In a dash board type manner, one can see the status of various communication channels for an individual. One can see the likelihood of reaching an associate by phone, IM, or video, or how responsive they are likely to be to. Work Mode information is passed along indicating whether an individual is working in the office, at home, is mobile, or is traveling. AES 4. IM - initiating IM session will launch will shift control to Office Communicator client into a conversation window of designated contact.
Users of Apple devices can make better use of their desktop. Flexible modes of operation enable the user to select a voice path via VoIP using the computer for audio or via an alternate voice path such as home phone or mobile phone. Regardless of mode selected, complete call control is achieved on the Mac. User benefits of convenience, flexibility, productivity, and cost savings can be realized.
End users can use one full-featured desktop or handheld device for most of their communications needs. Compliments their communication devices by allowing customer to natively take advantage of desktop device, but when choose to can go mobile but still be a part of enterprise community. Rich communications activities can take place from a Mac desktop or laptop via VoIP direct from the device, or a desktop or mobile device user chooses audio path. Additionally, provides an opportunity for associates to work from anywhere as convenient, improving employee moral by giving them the flexibility they need.
Ability to do more with more devices and be just as productive no matter the device. The intuitive user interface makes it simple to be available for rapid client response reducing missed opportunities and giving the ability to engage from virtually anywhere. Cost savings. Flexibility enables execution or creation of a business continuity plan, streamlines real estate requirements by enabling remote work. Remote work potential also extends the potential workforce available by extending beyond the local geography.
How does it Work? Voice SIP soft phone with voice path via VoIP using the computer for audio or via an alternate phone such as home phone or mobile phone Avaya Inc. Orders are placed using standard mechanisms e. This client is only available a la carte. Licenses are available a la carte, as an entitlement with Avaya Aura Enterprise Edition and the UC All Inclusive offer not including video calling feature. Your problem is most certainly in administration somewhere. Both on Sip security module and http. A call is able to be made now, but that error is there.
The codec set is set 1. If it wants a DSP first and you don't have DSPs that can process or for example, that might be the cause of a termination error. Maybe a denial event logged against the calling or called station might give some "bearer capability" problem? I see all clean 's all the way and I'm able to make calls, but I still have the red error about voip service limited. I'm calling 4 digit between CM's across totally different SM's and am making outbound pstn calls via the cm pri, no issue.
Just have that red error and it's not saying what it is. I installed the windows client on my office pc and it prompted me for a certificate. I installed the SM identity cert and it logged in straight away. Our root and subca's are already part of the windows build. I then went back to the 46xx file autoconfig method for my ios device and installed the identity cert and put the root and subca in as well, even though those are already in the device trust store.
The error cleared. So, not sure if it needed a cert in the private store and which of those it needed. Theoretically, the root and subca in the device trust store should have been all it needed. Unless there's some other setting the device is getting from the 46xx file that it likes that it can't get from the manual config method I've been doing. Still need to sort that before a prod rollout. Moving along.. Of course no information in the message. I see the traffic from the endpoint hitting the B1 interface, but it's dying in the SBC somewhere.
Avaya one-X Deskphone H Release Readme | pyxykaze.tk
We also see nothing on the SM trace. Is there any sort of static route that needs to be put in for the A1 and B1 subnets to be able to talk? That's basically what you get if you haven't configured the SBC correctly. It's not intuitive. Either that, or options pings from SM to SBC or vice versa fail and they don't see each other in service despite a correct configuration.
But that's unlikely. Any SIP level response is fine - forbidden, not found, wtv - if they send a sip message to one another and get any type of reply, it considers the SIP ping successful and keeps the link up. No, no incidents. Same with B1. Since these servers are in the dmz the security team typically blocks ping unless it's necessary for a function. Things like the ports for dns are open, but not ping. They're going to allow ping just to see if is the issue, but they don't believe so. Where is that connection? It's gotta be sbc config related if it's not passing along the remote worker flow and gives a sip - you know the certs are right.
Go thru the remote worker devconnect doc and doublecheck I guess. I've gone line by line of the doc and a stare and compare to a previously working lab POC and things from a config standpoint look spot on. What I am seeing and not sure if it's the culprit, or how to determine from what interface it is.. I said the SBC sucks with external certs and I wasn't kidding. Yea, I had already whitelisted everything possible in SM. But that's all on Client connections never hit SM.
- mp3 in stukken knippen mac!
- 2016 Avaya Inc. All Rights Reserved..
- Avaya one-x SIP software for Mac OSX - Spiceworks?
For the cert stuff I seem to get a clear connection into B1. The capture you recommended looks like connection resets after the tcp handshake. Don't even see the SSL handshake. If you that on a tshark right on the wire, I'd be curious to see what you mean by the "packets are able to be read". I mean, there's something readable in there, but it shouldn't be SIP signaling or anything relative to your remote worker flow.
