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6

I actually use these exact phones in my office. The phone is patched directly back to the PBX, and phone settings such as ext and name are configured at the PBX port level. You will need to locate the patch panel port that your phone is plugged into, and patch into the same port on your PBX when you swap it to retain the ext and what not.


4

Sounds like you need a basic primer on how IP routing works. Yes, each branch should be on its own subnet. Without that, you'd have L2 broadcast domains spanning your WAN which could get, well, interesting. Do the IP phones all connect back to a central softswitch at Branch 1? If so, as long as your routing is set up correctly (either static or dynamic ...


3

You can't directly access the loopback address of another computer. You can however access it via the IP address (assuming it's a typical network). Open a command prompt on the Win7 machine, run ipconfig, note the IP address. Type that IP address into whatever program you're using on the smartphone. Note: This answer is incredibly vague because you've given ...


3

You can monitor the statistics of every single port on the switch using SNMP. This will give you your individual throughput on each port, tally them all up at any given time and that gives you your total traffic. If that traffic is higher than your switches backplane, then your switch is the bottleneck. You should also monitor your uplink ports. If they ...


3

Fiber optic will likely be the way to go. However, it is not simple to terminate and requires special equipment. You'll probably have to pay someone to come out and terminate the cable for you.


2

You can register your own URL handler in the browser (how depends on browser) and some softphones have the ability to run on commandline with a number as parameter (certainly the proprietary softphone zultys supply with their systems does). Qutecom dox, limited as they are indicate this: qutecom -c call\123456 should work. The "hard bit" in my experience ...


2

There's no real reason you can't. Just wire all the ports to each other, and you'll be fine. You don't want to swap connectors or anything, just wire everything in parallel. This is exactly what a commercial phone splitter does internally.


2

There is always Asterisk. the open source PBX, which I believe, will allow you to do this, however, it would typically require an analogue or digital telephone line and an interface card. It may be able to ineract with your GSM device, but it's not something I've ever looked into.


2

Your best bet is to try and use AT commands; I happen to have one of the same devices (E160E) on my desk, but I've only ever used it as a data modem. Curiosly, lsusb reports it as Bus 002 Device 006: ID 12d1:1003 Huawei Technologies Co., Ltd. E220 HSDPA Modem / E270 HSDPA/HSUPA Modem Have a look at How to Use Modem AT Commands for USB Modem at Linux; with ...


1

You might find it easier to get it up and running using a phone with bluetooth and the chan_mobile module in Asterisk, the problem with using a USB connected GSM modem is getting the audio into it (placing data calls is easy) You could also look at the 'Festival' system in Asterisk that does text-to-speech, rather than pre-record a load of MP3's it could ...


1

from your branch configuration they are all in seperate /24 subnets, so thats not an issue. depending on your vpn configuration, it should simply be a matter of, giving the IP phone an IP within the subnet of the branch in which it is connected and telling it to connect to the IP of the VoIP system in Branch1, your vpn should automatically send out routes ...


1

My gut here is that this is either a routing issue or maybe a QoS issue. In my experience, one-way audio is almost always a routing issue. Let's be certain of the only devices involved (at L3): Phone <-> Router <-IPSEC-> Router <-> Phone. There may or may not be switches between the phones and the routers. Now... on these assumptions, can you post ...


1

Make sure routing works correctly; set up a notebook in the same network as the ip phone for testing (it's easier). Furthermore, can't you set the default route just on the IP phone? I didn't have really the time to dive into your routing setup, but it looks OK on the first glance. I'd say something might be set up wrongly on the phone. Oh and make sure ...


1

I'm assuming you mean the 7961 as I can't find any reference to a Cisco 7661 phone. Sorry I'm not familiar with the methods used for Cisco's to automatically discover voice VLAN's but a temporary bodge - press settings, **#, then Network configuration, then Operational VLAN ID, and manually enter the desired VLAN. Also it may be worth doing a factory reset ...


1

I don't know what you are using as switches, but on catalyst the following configuration should help: interface FastEthernet0/12 switchport access vlan <data vlan id> switchport voice vlan <voice vlan id> Another thing to look at IIRC: the Cisco phones are using CDP to discover the voice vlan and hop into it. While disabling CDP sounds ...


1

Typically m/a/c to phones like registering/unregistering etc. can be done through the PBX admin interfaces themselves. Most phones also allow you to enter "codes" on the phones for manual registration/de-registration if you choose to do so. It's probably unlikely that you'll get a lot of telecom help here, so I'm posting this as possibilities from my ...


1

It's been awhile since I've done a phone move but if this is a digital keyset then it usually follows that the output from the PBX for this keyset/extension is patched to a patch panel that feeds a particular wall jack. When moving the keyset to another wall jack you usually need to move the patch cable for that keyset to the new wall jack.


1

Anytime you dealing with the PSTN outside of a simple POTS line, the phone number and line are separate. For instance, at my work we have 23 "lines" provided by a VoIP-PRI, and almost 90 phone numbers on that. It can go the other way as well, you could have thousands of lines and only a single number. You need one 'line' per conversation.


1

No, this is NOT possible on any version of CCM, as far as I know.



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