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The end devices do, in both directions. I mean, the invite is coming from the Session Manager, are you saying it just passes on whatever it gets for SDP's? What if it is anchoring the media and it's going out to the PSTN? (or in my case another switch in my office, yes I have a few) Check out...

Greetings! I've got a SIP line that I have TLS working to, and I can get regular RTP working to, but I'm trying to get it to use SRTP. I've set up the codec's in my CM - The SIP line is connected to the SM, so I was wondering is there anything I need to check in the Session Manager to get...

@Janni78 - I have traces from both and just can't seem to figure out the problem spot myself. Any suggestions on the best places to look or what to look for? Check out my professional profile and connect with me on LinkedIn. http://lnkd.in/S4JdV3

Yeah, I was afraid of that. This is for a ACD integration, and I'm trying to figure out the difference for the customer. :( Check out my professional profile and connect with me on LinkedIn. http://lnkd.in/S4JdV3

I thought those settings might play in as well . . I've tested it with them on and off and every combination, and on my working lab system I can't break it. On the Non-working system changing them can't get it working. :( Check out my professional profile and connect with me on LinkedIn...

I have two IPO switches that DON'T work that way, so it makes me think there is a setting for one way or the other. These are 3rd Party SIP endpoints, just to clarify. I have a lab system that I can call out from the same line twice without the distant end answering either call and it is treated...

Greetings all, I'm having a strange problem that is baffling me. I have a 3rd Party SIP account, if I call outbound to a number, let it ring on the distant end and try calling again to a different number (or the same, doesn't matter) the second call gets a 503 Service Unavailable back. Now, if...

It appears with those settings, when we pick up the handset it still goes off hook on key 0. I would have assumed that "Idle Appearance Preference" set to yes would then ask you what your preference is? Check out my professional profile and connect with me on LinkedIn. http://lnkd.in/S4JdV3

How do you set what line on a multi line phone is your default line when going off hook? Thanks! Check out my professional profile and connect with me on LinkedIn. http://lnkd.in/S4JdV3

@KCFLHRC, that is exactly what I did; Hunt 1 - 30, then 30 HUNT'ed to a ACDN, which NCFW to port 31 then HUNT straight through to 60, then kicked to an ACDN that was NCFW to 61, HUNT'ed to 90, which went to an ACDN that NCFW to 91, HUNT'ed to 120 which kicked to an ACDN with a NCFW to 121, and...

All - I found my answers here - thread798-1307493 Thanks GHTROUT!!!! Check out my professional profile and connect with me on LinkedIn. http://lnkd.in/S4JdV3

I have servers that are connected via Dialogic Cards to the CS1000. It is for a 3rd party ACD system. I want callers to come in and possibly stack up 136 on the system and it to keep hunting to the next one if the current one is busy. So, I want the function of the HUNT but I want to go higher...

I have 136 ports that I need to "HUNT" to each other, I tried using the HUNT feature at first, and after a lot of troubleshooting finally found out there is a 31 max there. What is the max in creating a HUNT GROUP with the below instructions? Could I add all 136 ports? If I can't is there a...

Well I got that setting changed to Y, but I still have to dial the AAR FAC to call from the end device, back through the Core CM, then out to a SM SIP trunk to Asterisk. Any ideas on how to remove this. Check out my professional profile and connect with me on LinkedIn. http://lnkd.in/S4JdV3

Man I feel like an idiot, I found the problem. If I dial the AAR FAC and THEN the extension I can call through to the numbers I was previously unable to. So, I see this setting - "ARS/AAR Dialing without FAC? N" under display system-parameters customer.... however I am trying to figure out how...

Kyle555, I have inbound dialing from a SIP One-X phone through a CORE SIP trunk to an end device. I can't call from that end device back to the SIP one-x phone though. Also I have a SM SIP trunk out to an Asterisk switch. I can call from the Asterisk switch to the SIP One-X phone as well to the...

When I do a trace on the TAC here is the denial event I am getting no matter if it is a internal number or something that routes externally; 15:34:53 denial event 1166: Unassigned number D1=0xa023 D2=0x201 (I don't know if the D1 or D2 are numbers I can look up somewhere?) Check out my...

Thanks Kyle! I got inbound working I am now trying to get outbound working, so 4298@domain.com is coming into the core, but can't call a one-x phone off the core or a number that routes out a SIP trunk to another switch. In tracing the SM I don't ever see the call ever make it to the SM so I...

Well, I figured out part of it. I went into SM and under User Management made up some dummy information for a user and assigned them the 4 digit number of 4298. Now, when I call 4298 from an external phone it is passed through SM to the core and then out the trunk to the device I am using. Now...

Greetings all, I have a newly installed CM6 set up in my test lab and I am trying to get something working. I have a SIP trunk coming off the CORE going to a local computer device. I am able to dial the number (4298) I have routing down that trunk internally from the CM6. But, when I try to...