Hot answers tagged

13

Start by having a look at this voip-info web page: high availability design. It will explain what high availability is and isn't - in the context of Asterisk. (Its easy to confuse high availability with load balancing) Next, look at this voip-info web page: high availability products. It will explain what is out there for open source / commercial ...


10

The PBX is going to handle all of your endpoints, your hunt groups, your schedules, your auto-attendant, call flows/trees, recordings, etc.. Your SIP provider gives you phone lines. The PBX allows you to use those phone lines. A lot of SIP providers can also provide hosted PBX services, so they handle everything. You could set up a PBX in your office for ...


8

If you are using purely port-based VLANs, then what you have described here will work. Port-based VLANs do not need vlan-aware equipment attached to them. However as soon as you start to add "Voice VLAN" then you are probably going to be using 802.1q VLANs, in which case you do not want to be mixing unmanaged equipment onto those ports. You will want an 802....


6

VOIP. Voice over IP. It requires quite a few more things to be configured than plugging a phone into ethernet, but that puts you in the neighborhood.


5

It's a brute force, asterisk servers always get that kinda thing if connected to the public IP. My solutions are : Install fail2ban, fail2ban will set the iptables and reject ip with continous fail attempts to the asterisk Disable guest sip login, put allowguest=no in the sip.conf In case if you're using VPN, set the from sip external in your PBX Set the [...


4

Problem is with high contention, packets begin to queue up and VOIP packets cannot sit and wait. Are you sure you configured destination ports appropriately for traffic exiting your site? You can't packet shape ingress, only egress. Maybe you configured those destination ports for your LAN side which might include a lot of non-VOIP traffic as well ...


4

You're mixing layer 2 and layer 3 together and they're two different things. VLAN's "operate" at Layer 2 while subnets "operate" at layer 3. It sounds like you have inter-vlan routing configured and you're using that to route traffic between the subnets between the two companies. When you say that you have a /30 that bridges the two switches what I suspect ...


4

TLDR; If you can get the same SLA with your fiber as you can with the T1, go fiber! Not all fiber is the same, you can have Dedicated Ethernet, QoS Lines, and Business Class Internet. Dedicated Ethernet is typically point to point with a provider's head-end, and provides you with consistent speeds, an SLA and sometimes QoS (but not always). You could use ...


4

Try adding the valuehost=dynamic to your guest config in iax.conf


4

If you just want a simple phone, a VoIP equivalent of just renting a single phone line and plugging a dumb phone into it, you don't need the PBX, just a Sip Trunk. If you want the features that the PBX brings to a traditional phone line on your VoIP line, you need a PBX for that too. Some Sip Trunk provides provide a few of these features at the host end ...


4

I'd given up on this, but stumbled across the answer: http://*phone-ip-address*/admin/screendump.bmp Ref from Cisco forums. Tested today on SPA504G with software version 7.6.2a. It's correct right down to the flashing colon of the time and cursor. It delivers 128 x 64 BMP format. Be aware that someone looking at the screendumps can see pretty much ...


3

I don't know whether or not it's an official "best practice" or not, but it's certainly common. I've never heard of it causing an issue at any of my workplaces. We were not using roaming profiles, but we were using network-intensive applications and performing software installs via the network, and, again, we never had any report of slowness due to a call ...


3

This question can't be answered per se; the bandwidth needs of e.g. (a) a self-contained office that has a file server in-house, and (b) a site that constantly moves large image files in and out the building, makes extensive use of cloud storage, and has a bunch of VPNs to remote sites, can easily be two orders of magnitude different, even though both sites ...


3

It is because not all legs / providers properly cancel echo in their network. So, in theory you should not need to have a line side echocan on a digital line, but in practice you sometimes do. For example, a provider might not cancel echo out to each analog endpoint in their network. If you have a digital connection to this network, you get the signal ...