So, either you goofed and told the SBC to relay unencrypted stuff on - just telling it to use doesn't explicitly pop a TLS profile in there - or, you're somewhere in limbo with how it handles certs. It shouldn't be hard on the inside - the SBC just needs to trust what SM offers, that's an easier thing to do on it than configure a cert for it to offer endpoints, but still.
Try default certs on the inside part? I mean, not trying to start a handshake is weird. I've complained probably a few times in this thread and in others about how bad the SBC is at administering certs on - where it won't do what you tell it and won't let you know.
Do you think you've hit that part? I can give ya some oneliners to import certs via CLI, but its tedious, needs to be done on each SBC and is basically you throwing in the towel on the admin interface. Patch to latest? Open a ticket at Avaya if you've got the support? If my SM identity certs have been replaced with my PKI certs is that going to help with the default certs? Forgot why the SM didn't like the chain, but.. In any event this is a cert issue after all. The traceSM pumped the fatal cert errors about an unknown CA, which doesn't make sense.
Since both are cut from the same. Web browsers don't validate intermediate certs, the SBC does. Then, i can't remember if cat'ing many pems together of them, or converting to a pkcs7 might be needed, but suffice to say, whatever SM is offering the SBC, it doesn't like. For no other reason that to prove the TLS handshake. If that's the case, then you know the SBC isn't accepting what you've administered and I can give you some pointers on that.
The SM has these as separate certs in its trust store. I'll mess with that depth setting.. I saw it set for 1. I'll try pointing at CM. Winner winner chicken dinner.. Testing client from externally and now getting fault messages during that application exchange. Going to try bouncing the service on SM. Does it use a settings file?
- Ask a Question.
- Send Feedback.
- Related titles!
- Avaya One-X Communicator;
- uncle mac all things bright and beautiful.
There's something for sourced ppm proxy server. When I attempt a connection now, the error on the client I now get is invalid ex or password which I know is correct. The SBC incidents show a 'no subscriber flow match'. I rebooted the bloody thing and it works now. This was by far the biggest nightmare of an install.
Still have much to do, but this was the major piece. Thanks for help and patience.. Our initiative is remote workers using iOS and hard phones. The idea was to not deploy anything to the iPhones as long as the Avaya servers had our CA in its trust store. We are going with the method that even internal folks will go up to the SBC to avoid complicating things. For ASM: we replaced the identity certs with our own same cert in all containers. I get successful registration with the iOS devices and full functionality. If a user leaves the firm they just wipe the policy via MDM and that phone becomes disabled.
In the 46xx file for these phones.. I get successful registration. The confusion I have is based on this setup are these secure from a security standpoint to put out in prod and if not what did I miss? The other part is around the hard phone revocation process. That would toss a monkey wrench in this whole process and I need to be able to do that. Good on ya for sending everyone in or out thru the SBC. Your root CA cert is not private or under your control.
When you mean mutual auth on the outside that's the setting requiring peer verfication? My company website, per your example uses a verisign cert. Is there a way to force mutual only for deskphones and leave the ios as is? To say, the SBC being configured to offer a cert your devices trust and demand the devices provide a cert the SBC trusts. You can have 1 cert for "ios devices" but lose the ability to revoke it on a device by device basis.
You could just as well offer up a verisign cert to the devices from the SBC - that part doesn't matter - you're just asking the devices to trust your SBC, whether that's through your enterprise PKI or public, it's not entirely relevant as anyone can wireshark a TLS handshake to your SBC, grab your CA cert and decide to trust it. You can have remote workers need a peer validated enterprise CA cert to come on in. That conferencing case is one where you're trying to prove your interface's trustworthiness to John Q Public and you don't need anything back from him for him to participate in a screen sharing session.
You just want to send him a valid cert so he gets the little green lock in his browser so firefox allows the website to acquire his webcam and microphone. So yes, you can peel it out! Considering our network folks do not see drops for the udp port range on the FW, which I can't believe.
If the device is on the corporate wireless the 2 way audio is normal. What is strange is we are not doing anything where we register directly at ASM internally, so this shouldn't work regardless of where the endpoint is. Look at the SDP on the outside signaling. If the "public IP" of the SBC isn't in the network config, it's possible the endpoint outside is being told to send audio to Yea, those publics are in the network. When running the tracSBC I get success registration and when I attempt a call it goes through and connects just fine.
I see the public IP the endpoint had at the top of the trace, but in the invite from the endpoint to the SBC's.. The registration goes through fine, but I get nothing in the trace once I make and connect the call, which has 2 way audio.
