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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...

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...

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

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...

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...

I'm just trying to figure out where there isn't a category for the Genband A2, be it inskin with a 2100 or the standalone EXPERiUS platform. If there is I must not be looking in the right place because I can't find it! Thanks!!! Check out my professional profile and connect with me on...

All, I am looking for a recommendation for a Softphone H323 application that is not an Avaya brand. I would prefer a freeware version or something that has a demo I can use, but I wouldn't be opposed to buying one if it is was worth it. So, basically looking for any recommendations anyone has...

Greetings All, I have what I think should be something fairly easy here, I am just trying to figure out the cleanest way to do it. I want to route anything dialed besides a 44xx extension out a SIP trunk (the only numbers configured as extensions are 44xx numbers). Then let the other switch...

All, I have been bashing my head against a wall for some time now so I figured I would reach out for help. We just got a test system in house and I am trying to get it configured with no luck. This thing is VERY basic, just an IPO with 2 9608 phones off of it. For some reason, when tracing I...

This is probably a dumb question, but we have removed the PRI from our CS1000 and went SIP with it to a third party switch. I have the CS1000 talking to the third party switch via a SIP trunk, so all is well there, I am just trying to find where you re-route the 9 plus digits trough a DSC to my...

I am having a strange problem that I have WireSharked and I can't figure out what is causing this. I have a Unistim phone and when I call from my cell phone, via my SIP trunks to it, I get DTMF just fine. When I call from a TDM phone on the same switch though, I am getting no DTMF being passed...

All, I did some searching but I still have one question. I have a few users with IP phones, that have CFW on them. Their IP phones are currently not plugged in and they had CFW activated to their cell phones. Well we had to reboot the switch last night and that forwarding got turned off. Is...

Currently when we register a SIP phone to the CS1000E 7.5 it sets a timeout of 1 day, (86,400 seconds) before it needs to register again. This is causing problems if people try to register again because they lost connection over a VPN or whatnot, and it ends up having multiple registrations. Is...

All, Trying to do some testing with our ShoreTel system and can't seem to get Presence to work. I am about to use both PortGO and X-Lite to get a SIP softphone to connect. I can do inbound and outbound calls. I can't seem to figure out how to get Presence to work. Any tips? Thanks! Ryan

All, I am trying to wire up an Attcon here in the office for some testing. We USED to have one wired so I don't think it will be too hard. I have it wired through and the m2250 is powering up and display lights up. Now I try dialing 0 but it doesn't ring there. I am trying to figure out how to...

All, I have a few phones running SIP in here for some testing and right now all the DTMF is out of band, in accordance to RFC 2833, which is good. BUT, I want to set them to be in-band for some testing and don't know how to make them that way. I saw a reference that said the phone supported it...

All, I ran some searches and couldn't find what I was looking for. I am trying to figure out what the idle timeout is for a login session? I am running some LD 80 TRAC's to a log file and each time I get logged out for inactivity, I have to go back in and reconfigure my trace. I am trying to...

All, I am trying to troubleshoot something in my CS1000E and I am stumped. I actually want to turn VAD on and get comfort noise working. I went into my Node and Voice Gateway (VGW) and Codecs, and for Codec G711 I checked the box for Voice Activity Detection (VAD) and then sync'ed my node and...

Ok, long story short I tried to SSH into the CS1000 today and couldn't. So, I got on my COM port directly connected to the switch and the login works, so I am thinking it is some crazy expired in UCM but not the switch, (which I have seen before). So, I tried changing the password in EM, then...

All, I see this topic has been brought up a few times here and it seems every time it gets dropped. I am trying to configure my SIP trunks from my CS1000 to Cisco to pass DTMF presses. I have trunking working and can get calls to go in both directions. I can get DTMF to pass from my Cisco to the...