and for extension 1234, you enter "application" as "OPS" for off-premise station.
Trunk? is AAR, and 1234 in AAR picks a trunk to SM
That's how CM knows to route calls for SIP stations through Session Manager.
You don't need PSTN or outside world trunks to make that work - just a single trunk between CM and a SM is all that's required.
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