Cutting-edge communication solutions
Assuming that successful media traffic is bypassing the SBC, since it's on the internal network. Registration is one thing RTP is another. SIP uses both ports and IP's for phone traffic. What network guys need to know is SIP cannot change ports while establishing a call. If ports change than you will loss audio. In trunking with UDP you have this thing call ALG that will get in the way and cause these issues if not configured correctly turn off in one firewall model, but may need it on another model.
I don't think this is the issue, but you should do a PCAP and listen to audio and view ports. So this Try on home wifi and mobile. Some mobile carriers internationally suck and don't give you a public IP for your device so you're behind their NAT. I'd try at least a couple of different networks to confirm the theory. Where's the IP from anyway? Not your home wifi I'm pretty sure. I was on the public cell network originally.. When I go to whatsmyip from the device I get Have to find out where that private IP actually came from, but nonetheless.
From my home WiFi same deal.. Is that interworking profile picking up on Avaya-specific extensions? I could only believe the phone is stupid enough to offer a private IP because of a misconfiguration or lack of configuration of telling it where to route audio.
Either some interworking profile isn't passing the phone the intelligence it's looking for, or it might not be getting PPM. Is it the same deal from a Windows Equinox client? I know in the Windows ones there's a "send logs" that just attaches them in an email. It's not terrible if you find the right log file. I'm thinking this through on the fly I'm not entirely sure of the mechanics there, but that's the idea.
It was the endpoint flow media set to the private interface.
Questions tagged [avaya]
I think I need to take a few days away from this and clear my circuits. Ok, back in the fight. For redundancy purposes I have 2 Session Managers to utilize for this setup. The initial config I only have 1 defined in the SBC config. What's the right way to add in the additional SM? I've seen a few threads about the 2nd SM needing to be added as a separate and new policy.
Share this topic
I think you can add it as a second server profile and make the routing profile go "SM1 then SM2". So not sure if an additional port needs to be opened, instead of just But the overall how it works isn't making sense by not being able to just add an additional server to the existing policies. If I'm not mistaken and I haven't looked in a while , I think you just add another of everything and in the profiles there's a priority, so you choose which SM you'd like to go to first.
I believe that's why there's a specific PPM service configuration in the SBC different from the plain old application relay what version of SBC interop notes you're using referring to? Also, how many SMs in your environment total? There's not a 3rd, 4th, 5th that the users may home to normally from a SIP phone inside, correct? I'd suggest a remote worker flow to every SM in your environment with user profiles where you'd reasonably expect those users to try the SBC once in a while.
I'm pretty sure I know what this issue is, but the documentation doesn't say the interfaces that require this. My failover between SBC's was taking 10 full minutes for a client to re-establish connection to the other and am pretty certain it's due to GARP being disabled on our switches. I have to validate with the server team, but I'm reading about the vswitch 'notify switches' setting. I wouldn't think so. Haven't looked at the install docs in a while, and it might be a crossover between both SBC VMs to keep sync, but even if that's the case, I'd expect them to have their own unique IPs to speak to one another and not need GARP.
A crossover interface would obviously not be required. I don't look forward to the day most networking goes that way. Heck, even CM 7. On a normal CM duplex it's 5x ms heartbeats missed that make the standby go active. I can't imagine what kind of tweaking and tuning they did to avoid going split brain all the time.
They don't like the idea of stretching layer2 across dc's and I guess until what you mentioned above there were no capabilities for the geo layer 3 stuff. In circling back to separating mutual vs server auth for ios vs deskphone the current TLS client profiles are set for peer verification, which is what I want.
I created a new TLS server profile requiring peer verfication; However, the only place the TLS server profile is applied is to the signaling interface. Adding a new signaling interface and using the same port is not permitted. The endpoint flows only reference the tls client profiles, which are already set properly.
Yeah, I don't think you get that knob to tinker with at the User Agent level. I get that that's what you want to do, but the SBC probably doesn't see it that way. As in, the TLS profile decision is probably something very low-level. I mean, think about it. It'd have to TLS handshake with the endpoint to know what client type it was to know what security profile to apply. That's putting the cart in front of the horse and asking the horse to divide by 0 :.
So I'd need another IP set for the different phone types. I'm pretty sure, yeah. Avaya one-X Communicator is Avaya's advanced softphone which supports both H. The Avaya one-X Mobile enables business calling, call routing, visual voice mail and directory access through an interface. Avaya Extension-to-Cellular is an original, one-number access capability for business users bridging calls received by an employee's corporate number to mobile device, allowing both to ring simultaneously.
Applications Enablement Server integration with Microsoft Office Communicator or IBM Lotus Sametime extends telephony presence and click-to-call capabilities into these two desktop applications.