3

Set up a VLAN Put your phones onto that VLAN Put a DHCP Forwarder on the VLAN (your switch should be able to do this). Point the forwarder to the IP address of your existing DHCP server Put a scope on your DHCP server that covers the subnet that the DHCP Helper is assigned to If you are using the piggy-backing function on your phones to plug your PC into ...


3

Lack of audio in SIP calls is nearly always due to firewall or NAT issues. All traffic to/from your EC2 instance traverses a NAT, so I'd look into making sure that the requisite NAT setting in Elastix are set properly.


3

Right now, you only have 1 extension defined in your [test] context ... Extension 20. If you want to be able to place calls between 200 and 201, you'd need to have something like the following: [test] exten => 20,1, Answer() same => n, Playback(hello-world) same => n, Hangup() exten => 20[01], Verbose(2,Call for Extension ${EXTEN}) same =&...


3

There are multiple logs in /var/log/asterisk, check them all Debugging from the console You can also start asterisk from the console and enable debugging for iax2 # asterisk -r asterisk*CLI> iax2 set debug on asterisk*CLI> Then try connecting again from your client. You should see more info on the console. For more info in iax2 debugging asterisk*...


3

You absolutely can do this. Whether it's appropriate for your situation is a different matter. Essentially, SIP trunks are like POTS lines or T1/PRI trunks. They are just dial tone. They can be used (and combined) in various ways depending on the business application. Using that analogy: You could run a individual POTS lines to each desktop (i.e. associate ...


3

This happens due to automated scanners, which probably are trying to bruteforce your passwords. To get rid of such calls, you should disable anonymous users by placing the following options into the [general] section of your sip.conf [general] context=bogus allowguest=no alwaysauthreject=yes


3

First, the register line should have a path set at the end, like: register => myusername:mypassword:myusername@sip.flowroute.com/84106639 Then do a sip reload or service asterisk restart. After that, the sip show peers command should return some kind of status. The qualify=yes option is useful too to check IP connectivity and SIP service status. Also, ...


3

You are missing two items to be placed directly under your register => entry. [general] register => myusername:mypassword@sip.flowroute.com registertimeout=20 registerattempts=0 As soon as you enter these in sip.conf, at the asterisk console enter: *CLI> sip reload *CLI> sip show registry


2

No, it won't work. Dialup is painful enough as it is that making it worse with a VoIP hop is a non-starter. To provide usable dialup service, you need a clean, digital 56Kbps path from your digital modem to the end user's local DAC.


2

From what I have learned spikes in latency making VoIP and other applications unusable in cable modem systems stems from the buffer bloat issues described a few years back. Specifically, cable modem front ends (UBRs) that may hang onto a packet for many seconds while waiting for an open upstream slot to send it on. I am sure there are many much more expert ...


2

Take a look at this link, it will help you. Pay attention to the last column. http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml However g729 is not royalty free from what I understand. I use g711, the quality is great and at 64k per call you would need 8 calls to fill your pipe. g729 is most effective and utilizes ...


2

In my case, the answer was LLDP. By enabling LLDP on my Cisco switches, the phones were able to learn of the Voice VLAN before they started the DHCP process.


2

I would recomend setting up an autoprovision server. You can use TFTP,FTP,HTTP or HTTPS with v6x and v7x firmware Yealink phones (maybe others too). High level instruction on how to do it using DHCP boot provision and FTP file server: Set up FTP server with a user and password (or not, but I would recommend it) Set up DHCP server with boot option 66 (...


2

Do you trust your workstations to not submit data with a fake QoS header just to get more speed? You can not avoid that but on a VLAN you can prioritize the VLAN. Also, on windows doing QoS on the workstations can be tricky ;) That said, I would say in most cases you do not even need QOS; 30 is a small number of phones and workstations.


2

I recommend using a separate VLAN whether you use QOS or not. A separate VLAN allows you to set different DHCP options (such as Boot Server) for your phones without affecting your workstations.


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